sn4638 ignoriert pakete

JoeMoes · 12 Aug 2009

Moin,

ich hab eine Smartnode 4638 an einem kleinen Asterisk Server laufen. Der Asterisk hat die Version 1.6.1.1 und die Patton hat die Version 5.T vom 28.5.09 bzw. auch getestet mit der Version 5.2 vom 14.01.09.

Zum Aufbau:
Sip Telefon -> Asterisk -> 4638 -> PSTN (Anlagenanschluss mit 2 B Kanälen)
Asterisk: 192.168.216.5
Patton: 192.168.216.7

Zu meinem kleinen Problemchen:
Ein Anruf wird vom einem Sip Telefon in das Telefon Netz gemacht. In meinem Fall: 486 -> 0040432xxxxx. Als Sip Telefoner kann man die Gegenseite hören nur man kann nich gehört werden. Ist der Anruf andersherum, funktioniert alles wunderbar.

Noch mal schnell die Konfig von der Patton:

Code:

#----------------------------------------------------------------#
#                                                                #
# SN4638/5BIS                                                    #
# R5.T 2009-05-28 H323 SIP BRI                                   #
# 2009-08-12T08:26:05                                            #
# SN/00A0BA048B52                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
administrator root password xxx== encrypted
dns-client server 192.168.216.2
dns-client server 192.168.210.2
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 192.168.210.15 port 123 version 4
system hostname SN4638-HA1

system

  ic voice 0
    low-bitrate-codec g729

system
  clock-source 1 bri 0 0
  clock-source 2 bri 0 1
  clock-source 3 bri 0 2
  clock-source 4 bri 0 3
  clock-source 5 bri 0 4

profile napt NAPT_WAN

profile ppp default

profile tone-set default

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 transparent-clearmode rx-length 20 tx-length 20
  fax transmission 1 relay t38-udp

profile pstn default

profile sip default

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface WAN
    ipaddress dhcp
    use profile napt NAPT_WAN
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  interface LAN
    ipaddress 192.168.1.1 255.255.255.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  interface IF_IP_LAN
    ipaddress 192.168.216.7 255.255.255.0

context ip router
  route 0.0.0.0 0.0.0.0 192.168.216.1 0

context cs switch
  digit-collection timeout 3

  routing-table called-e164 RT_CDPN_OUT
    route .T dest-service HG_2_OUT

  interface isdn IF_S0_00
    route call dest-table RT_CDPN_OUT

  interface isdn IF_S0_01
    route call dest-table RT_CDPN_OUT

  interface isdn IF_S0_02
    route call dest-table RT_CDPN_OUT

  interface isdn IF_S0_03
    route call dest-table RT_CDPN_OUT

  interface isdn IF_S0_04
    route call dest-table RT_CDPN_OUT

  interface sip IF_VOIP_GATEWAY_HA
    bind context sip-gateway GW_SIP_HA
    route call dest-service HG_2_PBX
    remote asterisk.ha.xxx.de 5060
    early-connect
    early-disconnect

  interface sip IF_VOIP_GATEWAY_B
    bind context sip-gateway GW_SIP_B
    route call dest-service HG_2_PBX
    remote asterisk.b.xxx.de 5060
    early-connect
    early-disconnect

  interface sip IF_VOIP_GATEWAY_BM
    bind context sip-gateway GW_SIP_BM
    route call dest-service HG_2_PBX
    remote asterisk.bm.xxx.de 5060
    early-connect
    early-disconnect

  service hunt-group HG_2_OUT
    timeout 6
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable
    route call 1 dest-interface IF_VOIP_GATEWAY_HA
    route call 2 dest-interface IF_VOIP_GATEWAY_B
    route call 3 dest-interface IF_VOIP_GATEWAY_BM

  service hunt-group HG_2_PBX
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable
    route call 1 dest-interface IF_S0_00

context cs switch
  no shutdown

authentication-service SER_AUTH_OB
  username SN4638-HA1 password xxx== encrypted

location-service SER_LOC_CERT_HA
  domain 1 asterisk.ha.xxx.de

  identity SN4638-HA1

    authentication outbound
      authenticate 1 authentication-service SER_AUTH_OB username SN4638-HA1

    registration outbound
      registrar asterisk.ha.xxx.de 5060
      register auto

location-service SER_LOC_CERT_B
  domain 1 asterisk.b.xxx.de

  identity SN4638-HA1

    authentication outbound
      authenticate 1 authentication-service SER_AUTH_OB username SN4638-HA1

    registration outbound
      registrar asterisk.b.xxx.de 5060
      register auto

location-service SER_LOC_CERT_BM
  domain 1 asterisk.bm.xxx.de

  identity SN4638-HA1

    authentication outbound
      authenticate 1 authentication-service SER_AUTH_OB username SN4638-HA1

    registration outbound
      registrar asterisk.bm.xxx.de 5060
      register auto

context sip-gateway GW_SIP_HA

  interface IF_IP_LAN_HA
    bind interface IF_IP_LAN context router port 5060

context sip-gateway GW_SIP_HA
  bind location-service SER_LOC_CERT_HA
  no shutdown

context sip-gateway GW_SIP_B

  interface IF_IP_LAN_B
    bind interface IF_IP_LAN context router port 5061

context sip-gateway GW_SIP_B
  bind location-service SER_LOC_CERT_B
  no shutdown

context sip-gateway GW_SIP_BM

  interface IF_IP_LAN_BM
    bind interface IF_IP_LAN context router port 5062

context sip-gateway GW_SIP_BM
  bind location-service SER_LOC_CERT_BM
  no shutdown

port ethernet 0 0
  medium auto
  encapsulation ip
  bind interface WAN router
  no shutdown

port ethernet 0 1
  medium auto
  encapsulation ip
  bind interface LAN router
  bind interface IF_IP_LAN router
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_00 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_01 switch

port bri 0 1
  no shutdown

port bri 0 2
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_02 switch

port bri 0 2
  no shutdown

port bri 0 3
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_03 switch

port bri 0 3
  no shutdown

port bri 0 4
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_04 switch

port bri 0 4
  no shutdown

Nun zum Trace von der Asterisk Seite aus:

Code:

[Aug 12 09:31:53]     -- Executing [0040432xxxxx@trunk-halle:10011] Dial("SIP/74486-bc0f5598", "SIP/0040432xxxxx@sn4638-ha1,300") in new stack
[Aug 12 09:31:53]   == Using SIP RTP TOS bits 184
[Aug 12 09:31:53]   == Using SIP RTP CoS mark 5
[Aug 12 09:31:53] Audio is at 192.168.216.5 port 27848
[Aug 12 09:31:53] Adding codec 0x8 (alaw) to SDP
[Aug 12 09:31:53] Adding codec 0x2 (gsm) to SDP
[Aug 12 09:31:53] Adding codec 0x4 (ulaw) to SDP
[Aug 12 09:31:53] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 12 09:31:53] Reliably Transmitting (no NAT) to 192.168.216.7:5060:
INVITE sip:0040432xxxxx@sn4638-ha1 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Date: Wed, 12 Aug 2009 07:31:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 919511600 919511600 IN IP4 192.168.216.5
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.216.5
t=0 0
m=audio 27848 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Aug 12 09:31:53]     -- Called 0040432xxxxx@sn4638-ha1
[Aug 12 09:31:53]
<--- SIP read from UDP://192.168.216.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport=5060;received=192.168.216.5
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


<------------->
[Aug 12 09:31:53] --- (8 headers 0 lines) ---
[Aug 12 09:31:58]
<--- SIP read from UDP://192.168.216.7:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport=5060;received=192.168.216.5
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


<------------->
[Aug 12 09:31:58] --- (9 headers 0 lines) ---
[Aug 12 09:31:58]
<--- SIP read from UDP://192.168.216.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport=5060;received=192.168.216.5
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Supported: replaces
Content-Type: application/sdp
Content-Length: 197

v=0
o=MxSIP 0 14 IN IP4 192.168.216.7
s=SIP Call
c=IN IP4 192.168.216.7
t=0 0
m=audio 4886 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<------------->
[Aug 12 09:31:58] --- (11 headers 10 lines) ---
[Aug 12 09:31:58] Found RTP audio format 8
[Aug 12 09:31:58] Found RTP audio format 101
[Aug 12 09:31:58] Peer audio RTP is at port 192.168.216.7:4886
[Aug 12 09:31:58] Found audio description format PCMA for ID 8
[Aug 12 09:31:58] Found audio description format telephone-event for ID 101
[Aug 12 09:31:58] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Aug 12 09:31:58] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Aug 12 09:31:58] Peer audio RTP is at port 192.168.216.7:4886
[Aug 12 09:31:58] list_route: hop: <sip:[email protected]:5060>
[Aug 12 09:31:58] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
[Aug 12 09:31:58] set_destination: set destination to 192.168.216.7, port 5060
[Aug 12 09:31:58] Transmitting (no NAT) to 192.168.216.7:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK3d4319b6;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.1
Content-Length: 0


---
[Aug 12 09:31:58]     -- SIP/sn4638-ha1-0086e848 answered SIP/74486-bc0f5598
[Aug 12 09:31:58]     -- Native bridging SIP/74486-bc0f5598 and SIP/sn4638-ha1-0086e848
[Aug 12 09:31:58] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
[Aug 12 09:31:58] set_destination: set destination to 192.168.216.7, port 5060
[Aug 12 09:31:58] Audio is at 192.168.216.5 port 27848
[Aug 12 09:31:58] Adding codec 0x8 (alaw) to SDP
[Aug 12 09:31:58] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 12 09:31:58] Reliably Transmitting (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 919511600 919511601 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 56208 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Aug 12 09:31:59] Retransmitting #1 (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 919511600 919511601 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 56208 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Aug 12 09:32:00] Retransmitting #2 (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 919511600 919511601 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 56208 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Aug 12 09:32:02] Retransmitting #3 (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 919511600 919511601 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 56208 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Aug 12 09:32:06] Retransmitting #4 (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 919511600 919511601 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 56208 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Aug 12 09:32:06] Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
[Aug 12 09:32:06]   == Spawn extension (trunk-halle, 0040432xxxxx, 10011) exited non-zero on 'SIP/74486-bc0f5598'
[Aug 12 09:32:07] Really destroying SIP dialog 'a018690616b58739' Method: BYE
[Aug 12 09:32:14] Retransmitting #5 (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 919511600 919511601 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 56208 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Aug 12 09:32:14]
<--- SIP read from UDP://192.168.216.7:5060 --->
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.216.7:5060;branch=z9hG4bK603c43930bb645af8
Max-Forwards: 70
From: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
To: "Test User" <sip:[email protected]>;tag=as7a390b51
Call-ID: [email protected]
CSeq: 26187 BYE
User-Agent: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


<------------->
[Aug 12 09:32:14] --- (9 headers 0 lines) ---
[Aug 12 09:32:14] Sending to 192.168.216.7 : 5060 (no NAT)
[Aug 12 09:32:14] Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: BYE)
[Aug 12 09:32:14]
<--- Transmitting (no NAT) to 192.168.216.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.216.7:5060;branch=z9hG4bK603c43930bb645af8;received=192.168.216.7
From: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
To: "Test User" <sip:[email protected]>;tag=as7a390b51
Call-ID: [email protected]
CSeq: 26187 BYE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

Und nun die Seite von der Patton:

Code:

SN4638-HA1#
07:31:52  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:0040432xxxxx@sn4638-ha1 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Date: Wed, 12 Aug 2009 07:31:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 919511600 919511600 IN IP4 192.168.216.5
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.216.5
t=0 0
m=audio 27848 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

07:31:52  SIP_TR> [STACK] > Stack: to 192.168.216.5
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport=5060;received=192.168.216.5
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


07:31:56  SIP_TR> [STACK] > Stack: to 192.168.216.5
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport=5060;received=192.168.216.5
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


07:31:57  SIP_TR> [STACK] > Stack: to 192.168.216.5
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport=5060;received=192.168.216.5
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Supported: replaces
Content-Type: application/sdp
Content-Length: 197

v=0
o=MxSIP 0 14 IN IP4 192.168.216.7
s=SIP Call
c=IN IP4 192.168.216.7
t=0 0
m=audio 4886 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

07:31:57  SIP_TR> [STACK] < Stack: from 192.168.216.5
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK3d4319b6;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.1
Content-Length: 0


07:31:57  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 919511600 919511601 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 56208 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

07:31:58  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 919511600 919511601 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 56208 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

07:31:59  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 919511600 919511601 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 56208 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

07:32:01  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 919511600 919511601 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 56208 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

07:32:05  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 919511600 919511601 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 56208 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

07:32:13  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK495e7ff9;rport
Max-Forwards: 70
From: "Test User" <sip:[email protected]>;tag=as7a390b51
To: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 919511600 919511601 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 56208 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

07:32:13  SIP_TR> [STACK] > Stack: to 192.168.216.5
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.216.7:5060;branch=z9hG4bK603c43930bb645af8
Max-Forwards: 70
From: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
To: "Test User" <sip:[email protected]>;tag=as7a390b51
Call-ID: [email protected]
CSeq: 26187 BYE
User-Agent: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


07:32:13  SIP_TR> [STACK] < Stack: from 192.168.216.5
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.216.7:5060;branch=z9hG4bK603c43930bb645af8;received=192.168.216.7
From: <sip:0040432xxxxx@sn4638-ha1>;tag=859047562
To: "Test User" <sip:[email protected]>;tag=as7a390b51
Call-ID: [email protected]
CSeq: 26187 BYE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

Was mir auffällt ist, das der Asterisk Retransmissions sendet, die aber von der Patton gnadenlos ignoriert werden. :\

Hat jemand eine Idee?

JoeMoes · 28 Aug 2009

Ich bin die ganze Zeit weiter am testen woran der Fehler liegen könnte. Nun habe ich mich ein wenig auf die Patton versteift.

Raus bekommen habe ich, das folgende konstellation funktioniert, wenn ein Mobil Tel angerufen wird:
SIP Tel -> Asterisk -> Patton -> PSTN

Wird ein Festnetz Telefon angerufen, funktioniert es leider nicht.

Kurz zum Ablauf auf dem ISDN Kanal. Ich sehe auf einem Handy Gespräche folgendes auf dem ISDN Kanal:

Setup Pakete:

Code:

06:58:34  ISDN  > #   7 p: 0 from CC key: 8388618 L3SetupReq
06:58:34  ISDN  > #   8      Service:Transparent cing: sub: ced: sub:
06:58:34  ISDN  > #   9 cause : Transparent
06:58:34  ISDN  > #  10      hex:04 03 90 90 A3 1E 02 85 83 6C 05 00 80 34 38 36 70 0D 80 30 30 31 37 32 35 34 33 30 30 38 30
06:58:34  ISDN  > #  11      IE: Bearer
06:58:34  ISDN  > #  12      IE: ProgressInd
06:58:34  ISDN  > #  13      IE: CallingPartyNbr  xxx
06:58:34  ISDN  > #  14      IE: CalledPartyNbr  001725xxxxxx
06:58:34  ISDN  > #  15      *newPc BR-U, tei:  0, sapi:  0
06:58:34  ISDN  > #  16       key: 8388618, Stack:2V01b
06:58:34  ISDN  > #  17       EuroISDN
06:58:34  ISDN  > #  18      ---> Layer2  tei:  0 sapi:  0 Setup
06:58:34  ISDN  > #  19      hex:08 01 0A 05 04 03 90 90 A3 1E 02 85 83 6C 05 00 80 34 38 36 70 0D 80 30 30 31 37 32 35 34 33 30 30
06:58:34  ISDN  > #  20      IE: Bearer
06:58:34  ISDN  > #  21      IE: ProgressInd
06:58:34  ISDN  > #  22      IE: CallingPartyNbr  xxx
06:58:34  ISDN  > #  23      IE: CalledPartyNbr  001725xxxxxx
06:58:34  ISDN  > #  24      new call state : L3PcSt01U
06:58:34  ISDN  > #  25 p: 0 S: sapi: 0 cr=0 ea=0 tei:  0 ea=1 INFO Nr( 51) Ns( 46) pf=0

06:58:34  ISDN  > #  26 p: 0 R: sapi: 0 cr=0 ea=0 tei:  0 ea=1 RR   Nr( 47)         pf=0
06:58:34  ISDN  > #  27 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr( 47) Ns( 51) pf=0

06:58:34  ISDN  > #  28 p: 0 <--- Layer2  tei:  0 sapi:  0 SetupAck
06:58:34  ISDN  > #  29      hex:08 01 8A 0D 18 01 89
06:58:34  ISDN  > #  30      IE: ChannelId  BCh0 excl  otherIf
06:58:34  ISDN  > #  31      ALLOC bchan BCh0 cref : 0x0a
06:58:34  ISDN  > #  32      to  CC key: 8388618 L3MoreInfoInd
06:58:34  ISDN  > #  33      new call state : L3PcSt02U
06:58:34  ISDN  > #  34 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr( 52)         pf=0

Alle sind sich einig... Alerting folgt nun:

Code:

06:58:41  ISDN  > #  35 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr( 47) Ns( 52) pf=0
06:58:41  ISDN  > #  36 p: 0 <--- Layer2  tei:  0 sapi:  0 Alerting
06:58:41  ISDN  > #  37      hex:08 01 8A 01
06:58:41  ISDN  > #  38      to  CC key: 8388618 L3AlertingInd
06:58:41  ISDN  > #  39      new call state : L3PcSt04U
06:58:41  ISDN  > #  40 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr( 53)         pf=0

Ein Traum soweit... nun das gleiche auf ein Festnetz Nummer:

Code:

07:00:15  ISDN  > # 162 p: 0 from CC key: 8388620 L3SetupReq
07:00:15  ISDN  > # 163      Service:Transparent cing: sub: ced: sub:
07:00:15  ISDN  > # 164 cause : Transparent
07:00:15  ISDN  > # 165      hex:04 03 90 90 A3 1E 02 85 83 6C 05 00 80 34 38 36 70 0D 80 30 30 34 30 34 33 32 30 34 34 38 36
07:00:15  ISDN  > # 166      IE: Bearer
07:00:15  ISDN  > # 167      IE: ProgressInd
07:00:15  ISDN  > # 168      IE: CallingPartyNbr  xxx
07:00:15  ISDN  > # 169      IE: CalledPartyNbr  00404xxxxxxx
07:00:15  ISDN  > # 170      *newPc BR-U, tei:  0, sapi:  0
07:00:15  ISDN  > # 171       key: 8388620, Stack:2V01b
07:00:15  ISDN  > # 172       EuroISDN
07:00:15  ISDN  > # 173      ---> Layer2  tei:  0 sapi:  0 Setup
07:00:15  ISDN  > # 174      hex:08 01 0C 05 04 03 90 90 A3 1E 02 85 83 6C 05 00 80 34 38 36 70 0D 80 30 30 34 30 34 33 32 30 34 34
07:00:15  ISDN  > # 175      IE: Bearer
07:00:15  ISDN  > # 176      IE: ProgressInd
07:00:15  ISDN  > # 177      IE: CallingPartyNbr  xxx
07:00:15  ISDN  > # 178      IE: CalledPartyNbr  00404xxxxxxx
07:00:15  ISDN  > # 179      new call state : L3PcSt01U
07:00:15  ISDN  > # 180 p: 0 S: sapi: 0 cr=0 ea=0 tei:  0 ea=1 INFO Nr( 60) 
Ns( 52) pf=0
07:00:15  ISDN  > # 181 p: 0 R: sapi: 0 cr=0 ea=0 tei:  0 ea=1 RR   Nr( 53)         pf=0
07:00:15  ISDN  > # 182 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr( 53) Ns( 60) pf=0
07:00:15  ISDN  > # 183 p: 0 <--- Layer2  tei:  0 sapi:  0 SetupAck
07:00:15  ISDN  > # 184      hex:08 01 8C 0D 18 01 89
07:00:15  ISDN  > # 185      IE: ChannelId  BCh0 excl  otherIf
07:00:15  ISDN  > # 186      ALLOC bchan BCh0 cref : 0x0c
07:00:15  ISDN  > # 187      to  CC key: 8388620 L3MoreInfoInd
07:00:15  ISDN  > # 188      new call state : L3PcSt02U
07:00:15  ISDN  > # 189 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr( 61)         pf=0

Nun kommt folgendes... Proceeding?!

Code:

07:00:19  ISDN  > # 190 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr( 53) Ns( 61) pf=0
07:00:19  ISDN  > # 191 p: 0 <--- Layer2  tei:  0 sapi:  0 CallProc
07:00:19  ISDN  > # 192      hex:08 01 8C 02
07:00:19  ISDN  > # 193      to  CC key: 8388620 L3ProceedingInd
07:00:19  ISDN  > # 194      new call state : L3PcSt03U
07:00:19  ISDN  > # 195 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr( 62)         pf=0

Fragt mich nicht, warum ich vom Amt nun Proceeding bekomme, aber das ist der Moment, wo die Patton aus dem Tritt kommt und natürlich das Gespräch aufbaut zum Asterisk. Doch dieser hat es irgendwie nicht geschnallt (siehe Trace von oben).

Jemand eine Idee?

JoeMoes · 31 Aug 2009

So... nun hab ich noch ein Ersatzgerät gekauft um das einmal zu testen doch leider war es der gleiche Erfolg. :\ Also an der Patton hardwareseitig ist alles sauber.

Dennoch verstehe ich nicht, warum die Patton die Verbindung zum laufen bringt. Sogar laut Wikipedia steht, das ein ISDN Verbindung folgenden Standard Ablauf besitzt (http://de.wikipedia.org/wiki/ISDN-Verbindungsaufbau:

Code:

Progress (Setup)
MoreInfo (Info)
Proceeding
Alerting
(Connect)

Laut Wiki sollte die Verbindung zum Handy eher nicht funktionieren als zum Festapparat.

Proceeding ist wohl aber keine Pflichtmeldung auf dem ISDN Stack.

Ich pack noch mal mein letzten Trace rein... vielleicht packe ich das Problem auch an der falschen Stelle an (SIP Tel baut eine Verbindung zu einer 0404x Nummer auf über eine Asterisk -> Patton Anlage):

Code:

SN4638-HA1#
SN4638-HA1#13:47:59  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:00404xxx@sn4638-ha1 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK1093853d;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as6791dd42
To: <sip:00404xxx@sn4638-ha1>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Date: Mon, 31 Aug 2009 13:48:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 1366886348 1366886348 IN IP4 192.168.216.5
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.216.5
t=0 0
m=audio 6074 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Network -> GW_SIP_HA/192.168.216.7
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: E164-Number -> 486
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: URI -> sip:[email protected]
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Type-Of-Number -> Unknown
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Numbering-Plan -> Unknown
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Presentation-Indicator -> Presentation allowed
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Name -> "xxx"
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Screening-Indicator -> User provided, not screened
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Supports Overlap-Sending -> true
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Supported Codecs -> Voice: G.711 A-law[20/20]
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Codec Negotiating -> Idle
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Unique Identifier -> [email protected]
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: IP-Address -> 192.168.216.5
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Call-Leg-ID -> 0x00cb42d8
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: State -> CONNECTED
13:47:59  CC    > [Call 00fae318] Set call property: Context -> 0x00000022
13:47:59  CC    > [Call 00fae318] Set call property: Information-Transfer-Capability -> 3.1kHz Audio
13:47:59  CC    > [Call 00fae318] Set call property: Hops -> 0x00000010
13:47:59  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Dial to provider router (IF_VOIP_GATEWAY_HA-precall-service) using call 00fae318
13:47:59  CC    > [EP router-00cb5498/incoming] Accept call 00fae318
13:47:59  CC    > [EP router-00cb5498/incoming] Set call-leg property: E164-Number -> 00404xxx
13:47:59  CC    > [EP router-00cb5498/incoming] Set call-leg property: Type-Of-Number -> Unknown
13:47:59  CC    > [EP router-00cb5498/incoming] Set call-leg property: Numbering-Plan -> Unknown
13:47:59  CC    > [EP router-00cb5498/incoming] Set call-leg property: Name ->
13:47:59  CC    > [EP router-00cb5498/incoming] Set call-leg property: Alert-Info ->
13:47:59  CC    > [EP router-00cb5498/incoming] Set call-leg property: URI -> sip:00404xxx@sn4638-ha1
13:47:59  CC    > [EP router-00cb5498/incoming] Set call-leg property: Network -> router
13:47:59  CC    > [EP router-00cb5498/incoming] Set call-leg property: Call-Leg-ID -> 0x00cb4088
13:47:59  CC    > [EP router-00cb5498/incoming] Set call-leg property: State -> TRYING
13:47:59  CC    > [EP router-00cb5498] Start route-lookup
13:47:59  CR    > [switch] Routing-Lookup:
13:47:59  CR    >   Execute all entries in table IF_VOIP_GATEWAY_HA-precall-service
13:47:59  CR    >   Execute all entries in table HG_2_PBX-dest
13:47:59  CR    >   Execute all entries in table route-found-place-call
13:47:59  CR    >   Lookup result: Route found; place call (timeout=0)
13:47:59  CC    > [EP router-00cb5498] Route found; immediately place call
13:47:59  CC    > [EP router-00cb5498] Route to provider 'HG_2_PBX'
13:47:59  CC    > [EP router-00cb5498/outgoing] Set call-leg property: E164-Number -> 486
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: URI -> sip:[email protected]
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: Type-Of-Number -> Unknown
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: Numbering-Plan -> Unknown
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: Presentation-Indicator -> Presentation allowed
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: Name -> "xxx"
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: Screening-Indicator -> User provided, not screened
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: Supports Overlap-Sending -> true
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: Supported Codecs -> Voice: G.711 A-law[20/20]
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: Codec Negotiating -> Idle
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: Unique Identifier -> [email protected]
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: IP-Address -> 192.168.216.5
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: Network -> router
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: Call-Leg-ID -> 0x00cafb08
13:48:00  CC    > [EP router-00cb5498/outgoing] Set call-leg property: State -> CONNECTED
13:48:00  CC    > [Call 00cafa10] Set call property: Context -> 0x00000022
13:48:00  CC    > [Call 00cafa10] Set call property: Information-Transfer-Capability -> 3.1kHz Audio
13:48:00  CC    > [Call 00cafa10] Set call property: Hops -> 0x0000000f
13:48:00  CC    > [EP router-00cb5498/outgoing] Dial to provider HG_2_PBX () using call 00cafa10
13:48:00  CC    > [EP HG_2_PBX-00a45458/incoming] Accept call 00cafa10
13:48:00  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: E164-Number -> 00404xxx
13:48:00  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: Type-Of-Number -> Unknown
13:48:00  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: Numbering-Plan -> Unknown
13:48:00  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: Name ->
13:48:00  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: Alert-Info ->
13:48:00  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: URI -> sip:00404xxx@sn4638-ha1
13:48:00  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: Network -> HG_2_PBX
13:48:00  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: Call-Leg-ID -> 0x00a45738
13:48:00  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: State -> TRYING
13:48:00  CC    > [EP HG_2_PBX-00a45458] Hunt to IF_S0_00 ()
13:48:00  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: Allows Push-Back -> false
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: E164-Number -> 486
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: URI -> sip:[email protected]
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Type-Of-Number -> Unknown
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Numbering-Plan -> Unknown
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Presentation-Indicator -> Presentation allowed
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Name -> "xxx"
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Screening-Indicator -> User provided, not screened
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Supports Overlap-Sending -> true
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Supported Codecs -> Voice: G.711 A-law[20/20]
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Codec Negotiating -> Idle
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Unique Identifier -> [email protected]
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: IP-Address -> 192.168.216.5
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Allows Push-Back -> false
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Network -> HG_2_PBX
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Call-Leg-ID -> 0x00a46800
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: State -> CONNECTED
13:48:00  CC    > [Call 00a46718] Set call property: Context -> 0x00000022
13:48:00  CC    > [Call 00a46718] Set call property: Information-Transfer-Capability -> 3.1kHz Audio
13:48:00  CC    > [Call 00a46718] Set call property: Hops -> 0x0000000e
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Dial to provider IF_S0_00 () using call 00a46718
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Accept call 00a46718
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: E164-Number -> 00404xxx
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Type-Of-Number -> Unknown
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Numbering-Plan -> Unknown
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Name ->
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Alert-Info ->
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: URI -> sip:00404xxx@sn4638-ha1
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Quality-Of-Service -> MOS 4.50, DS0
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Network -> IF_S0_00
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Call-Leg-ID -> 0x00a4a1b0
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: State -> TRYING
13:48:00  ISDN  > #2624 p: 0 from CC key: 8388628 L3SetupReq
13:48:00  ISDN  > #2625      Service:Transparent cing: sub: ced: sub:
13:48:00  ISDN  > #2626 cause : Transparent
13:48:00  ISDN  > #2627      hex:04 03 90 90 A3 1E 02 85 83 6C 05 00 80 34 38 36 70 0D 80 30 30 34 30 34 33 32 30 34 34 38 36
13:48:00  ISDN  > #2628      IE: Bearer
13:48:00  ISDN  > #2629      IE: ProgressInd
13:48:00  ISDN  > #2630      IE: CallingPartyNbr  486
13:48:00  ISDN  > #2631      IE: CalledPartyNbr  00404xxx
13:48:00  ISDN  > #2632      *newPc BR-U, tei:  0, sapi:  0
13:48:00  ISDN  > #2633       key: 8388628, Stack:2V01b
13:48:00  ISDN  > #2634       EuroISDN
13:48:00  ISDN  > #2635      ---> Layer2  tei:  0 sapi:  0 Setup
13:48:00  ISDN  > #2636      hex:08 01 14 05 04 03 90 90 A3 1E 02 85 83 6C 05 00 80 34 38 36 70 0D 80 30 30 34 30 34 33 32 30 34 34
13:48:00  ISDN  > #2637      IE: Bearer
13:48:00  ISDN  > #2638      IE: ProgressInd
13:48:00  ISDN  > #2639      IE: CallingPartyNbr  486
13:48:00  ISDN  > #2640      IE: CalledPartyNbr  00404xxx
13:48:00  ISDN  > #2641      new call state : L3PcSt01U
13:48:00  ISDN  > #2642 p: 0 S: sapi: 0 cr=0 ea=0 tei:  0 ea=1 INFO Nr( 23) Ns(125) pf=0
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Endpoint-Is-Isdn -> true
13:48:00  ISDN  > #2643 p: 0 R: sapi: 0 cr=0 ea=0 tei:  0 ea=1 RR   Nr(126)         pf=0
13:48:00  CC    > [Call 00fae318] Set call property: Hops -> 0x0000000f
13:48:00  CC    > [EP router-00cb5498/incoming] Set call-leg property: Allows Push-Back -> false
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Network -> router
13:48:00  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Call-Leg-ID -> 0x00cafb08
13:48:00  ISDN  > #2644 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr(126) Ns( 23) pf=0
13:48:00  ISDN  > #2645 p: 0 <--- Layer2  tei:  0 sapi:  0 SetupAck
13:48:00  ISDN  > #2646      hex:08 01 94 0D 18 01 89
13:48:00  ISDN  > #2647      IE: ChannelId  BCh0 excl  otherIf
13:48:00  ISDN  > #2648      ALLOC bchan BCh0 cref : 0x14
13:48:00  ISDN  > #2649      to  CC key: 8388628 L3MoreInfoInd
13:48:00  ISDN  > #2650      new call state : L3PcSt02U
13:48:00  ISDN  > #2651 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr( 24)         pf=0
13:48:00  DP    > TDM-00/00/00: registerEventCallback(0xa4b4a4)
13:48:00  DP    > TDM-00/00/00: addToContext(00000022)
13:48:00  DP    > TDM-00/00/00: configure(Pstn-Config)
13:48:00  DP    > TDM-00/00/00: The following config package could not be converted:
SEQUENCE {
  packageId UUIDpackageId UUID = CfgToneDetection
  argument SEQUENCE {
    profile GeneralString IMPLICIT [CONTEXT 1] = 'default' OPTIONAL
    detectDtmf BOOLEAN IMPLICIT [CONTEXT 2] = false OPTIONAL
  }
}
13:48:00  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: State -> ADDRESS-INCOMPLETE
13:48:00  DP    > TDM-00/00/00: setOperationMode(send/receive)
13:48:00  TDM   > TDM-00/00/00: Acquiring DSP resource.
13:48:00  TDM   > TDM-00/00/00: Reconfigure resource.
13:48:00  TDM   > [DSP 0xe7f738] Received RTP Media Config.
13:48:00  TDM   > [DSP 0xe7f738] voice config.
13:48:00  TDM   > [DSP 0xe7f738] dejitter config.
13:48:00  TDM   > [DSP 0xe7f738] pstn config.
13:48:00  TDM   > [DSP 0xe7f738] caller-id config (None).
13:48:00  TDM   > [DSP 0xe7f738] Admin operation mode inactive -> send/receive
13:48:00  TDM   > [DSP 0xe7f738] Dsp operation mode inactive -> send/receive
13:48:00  TDM   > [DSP 0xe7f738]: Connected port 0 bChannel 0 to DSP timeslot 0
13:48:00  DSP   > [00000003][344782510] State=closed: Event=openVoice
13:48:00  DSP   > [00000003][344782510] Set tx plugin to voice.
13:48:00  DSP   > [00000003][344782510] Opening
13:48:00  DSP   > [00000003][344782510] Dsp 0 Channel 0 configured for G.711aLaw on timeslot 0:
    Silence compression OFF
    Echo canceller ON (NLPM Adaptive, HybridLoss 6)
    Post filter: ON, HighPass filter: ON
    Output gain: 0dB
    Input gain:  0dB
    Transmission: Fax None / Modem: None
    Max bit rate (relay): Fax 14400 bit/s / Modem 9600 bit/s
    Fax/modem gain (relay): -9.5dB
    Fax/modem bypass codec: G.711aLaw
    Modem Dejitter buffer size: 200ms
    Fax Dejitter buffer size: 200ms
    T.38 Error correction: ON
    T.38 HDLC image tx: ON
    Fax protocol mode: T.38 UDP
    DTMF signal gain: lf -2dB, hf -1dB, mute encoder OFF
    Fax detection forced: OFF
    Caller-ID: Disabled

13:48:00  DSP   > [Channel 0 0] ACTIVATING CHANNEL
13:48:00  DSP   > [00000003][344782510] Set tx media type 2.
13:48:00  DSP   > [00000003][344782510] Set rx media type 2.
13:48:00  DSP   > [00000003][344782510] New state=voice
13:48:00  TDM   > [DSP 0xe7f738] Activate VOICE dejitter configuration
13:48:00  Dejit > Input length: 80
13:48:00  Dejit > [0xe7fa10] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

13:48:00  Dejit > Input length: 80
13:48:00  Dejit > [0xe7fec0] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

13:48:00  Dejit > Input length: 80
13:48:00  Dejit > [0xe80180] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

13:48:00  TDM   > [DSP 0xe7f738] Activate VOICE tx-buffer configuration
13:48:00  TDM   > [DSP 0xe7f738] Activating TX.
13:48:00  TDM   > [DSP 0xe7f738] Activating RX.
13:48:00  DSP   > Scheduler: added source processor 0xe7fb8c
13:48:00  CC    > [EP HG_2_PBX-00a45458] Hunting tentatively succeeded
13:48:00  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: State -> ADDRESS-INCOMPLETE
13:48:00  DSP   > [Channel 0 0] FIRST_PACKET_RECEIVED
13:48:00  SIP_TR> [STACK] > Stack: to 192.168.216.5
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK1093853d;rport=5060;received=192.168.216.5
From: "xxx" <sip:[email protected]>;tag=as6791dd42
To: <sip:00404xxx@sn4638-ha1>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


13:48:00  CC    > [EP router-00cb5498/incoming] Set call-leg property: State -> ADDRESS-INCOMPLETE

SN4638-HA1#
SN4638-HA1#
SN4638-HA1#
SN4638-HA1#
SN4638-HA1#
SN4638-HA1#13:48:04  ISDN  > #2652 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr(126) Ns( 24) pf=0
13:48:04  ISDN  > #2653 p: 0 <--- Layer2  tei:  0 sapi:  0 CallProc
13:48:04  ISDN  > #2654      hex:08 01 94 02
13:48:04  ISDN  > #2655      to  CC key: 8388628 L3ProceedingInd
13:48:04  ISDN  > #2656      new call state : L3PcSt03U
13:48:04  ISDN  > #2657 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr( 25)         pf=0
13:48:04  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: State -> PROCEEDING
13:48:04  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: State -> PROCEEDING
13:48:04  ISDN  > #2658 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr(126) Ns( 25) pf=0
13:48:04  ISDN  > #2659 p: 0 <--- Layer2  tei:  0 sapi:  0 Alerting
13:48:04  ISDN  > #2660      hex:08 01 94 01 1E 02 81 88
13:48:04  ISDN  > #2661      IE: ProgressInd
13:48:04  ISDN  > #2662      to  CC key: 8388628 L3AlertingInd
13:48:04  ISDN  > #2663      new call state : L3PcSt04U
13:48:04  ISDN  > #2664 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr( 26)         pf=0
13:48:04  CC    > [EP router-00cb5498/incoming] Set call-leg property: State -> PROCEEDING
13:48:04  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: State -> ALERTING
13:48:04  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Provides Data -> true
13:48:04  SIP_TR> [STACK] > Stack: to 192.168.216.5
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK1093853d;rport=5060;received=192.168.216.5
From: "xxx" <sip:[email protected]>;tag=as6791dd42
To: <sip:00404xxx@sn4638-ha1>;tag=1967376477
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


13:48:04  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: Provides Data -> true
13:48:04  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: State -> ALERTING
13:48:04  CC    > [EP router-00cb5498/incoming] Set call-leg property: Provides Data -> true
13:48:04  CC    > [EP router-00cb5498] Routing succeeded
13:48:04  CC    > [EP router-00cb5498/incoming] Transfer call 00cafa10 to 00fae318 ==> conference
13:48:04  CC    > [EP router-00cb5498/incoming] Drop call 00fae318
13:48:04  CC    > [EP router-00cb5498/incoming] Set call-leg property: Provides Data -> false
13:48:04  CC    > [EP router-00cb5498/incoming] Set call-leg property: Cause -> Normal call clearing
13:48:04  CC    > [EP router-00cb5498/incoming] Set call-leg property: State -> RELEASED
13:48:04  CC    > [EP router-00cb5498/outgoing] Drop call 00cafa10
13:48:04  CC    > [EP router-00cb5498/outgoing] Set call-leg property: Cause -> Normal call clearing
13:48:04  CC    > [EP router-00cb5498/outgoing] Set call-leg property: State -> RELEASED
13:48:04  DP    > RTP-00/0022: registerEventCallback(0xa4bdfc)
13:48:04  DP    > RTP-00/0022: configure(Voice-Config)
13:48:04  DP    > RTP-00/0022: event(General/ConfigChanged-Event)
13:48:04  DP    > RTP-00/0022: configure(Dejitter-Config)
13:48:04  DP    > RTP-00/0022: event(General/ConfigChanged-Event)
13:48:04  DP    > RTP-00/0022: addToContext(00000022)
13:48:04  DP    > RTP-00/0022: addConnector(ffffffff)
13:48:04  DP    > TDM-00/00/00: addConnector(ffffffff)
13:48:04  TDM   > [DSP 0xe7f738] Add connector notify. ID 0
13:48:04  DSP   > Scheduler: removed source processor 0xe7fb8c
13:48:04  DSP   > Scheduler: added source processor 0xe7fb8c
13:48:04  DP    > RTP-00/0022: getConfiguration(00010300)
13:48:04  DP    > RTP-00/0022: getConfiguration(00010b00)
13:48:04  DP    > TDM-00/00/00: configure(Voice-Config)
13:48:04  TDM   > [DSP 0xe7f738] voice config.
13:48:04  TDM   > [DSP 0xe7f738] reconfiguring DSP.
13:48:04  DSP   > [00000003][344786870] State=voice: Event=close
13:48:04  DSP   > [00000003][344786870] Set tx plugin to none.
13:48:04  DSP   > [00000003][344786870] Set tx media type 2.
13:48:04  DSP   > [00000003][344786870] Set rx media type 2.
13:48:04  DSP   > [00000003][344786880] Closing (dsp 0, channel 0)
13:48:04  DSP   > [Channel 0 0] SENDING IDLE PACKET
13:48:04  DSP   > [00000003][344786880] New state=closed
13:48:04  DSP   > [00000003][344786880] State=closed: Event=openVoice
13:48:04  DSP   > [00000003][344786880] Set tx plugin to voice.
13:48:04  DSP   > [00000003][344786880] Opening
13:48:04  DSP   > [00000003][344786880] Dsp 0 Channel 0 configured for G.711aLaw on timeslot 0:
    Silence compression OFF
    Echo canceller ON (NLPM Adaptive, HybridLoss 6)
    Post filter: ON, HighPass filter: ON
    Output gain: 0dB
    Input gain:  0dB
    Transmission: Fax Relay / Modem: None
    Max bit rate (relay): Fax 14400 bit/s / Modem 9600 bit/s
    Fax/modem gain (relay): -9.5dB
    Fax/modem bypass codec: G.711aLaw
    Modem Dejitter buffer size: 200ms
    Fax Dejitter buffer size: 200ms
    T.38 Error correction: ON
    T.38 HDLC image tx: ON
    Fax protocol mode: T.38 UDP
    DTMF signal gain: lf -2dB, hf -1dB, mute encoder ON
    Fax detection forced: OFF
    Caller-ID: Disabled

13:48:04  DSP   > [00000003][344786880] Set tx media type 2.
13:48:04  DSP   > [00000003][344786880] Set rx media type 2.
13:48:04  DSP   > [00000003][344786880] New state=voice
13:48:04  TDM   > [DSP 0xe7f738] Activate VOICE dejitter configuration
13:48:04  Dejit > Input length: 80
13:48:04  Dejit > [0xe7fa10] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

13:48:04  Dejit > Input length: 80
13:48:04  Dejit > [0xe7fec0] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

13:48:04  Dejit > Input length: 80
13:48:04  Dejit > [0xe80180] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

13:48:04  TDM   > [DSP 0xe7f738] Activate VOICE tx-buffer configuration
13:48:04  DP    > RTP-00/0022: getConfiguration(00010e00)
13:48:04  DP    > TDM-00/00/00: configure(Dejitter-Config)
13:48:04  TDM   > [DSP 0xe7f738] dejitter config.
13:48:04  TDM   > [DSP 0xe7f738] Reconfiguring dejitter.
13:48:04  TDM   > [DSP 0xe7f738] Activate VOICE dejitter configuration
13:48:04  Dejit > Input length: 80
13:48:04  Dejit > [0xe7fa10] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 6

13:48:04  Dejit > Input length: 80
13:48:04  Dejit > [0xe7fec0] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 6

13:48:04  Dejit > Input length: 80
13:48:04  Dejit > [0xe80180] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 6

13:48:04  DP    > RTP-00/0022: registerEventCallback(0xda5970)
13:48:04  DP    > TDM-00/00/00: registerEventCallback(0xda5970)
13:48:04  DSP   > [Channel 0 0] IDLE_PACKET_RECEIVED
13:48:04  DSP   > [Channel 0 0] ACTIVATING CHANNEL
13:48:04  DP    > RTP-00/0022: configure(RTP-Config)
13:48:04  RTP   > [02000022] Configure local source:  0xc0a8d807/4940/10773480
13:48:04  RTP   > [02000022] Configure remote source: 0xc0a8d805/6074/14949600
13:48:04  RTP   > [02000022] Next hop gateway is:     0x00000000
13:48:04  RTP   > [TERM 2000022] Config changed (codec=G.711 A-law | media=audio)
13:48:04  DP    > RTP-00/0022: event(General/ConfigChanged-Event)
13:48:04  DP    > RTP-00/0022: getConfiguration(00010300)
13:48:04  DP    > TDM-00/00/00: configure(MediaType/RTP-Config)
13:48:04  TDM   > TDM-00/00/00: Received Media Config.
13:48:04  TDM   > TDM-00/00/00: Resource can be re-used.
13:48:04  TDM   > [DSP 0xe7f738] Received RTP Media Config.
13:48:04  DSP   > [00000003][344786900] Try codec update for G.711aLaw.
13:48:04  TDM   > [DSP 0xe7f738] Reconfiguring dejitter.
13:48:04  TDM   > [DSP 0xe7f738] Activate VOICE dejitter configuration
13:48:04  Dejit > Input length: 80
13:48:04  Dejit > [0xe7fa10] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 6

13:48:04  Dejit > Input length: 80
13:48:04  Dejit > [0xe7fec0] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 6

13:48:04  Dejit > Input length: 80
13:48:04  Dejit > [0xe80180] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 6

13:48:04  TDM   > [DSP 0xe7f738] Reconfiguring tx-buffer.
13:48:04  TDM   > [DSP 0xe7f738] Activate VOICE tx-buffer configuration
13:48:04  DP    > RTP-00/0022: setOperationMode(send/receive)
13:48:04  RTP   > [02000022] Set mode: INACTIVE -> TX/RX
13:48:04  DPMUX > Changed next processor of port 0x0022face: 0xe420a0
13:48:04  RTP   > [02000022] (BCD) Event=enable | New State=broken
13:48:04  TDM   > DATA_BUFF_TX: Update Connection. Processor: e42160
13:48:04  DPMUX > Activating directpath port: 0x0022face
13:48:04  DPMUX >   Protocol:                 RTP
13:48:04  DPMUX >   Local transport address:  0xc0a8d807/4940 (2 port(s))
13:48:04  DPMUX >   Remote transport address: 0xc0a8d805/6074 (2 port(s))
13:48:04  DPMUX >   IP Interface:             3
13:48:04  RTP   > [02000022] Next hop gateway is:     0xc0a8d805
13:48:04  NTE   > [00e42578] Tx activation request.
13:48:04  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Provides Data -> true
13:48:04  DSP   > [Channel 0 0] FIRST_PACKET_RECEIVED
13:48:04  SIP_TR> [STACK] > Stack: to 192.168.216.5
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK1093853d;rport=5060;received=192.168.216.5
From: "xxx" <sip:[email protected]>;tag=as6791dd42
To: <sip:00404xxx@sn4638-ha1>;tag=1967376477
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Supported: replaces
Content-Type: application/sdp
Content-Length: 197

v=0
o=MxSIP 0 43 IN IP4 192.168.216.7
s=SIP Call
c=IN IP4 192.168.216.7
t=0 0
m=audio 4940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

13:48:04  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Network -> GW_SIP_HA/192.168.216.7
13:48:04  SIP_TR> [STACK] < Stack: from 192.168.216.5
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK629ad274;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as6791dd42
To: <sip:00404xxx@sn4638-ha1>;tag=1967376477
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.1
Content-Length: 0


13:48:04  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK1093853d;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as6791dd42
To: <sip:00404xxx@sn4638-ha1>;tag=1967376477
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1366886348 1366886349 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 52624 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

13:48:04  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Call-Leg-ID -> 0x00cb42d8
13:48:04  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Provides Data -> true
13:48:04  DP    > RTP-00/0022: getStatistics(00020900)
13:48:04  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Quality-Of-Service -> MOS 4.40, RTP, G.711 A-law (20ms), Local: Rx 0 pkts, 0 bytes, 0 lost, jitter 0 ms, Tx 3 pkts, 480 bytes, rtt 0 ms, Remote: Rx 0 pkts, 0 bytes, 0 lost, jitter 0 ms, Tx 0 pkts, 0 bytes, rtt 0 ms
13:48:04  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Quality-Of-Service -> MOS 4.40, RTP, G.711 A-law (20ms), Local: Rx 0 pkts, 0 bytes, 0 lost, jitter 0 ms, Tx 3 pkts, 480 bytes, rtt 0 ms, Remote: Rx 0 pkts, 0 bytes, 0 lost, jitter 0 ms, Tx 0 pkts, 0 bytes, rtt 0 ms
13:48:04  DP    > RTP-00/0022: event(Event/Mediatype-RTP/Connection-Established)
13:48:04  RTP   > [02000022] (BCD) Event=rx-rtp | New State=established
13:48:04  RTP   > [02000022] Rx first packet (seq=62179)
13:48:04  Dejit > Input length: 160
13:48:04  Dejit > [0xe7fa10] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 3

13:48:04  MEDIA > [00000003][344787150] Codec=G711A | Media=VOICE | Ecan=ON | Vad=OFF
13:48:04  Dejit > [0xe7fa10] underrun phase detected, freezed packet measurements
13:48:05  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK1093853d;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as6791dd42
To: <sip:00404xxx@sn4638-ha1>;tag=1967376477
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1366886348 1366886349 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 52624 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

13:48:05  DP    > RTP-00/0022: event(Event/Mediatype-RTP/Connection-Broken)
13:48:05  RTP   > [02000022] (BCD) Event=timeout | New State=broken
13:48:06  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK1093853d;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as6791dd42
To: <sip:00404xxx@sn4638-ha1>;tag=1967376477
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1366886348 1366886349 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 52624 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

13:48:07  RTP   > [02000022] RTCP TX (SR) Timestamp=344789350
13:48:07  RTP   > [02000022]  Local TX-Info: packets=121, octets=19360
13:48:07  RTP   > [02000022]  Local RX-Info: packets=4, octets=640, lost=0, jitter=18, since last=0mS
13:48:08  ISDN  > #2665 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr(126) Ns( 26) pf=0
13:48:08  ISDN  > #2666 p: 0 <--- Layer2  tei:  0 sapi:  0 Connect
13:48:08  ISDN  > #2667      hex:08 01 94 07 29 05 09 08 1F 0F 30 4C 0D 00 81 30 34 30 34 33 32 30 34 34 38 36
13:48:08  ISDN  > #2668      IE: DateTime
13:48:08  ISDN  > #2669      IE: ConnectedNbr  0404xxx
13:48:08  ISDN  > #2670      to  CC key: 8388628 L3SetupConf
13:48:08  ISDN  > #2671      Connects bchan BCh0  cref : 0x14
13:48:08  ISDN  > #2672      ---> Layer2  tei:  0 sapi:  0 ConnectAck
13:48:08  ISDN  > #2673      hex:08 01 14 0F
13:48:08  ISDN  > #2674      new call state : L3PcSt10U
13:48:08  ISDN  > #2675 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr( 27)         pf=0
13:48:08  ISDN  > #2676 p: 0 S: sapi: 0 cr=0 ea=0 tei:  0 ea=1 INFO Nr( 27) Ns(126) pf=0
13:48:08  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Screening-Indicator -> User provided, verified and passed
13:48:08  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Presentation-Indicator -> Presentation allowed
13:48:08  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: E164-Number -> 0404xxx
13:48:08  ISDN  > #2677 p: 0 R: sapi: 0 cr=0 ea=0 tei:  0 ea=1 RR   Nr(127)         pf=0
13:48:08  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: State -> CONNECTED
13:48:08  DP    > TDM-00/00/00: The following config package could not be converted:
SEQUENCE {
  packageId UUIDpackageId UUID = CfgToneDetection
  argument SEQUENCE {
    profile GeneralString IMPLICIT [CONTEXT 1] = 'default' OPTIONAL
    detectDtmf BOOLEAN IMPLICIT [CONTEXT 2] = true OPTIONAL
  }
}
13:48:08  CC    > [EP HG_2_PBX-00a45458] Hunting definitely succeeded
13:48:08  CC    > [EP HG_2_PBX-00a45458/incoming] Transfer call 00a46718 to 00fae318 ==> conference
13:48:08  CC    > [EP HG_2_PBX-00a45458/incoming] Drop call 00fae318
13:48:08  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: Provides Data -> false
13:48:08  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: Cause -> Normal call clearing
13:48:08  CC    > [EP HG_2_PBX-00a45458/incoming] Set call-leg property: State -> RELEASED
13:48:08  CC    > [EP HG_2_PBX-00a45458/outgoing] Drop call 00a46718
13:48:08  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Provides Data -> false
13:48:08  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: Cause -> Normal call clearing
13:48:08  CC    > [EP HG_2_PBX-00a45458/outgoing] Set call-leg property: State -> RELEASED
13:48:08  ISDN  > #2678 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr(127) Ns( 27) pf=0
13:48:08  ISDN  > #2679 p: 0 <--- Layer2  tei:  0 sapi:  0 Disconnect
13:48:08  ISDN  > #2680      hex:08 01 94 45 08 02 80 90 1E 02 81 88
13:48:08  ISDN  > #2681      IE: Cause  NormalCallClearing
13:48:08  ISDN  > #2682      IE: ProgressInd
13:48:08  ISDN  > #2683      to  CC key: 8388628 L3DisconnectInd
13:48:08  ISDN  > #2684      new call state : L3PcSt12U
13:48:08  ISDN  > #2685 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr( 28)         pf=0
13:48:08  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Cause -> Normal call clearing
13:48:08  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: State -> DISCONNECTING
13:48:08  DP    > TDM-00/00/00: The following config package could not be converted:
SEQUENCE {
  packageId UUIDpackageId UUID = CfgToneDetection
  argument SEQUENCE {
    profile GeneralString IMPLICIT [CONTEXT 1] = 'default' OPTIONAL
    detectDtmf BOOLEAN IMPLICIT [CONTEXT 2] = false OPTIONAL
  }
}
13:48:08  DP    > TDM-00/00/00: setOperationMode(receive-only)
13:48:08  TDM   > [DSP 0xe7f738] Admin operation mode send/receive -> receive-only
13:48:08  TDM   > [DSP 0xe7f738] Dsp operation mode send/receive -> receive-only
13:48:08  TDM   > [DSP 0xe7f738] Deactivating TX.
13:48:08  DSP   > Scheduler: removed source processor 0xe7fb8c
13:48:08  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Drop call 00fae318
13:48:08  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Provides Data -> false
13:48:08  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: Cause -> Normal call clearing
13:48:08  CC    > [EP IF_VOIP_GATEWAY_HA-00fae580/active] Set call-leg property: State -> RELEASED
13:48:08  DP    > RTP-00/0022: subtractFromContext(00000022)
13:48:08  DP    > TDM-00/00/00: unregisterEventCallback(0xda5970)
13:48:08  DP    > RTP-00/0022: unregisterEventCallback(0xda5970)
13:48:08  TDM   > DATA_BUFF_TX: Update Connection. Processor: 0
13:48:08  DP    > TDM-00/00/00: removeConnector(00000000)
13:48:08  TDM   > [DSP 0xe7f738] Remove connector notify. ID: 0
13:48:08  DP    > RTP-00/0022: removeConnector(00000000)
13:48:08  SIP_TR> [STACK] > Stack: to 192.168.216.5
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.216.7:5060;branch=z9hG4bK1484cad478aa69f5c
Max-Forwards: 70
From: <sip:00404xxx@sn4638-ha1>;tag=1967376477
To: "xxx" <sip:[email protected]>;tag=as6791dd42
Call-ID: [email protected]
CSeq: 21927 BYE
User-Agent: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


13:48:08  DP    > TDM-00/00/00: subtractFromContext(00000022)
13:48:08  SIP_TR> [STACK] < Stack: from 192.168.216.5
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.216.7:5060;branch=z9hG4bK1484cad478aa69f5c;received=192.168.216.7
From: <sip:00404xxx@sn4638-ha1>;tag=1967376477
To: "xxx" <sip:[email protected]>;tag=as6791dd42
Call-ID: [email protected]
CSeq: 21927 BYE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


13:48:08  CC    > [EP IF_S0_00-00a47ae0/active] Drop call 00fae318
13:48:08  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: Provides Data -> false
13:48:08  CC    > [EP IF_S0_00-00a47ae0/active] Set call-leg property: State -> RELEASED
13:48:08  DP    > TDM-00/00/00: setOperationMode(inactive)
13:48:08  TDM   > [DSP 0xe7f738] Admin operation mode receive-only -> inactive
13:48:08  TDM   > [DSP 0xe7f738] Dsp operation mode receive-only -> inactive
13:48:08  TDM   > [DSP 0xe7f738] Deactivating RX.
13:48:08  DSP   > [00000003][344790660] State=voice: Event=close
13:48:08  DSP   > [00000003][344790660] Set tx plugin to none.
13:48:08  DSP   > [00000003][344790660] Set tx media type 2.
13:48:08  DSP   > [00000003][344790660] Set rx media type 2.
13:48:08  DSP   > [00000003][344790660] Closing (dsp 0, channel 0)
13:48:08  DSP   > [Channel 0 0] SENDING IDLE PACKET
13:48:08  DSP   > [00000003][344790660] New state=closed
13:48:08  TDM   > [DSP 0xe7f738] Disconnected port 0 bChannel 0 from DSP timeslot 0
13:48:08  TDM   > TDM-00/00/00 Releasing DSP resource.
13:48:08  ISDN  > #2686 p: 0 from CC key: 8388628 L3ReleaseReq
13:48:08  ISDN  > #2687 cause : NormalCallClearing
13:48:08  ISDN  > #2688      Disconnects bchan BCh0  cref : 0x14
13:48:08  ISDN  > #2689      ---> Layer2  tei:  0 sapi:  0 Release
13:48:08  ISDN  > #2690      hex:08 01 14 4D 08 02 80 90
13:48:08  ISDN  > #2691      IE: Cause  NormalCallClearing
13:48:08  ISDN  > #2692      new call state : L3PcSt19U
13:48:08  ISDN  > #2693 p: 0 S: sapi: 0 cr=0 ea=0 tei:  0 ea=1 INFO Nr( 28) Ns(127) pf=0
13:48:08  DSP   > [Channel 0 0] IDLE_PACKET_RECEIVED
13:48:08  DP    > RTP-00/0022: unregisterEventCallback(0xa4bdfc)
13:48:08  DP    > RTP-00/0022: setOperationMode(inactive)
13:48:08  RTP   > [02000022] Set mode: TX/RX -> INACTIVE
13:48:08  DPMUX > Changed next processor of port 0x0022face: 0x0
13:48:08  RTP   > [02000022] (BCD) Event=disable | New State=disabled
13:48:08  DPMUX > Deactivating directpath port: 0x0022face
13:48:08  NTE   > [00e42578] Tx reset request.
13:48:08  NTE   > Scheduler: Unregister element immediately 0x00e42578
13:48:08  NTE   > Scheduler: Unregistered element: 00e42578
13:48:08  RTP   > [TERM 2000022] Tx-State=idle | Tx-Event=reset
13:48:08  RTP   > [TERM 2000022] Rx-State=idle | Rx-Event=reset
13:48:08  RTP   > [TERM 2000022] Stop Rx Fiter-Timeout
13:48:08  ISDN  > #2694 p: 0 R: sapi: 0 cr=0 ea=0 tei:  0 ea=1 RR   Nr(  0)         pf=0
13:48:08  ISDN  > #2695 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr(  0) Ns( 28) pf=0
13:48:08  ISDN  > #2696 p: 0 <--- Layer2  tei:  0 sapi:  0 ReleaseCom
13:48:08  ISDN  > #2697      hex:08 01 94 5A
13:48:08  ISDN  > #2698      to  CC key: 8388628 L3ReleaseConf
13:48:08  ISDN  > #2699      DEALLOC bchan BCh0 cref : 0x14
13:48:08  ISDN  > #2700      new call state : L3PcSt00U
13:48:08  ISDN  > #2701      *delPc U, tei:  0 sapi:  0
13:48:08  ISDN  > #2702 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr( 29)         pf=0
13:48:08  DP    > TDM-00/00/00: unregisterEventCallback(0xa4b4a4)

Hat noch niemand dieses Problem gehabt?

JoeMoes · 3 Sep 2009

Ich gebe nicht auf

Auch wenn ich hier ein Monolog halte, aber ich hoffe es hilft vielleicht irgend jemand später einmal.

Ich bin ein Schritt weiter gekommen in dem ich folgende Einstellung deaktiviert habe:

Inband Info - Accept Transparent

Jetzt sieht das Phänomen gleich aus - egal ob es ein Festnetz oder Mobilfunk Telefonat ist.

Zur Erinnerung... immer noch der gleiche Aufbau: SIP Telefon -> Asterisk -> Patton -> PSTN

Ich höre nun endlich ein klingeln und sehe auch wenn die Gegenseite abgenommen hat. Doch leider kann die eine Seite nicht sprechen bzw. man kann nichts hören. Den PSTN Teilnehmer kann man hören aber leider das SIP Telefon nicht.

Ein aktueller Auszug meiner Konfig (Patton):

Code:

#----------------------------------------------------------------#
#                                                                #
# SN4638/5BIS                                                    #
# R5.T 2009-05-28 H323 SIP BRI                                   #
# 2009-09-03T08:36:19                                            #
# SN/00A0BA048B52                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
administrator root password xxx== encrypted
dns-client server 192.168.216.2
dns-client server 192.168.210.2
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary 192.168.210.15 port 123 version 4
system hostname SN4638-HA1

system

  ic voice 0
    low-bitrate-codec g729

system
  clock-source 1 bri 0 0
  clock-source 2 bri 0 1
  clock-source 3 bri 0 2
  clock-source 4 bri 0 3
  clock-source 5 bri 0 4

profile napt NAPT_WAN

profile ppp default

profile tone-set default

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 transparent-clearmode rx-length 20 tx-length 20
  fax transmission 1 relay t38-udp

profile pstn default

profile sip default

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface WAN
    ipaddress dhcp
    use profile napt NAPT_WAN
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  interface LAN
    ipaddress 192.168.1.1 255.255.255.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  interface IF_IP_LAN
    ipaddress 192.168.216.7 255.255.255.0

context ip router
  route 0.0.0.0 0.0.0.0 192.168.216.1 0

context cs switch
  digit-collection timeout 3

  routing-table called-e164 RT_CDPN_OUT
    route .T dest-service HG_2_OUT

  interface isdn IF_S0_00
    route call dest-table RT_CDPN_OUT
    no inband-info accept transparent

  interface isdn IF_S0_01
    route call dest-table RT_CDPN_OUT

  interface isdn IF_S0_02
    route call dest-table RT_CDPN_OUT

  interface isdn IF_S0_03
    route call dest-table RT_CDPN_OUT

  interface isdn IF_S0_04
    route call dest-table RT_CDPN_OUT

  interface sip IF_VOIP_GATEWAY_HA
    bind context sip-gateway GW_SIP_HA
    route call dest-service HG_2_PBX
    remote asterisk.ha.xxx.de 5060
    early-connect
    early-disconnect

  interface sip IF_VOIP_GATEWAY_B
    bind context sip-gateway GW_SIP_B
    route call dest-service HG_2_PBX
    remote asterisk.b.xxx.de 5060
    early-connect
    early-disconnect

  interface sip IF_VOIP_GATEWAY_BM
    bind context sip-gateway GW_SIP_BM
    route call dest-service HG_2_PBX
    remote asterisk.bm.xxx.de 5060
    early-connect
    early-disconnect

  service hunt-group HG_2_OUT
    timeout 6
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable
    route call 1 dest-interface IF_VOIP_GATEWAY_HA
    route call 2 dest-interface IF_VOIP_GATEWAY_B
    route call 3 dest-interface IF_VOIP_GATEWAY_BM

  service hunt-group HG_2_PBX
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable
    route call 1 dest-interface IF_S0_00

context cs switch
  no shutdown

authentication-service SER_AUTH_OB
  username SN4638-HA1 password wXMAn9SZYkL0gxhmDYAw0g== encrypted

location-service SER_LOC_CERT_HA
  domain 1 asterisk.ha.xxx.de

  identity SN4638-HA1

    authentication outbound
      authenticate 1 authentication-service SER_AUTH_OB username SN4638-HA1

    authentication inbound
      authenticate 1 authentication-service SER_AUTH_OB username SN4638-HA1

    registration outbound
      registrar asterisk.ha.xxx.de 5060
      proxy none
      lifetime 3600
      register auto
      retry-timeout on-system-error 10
      retry-timeout on-client-error 10
      retry-timeout on-server-error 10

    registration inbound
      contact asterisk.ha.xxx.de 5060 switch IF_VOIP_GATEWAY_HA priority 1

location-service SER_LOC_CERT_B
  domain 1 asterisk.b.xxx.de

  identity SN4638-HA1

    authentication outbound
      authenticate 1 authentication-service SER_AUTH_OB username SN4638-HA1

    registration outbound
      registrar asterisk.b.xxx.de 5060
      register auto

location-service SER_LOC_CERT_BM
  domain 1 asterisk.bm.xxx.de

  identity SN4638-HA1

    authentication outbound
      authenticate 1 authentication-service SER_AUTH_OB username SN4638-HA1

    registration outbound
      registrar asterisk.bm.xxx.de 5060
      register auto

context sip-gateway GW_SIP_HA

  interface IF_IP_LAN_HA
    bind interface IF_IP_LAN context router port 5060

context sip-gateway GW_SIP_HA
  bind location-service SER_LOC_CERT_HA
  no shutdown

context sip-gateway GW_SIP_B

  interface IF_IP_LAN_B
    bind interface IF_IP_LAN context router port 5061

context sip-gateway GW_SIP_B
  bind location-service SER_LOC_CERT_B
  no shutdown

context sip-gateway GW_SIP_BM

  interface IF_IP_LAN_BM
    bind interface IF_IP_LAN context router port 5062

context sip-gateway GW_SIP_BM
  bind location-service SER_LOC_CERT_BM
  no shutdown

port ethernet 0 0
  medium auto
  encapsulation ip
  bind interface WAN router
  no shutdown

port ethernet 0 1
  medium auto
  encapsulation ip
  bind interface LAN router
  bind interface IF_IP_LAN router
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_00 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_01 switch

port bri 0 1
  no shutdown

port bri 0 2
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_02 switch

port bri 0 2
  no shutdown

port bri 0 3
  clock auto
  encapsulation q921

  q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_03 switch

port bri 0 3
  no shutdown

port bri 0 4
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_04 switch

port bri 0 4
  no shutdown

Asterisk:

Code:

[SN4638-HA1]
name=SN4638-HA1
host=dynamic
nat=no
type=peer
cancallforward=yes
canreinvite=yes
context=incoming_sn4638-ha1
dtmfmode=rfc2833
insecure=port,invite
language=de
secret=xxx
maxexpiry=3600
disallow=all
allow=alaw
subscribecontext=sipextensions
defaultuser=SN4638-HA1
limitonpeers=yes
notifyringing=yes
notifybusy=yes
notifyhold=yes
useclientcode=yes

Und ein aktueller Trace (vom Asterisk aus):

Code:

[Sep  3 10:56:47]   == Using SIP RTP TOS bits 184
[Sep  3 10:56:47]   == Using SIP RTP CoS mark 5
[Sep  3 10:56:47]     -- Executing [00404xxx@trunk-ha:10011] Dial("SIP/74486-008cc1d8", "SIP/00404xxx@sn4638-ha1,300") in new stack
[Sep  3 10:56:47]   == Using SIP RTP TOS bits 184
[Sep  3 10:56:47]   == Using SIP RTP CoS mark 5
[Sep  3 10:56:47] Audio is at 192.168.216.5 port 12484
[Sep  3 10:56:47] Adding codec 0x8 (alaw) to SDP
[Sep  3 10:56:47] Adding codec 0x2 (gsm) to SDP
[Sep  3 10:56:47] Adding codec 0x4 (ulaw) to SDP
[Sep  3 10:56:47] Adding non-codec 0x1 (telephone-event) to SDP
[Sep  3 10:56:47] Reliably Transmitting (no NAT) to 192.168.216.7:5060:
INVITE sip:00404xxx@sn4638-ha1 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Date: Thu, 03 Sep 2009 08:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 395895816 395895816 IN IP4 192.168.216.5
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.216.5
t=0 0
m=audio 12484 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep  3 10:56:47]     -- Called 00404xxx@sn4638-ha1
[Sep  3 10:56:48]
<--- SIP read from UDP://192.168.216.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport=5060;received=192.168.216.5
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


<------------->
[Sep  3 10:56:48] --- (8 headers 0 lines) ---
[Sep  3 10:56:52]
<--- SIP read from UDP://192.168.216.7:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport=5060;received=192.168.216.5
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


<------------->
[Sep  3 10:56:52] --- (9 headers 0 lines) ---
[Sep  3 10:56:52]
<--- SIP read from UDP://192.168.216.7:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport=5060;received=192.168.216.5
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


<------------->
[Sep  3 10:56:52] --- (9 headers 0 lines) ---
[Sep  3 10:56:52]     -- SIP/sn4638-ha1-00951f88 is ringing
[Sep  3 10:57:00]
<--- SIP read from UDP://192.168.216.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport=5060;received=192.168.216.5
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Supported: replaces
Content-Type: application/sdp
Content-Length: 197

v=0
o=MxSIP 0 59 IN IP4 192.168.216.7
s=SIP Call
c=IN IP4 192.168.216.7
t=0 0
m=audio 4932 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<------------->
[Sep  3 10:57:00] --- (11 headers 10 lines) ---
[Sep  3 10:57:00] Found RTP audio format 8
[Sep  3 10:57:00] Found RTP audio format 101
[Sep  3 10:57:00] Peer audio RTP is at port 192.168.216.7:4932
[Sep  3 10:57:00] Found audio description format PCMA for ID 8
[Sep  3 10:57:00] Found audio description format telephone-event for ID 101
[Sep  3 10:57:00] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Sep  3 10:57:00] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep  3 10:57:00] Peer audio RTP is at port 192.168.216.7:4932
[Sep  3 10:57:00] list_route: hop: <sip:[email protected]:5060>
[Sep  3 10:57:00] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
[Sep  3 10:57:00] set_destination: set destination to 192.168.216.7, port 5060
[Sep  3 10:57:00] Transmitting (no NAT) to 192.168.216.7:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK799ef6e3;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.1
Content-Length: 0


---
[Sep  3 10:57:00]     -- SIP/sn4638-ha1-00951f88 answered SIP/74486-008cc1d8
[Sep  3 10:57:00]     -- Native bridging SIP/74486-008cc1d8 and SIP/sn4638-ha1-00951f88
[Sep  3 10:57:00] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
[Sep  3 10:57:00] set_destination: set destination to 192.168.216.7, port 5060
[Sep  3 10:57:00] Audio is at 192.168.216.5 port 12484
[Sep  3 10:57:00] Adding codec 0x8 (alaw) to SDP
[Sep  3 10:57:00] Adding non-codec 0x1 (telephone-event) to SDP
[Sep  3 10:57:00] Reliably Transmitting (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep  3 10:57:01] Retransmitting #1 (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep  3 10:57:02] Retransmitting #2 (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep  3 10:57:04] Retransmitting #3 (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep  3 10:57:08] Retransmitting #4 (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep  3 10:57:16] Retransmitting #5 (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep  3 10:57:32] Retransmitting #6 (no NAT) to 192.168.216.7:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep  3 10:57:33]
<--- SIP read from UDP://192.168.216.7:5060 --->
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.216.7:5060;branch=z9hG4bKdc648d9854c70686f
Max-Forwards: 70
From: <sip:00404xxx@sn4638-ha1>;tag=2210277577
To: "xxx" <sip:[email protected]>;tag=as41f51440
Call-ID: [email protected]
CSeq: 18011 BYE
User-Agent: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


<------------->
[Sep  3 10:57:33] --- (9 headers 0 lines) ---
[Sep  3 10:57:33] Sending to 192.168.216.7 : 5060 (no NAT)
[Sep  3 10:57:33]
<--- Transmitting (no NAT) to 192.168.216.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.216.7:5060;branch=z9hG4bKdc648d9854c70686f;received=192.168.216.7
From: <sip:00404xxx@sn4638-ha1>;tag=2210277577
To: "xxx" <sip:[email protected]>;tag=as41f51440
Call-ID: [email protected]
CSeq: 18011 BYE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
[Sep  3 10:57:33]   == Spawn extension (trunk-ha, 00404xxx, 10011) exited non-zero on 'SIP/74486-008cc1d8'
[Sep  3 10:57:33] Really destroying SIP dialog '[email protected]' Method: BYE

Und von der Patton Seite aus:

Code:

SN4638-HA1#
SN4638-HA1#08:56:46  ISDN  > #1186 p: 0 S: sapi: 0 cr=0 ea=0 tei:  0 ea=1 RR   Nr(106)         pf=1
08:56:46  ISDN  > #1187 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(  1)         pf=1
08:56:46  ISDN  > #1188 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(106)         pf=1
08:56:46  ISDN  > #1189 p: 0 R: sapi: 0 cr=0 ea=0 tei:  0 ea=1 RR   Nr(  1)         pf=1
08:56:47  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:00404xxx@sn4638-ha1 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Date: Thu, 03 Sep 2009 08:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 395895816 395895816 IN IP4 192.168.216.5
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.216.5
t=0 0
m=audio 12484 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

08:56:47  SIP_SI> [STACK] Forward INVITE request to GW_SIP_HA-IF_IP_LAN_HA
08:56:47  SIP_SI> [GW GW_SIP_HA] Received INVITE request
08:56:47  SIP_SI> [PR IF_VOIP_GATEWAY_HA] Added endpoint IF_VOIP_GATEWAY_HA-00ca88d0
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > LocCp: Build Local Capabilities merging Configured/Peer Capabilities
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0]          Configured Codecs:       Voice: G.711 A-law[20/20], Clear Mode[20/20] / Fax: T.38 UDP[1/1][rel]
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0]          Peer Codecs:             N/A
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0]          Resulting Local Codecs:  Voice: G.711 A-law[20/20], Clear Mode[20/20] / Fax: T.38 UDP[1/1][rel]
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > RemCp: Build Remote Capabilities merging Configured/Remote Capabilities
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0]          Configured Codecs:       Voice: G.711 A-law[20/20], Clear Mode[20/20] / Fax: T.38 UDP[1/1][rel]
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0]          Remote Codecs:           N/A
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0]          Resulting Remote Codecs: Voice: G.711 A-law[20/20], Clear Mode[20/20] / Fax: T.38 UDP[1/1][rel]
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] Creating a new session
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] Adding a new session
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] < Stack: INVITE <sip:00404xxx@sn4638-ha1>
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0]          From:                "xxx" <sip:[email protected]>
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0]          To:                  <sip:00404xxx@sn4638-ha1>
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0]          P-Asserted-Identity:
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] < SM   : Event 'Stk-Invited' (State 'Idle')
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] < SM   : Event 'RX-Invite' (State 'Idle')
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] > SM   : Action 'Negotiate-Received-Offer'
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] < Stack: Negotiating offer
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] > SM   : New state 'Final-Offer-Received'
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] > SM   : Action 'Publish-Remote-Capabilities'
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > RemCp: Build Remote Capabilities merging Configured/Remote Capabilities
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0]          Configured Codecs:       Voice: G.711 A-law[20/20], Clear Mode[20/20] / Fax: T.38 UDP[1/1][rel]
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xca88d0 SDP] DTMF over RTP enabled.
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0]          Remote Codecs:           Voice: G.711 A-law[any/any][ss]
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0]          Resulting Remote Codecs: Voice: G.711 A-law[20/20]
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : New state 'Establishing-Incoming'
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : Action 'App-Offered'
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < SM   : Event 'Active-Session-Offered' (Active State 'Idle', Passive State 'Idle')
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > SM   : New active state 'Peer-Trying' (Code: 0)
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > SM   : Action 'Call-Control-Dial'
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > RemCp: Build Remote Capabilities merging Configured/Remote Capabilities
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0]          Configured Codecs:       Voice: G.711 A-law[20/20], Clear Mode[20/20] / Fax: T.38 UDP[1/1][rel]
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xca88d0 SDP] DTMF over RTP enabled.
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0]          Remote Codecs:           Voice: G.711 A-law[any/any][ss]
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0]          Resulting Remote Codecs: Voice: G.711 A-law[20/20]
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Network -> GW_SIP_HA/192.168.216.7
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: E164-Number -> 486
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: URI -> sip:[email protected]
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Type-Of-Number -> Unknown
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Numbering-Plan -> Unknown
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Presentation-Indicator -> Presentation allowed
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Name -> "xxx"
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Screening-Indicator -> User provided, not screened
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Supports Overlap-Sending -> true
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Supported Codecs -> Voice: G.711 A-law[20/20]
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Codec Negotiating -> Idle
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Unique Identifier -> [email protected]
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: IP-Address -> 192.168.216.5
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Call-Leg-ID -> 0x00cab470
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: State -> CONNECTED
08:56:47  CC    > [Call 00cae348] Set call property: Context -> 0x00000022
08:56:47  CC    > [Call 00cae348] Set call property: Information-Transfer-Capability -> 3.1kHz Audio
08:56:47  CC    > [Call 00cae348] Set call property: Hops -> 0x00000010
08:56:47  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Dial to provider router (IF_VOIP_GATEWAY_HA-precall-service) using call 00cae348
08:56:47  CC    > [EP router-00caed90/incoming] Accept call 00cae348
08:56:47  CC    > [EP router-00caed90/incoming] Set call-leg property: E164-Number -> 00404xxx
08:56:47  CC    > [EP router-00caed90/incoming] Set call-leg property: Type-Of-Number -> Unknown
08:56:47  CC    > [EP router-00caed90/incoming] Set call-leg property: Numbering-Plan -> Unknown
08:56:47  CC    > [EP router-00caed90/incoming] Set call-leg property: Name ->
08:56:47  CC    > [EP router-00caed90/incoming] Set call-leg property: Alert-Info ->
08:56:47  CC    > [EP router-00caed90/incoming] Set call-leg property: URI -> sip:00404xxx@sn4638-ha1
08:56:47  CC    > [EP router-00caed90/incoming] Set call-leg property: Network -> router
08:56:47  CC    > [EP router-00caed90/incoming] Set call-leg property: Call-Leg-ID -> 0x00caf5a0
08:56:47  CC    > [EP router-00caed90/incoming] Set call-leg property: State -> TRYING
08:56:47  CC    > [EP router-00caed90] Start route-lookup
08:56:47  CR    > [switch] Routing-Lookup:
08:56:47  CR    >   Execute all entries in table IF_VOIP_GATEWAY_HA-precall-service
08:56:47  CR    >   Execute all entries in table HG_2_PBX-dest
08:56:47  CR    >   Execute all entries in table route-found-place-call
08:56:47  CR    >   Lookup result: Route found; place call (timeout=0)
08:56:47  CC    > [EP router-00caed90] Route found; immediately place call
08:56:47  CC    > [EP router-00caed90] Route to provider 'HG_2_PBX'
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: E164-Number -> 486
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: URI -> sip:[email protected]
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: Type-Of-Number -> Unknown
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: Numbering-Plan -> Unknown
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: Presentation-Indicator -> Presentation allowed
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: Name -> "xxx"
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: Screening-Indicator -> User provided, not screened
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: Supports Overlap-Sending -> true
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: Supported Codecs -> Voice: G.711 A-law[20/20]
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: Codec Negotiating -> Idle
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: Unique Identifier -> [email protected]
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: IP-Address -> 192.168.216.5
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: Network -> router
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: Call-Leg-ID -> 0x00cabc38
08:56:47  CC    > [EP router-00caed90/outgoing] Set call-leg property: State -> CONNECTED
08:56:47  CC    > [Call 00cabb40] Set call property: Context -> 0x00000022
08:56:47  CC    > [Call 00cabb40] Set call property: Information-Transfer-Capability -> 3.1kHz Audio
08:56:47  CC    > [Call 00cabb40] Set call property: Hops -> 0x0000000f
08:56:47  CC    > [EP router-00caed90/outgoing] Dial to provider HG_2_PBX () using call 00cabb40
08:56:47  CC    > [EP HG_2_PBX-00caba80/incoming] Accept call 00cabb40
08:56:47  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: E164-Number -> 00404xxx
08:56:47  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: Type-Of-Number -> Unknown
08:56:47  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: Numbering-Plan -> Unknown
08:56:47  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: Name ->
08:56:47  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: Alert-Info ->
08:56:47  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: URI -> sip:00404xxx@sn4638-ha1
08:56:47  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: Network -> HG_2_PBX
08:56:47  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: Call-Leg-ID -> 0x00cac478
08:56:47  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: State -> TRYING
08:56:47  CC    > [EP HG_2_PBX-00caba80] Hunt to IF_S0_00 ()
08:56:47  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: Allows Push-Back -> false
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: E164-Number -> 486
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: URI -> sip:[email protected]
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Type-Of-Number -> Unknown
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Numbering-Plan -> Unknown
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Presentation-Indicator -> Presentation allowed
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Name -> "xxx"
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Screening-Indicator -> User provided, not screened
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Supports Overlap-Sending -> true
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Supported Codecs -> Voice: G.711 A-law[20/20]
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Codec Negotiating -> Idle
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Unique Identifier -> [email protected]
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: IP-Address -> 192.168.216.5
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Allows Push-Back -> false
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Network -> HG_2_PBX
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Call-Leg-ID -> 0x00cad8d8
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: State -> CONNECTED
08:56:47  CC    > [Call 00cad810] Set call property: Context -> 0x00000022
08:56:47  CC    > [Call 00cad810] Set call property: Information-Transfer-Capability -> 3.1kHz Audio
08:56:47  CC    > [Call 00cad810] Set call property: Hops -> 0x0000000e
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Dial to provider IF_S0_00 () using call 00cad810
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Accept call 00cad810
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: E164-Number -> 00404xxx
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Type-Of-Number -> Unknown
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Numbering-Plan -> Unknown
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Name ->
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Alert-Info ->
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: URI -> sip:00404xxx@sn4638-ha1
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Quality-Of-Service -> MOS 4.50, DS0
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Network -> IF_S0_00
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Call-Leg-ID -> 0x00cb3118
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: State -> TRYING
08:56:47  ISDN  > #1190 p: 0 from CC key: 8388623 L3SetupReq
08:56:47  ISDN  > #1191      Service:Transparent cing: sub: ced: sub:
08:56:47  ISDN  > #1192 cause : Transparent
08:56:47  ISDN  > #1193      hex:04 03 90 90 A3 1E 02 85 83 6C 05 00 80 34 38 36 70 0D 80 30 30 34 30 34 33 32 30 34 34 38 36
08:56:47  ISDN  > #1194      IE: Bearer
08:56:47  ISDN  > #1195      IE: ProgressInd
08:56:47  ISDN  > #1196      IE: CallingPartyNbr  486
08:56:47  ISDN  > #1197      IE: CalledPartyNbr  00404xxx
08:56:47  ISDN  > #1198      *newPc BR-U, tei:  0, sapi:  0
08:56:47  ISDN  > #1199       key: 8388623, Stack:2V01b
08:56:47  ISDN  > #1200       EuroISDN
08:56:47  ISDN  > #1201      ---> Layer2  tei:  0 sapi:  0 Setup
08:56:47  ISDN  > #1202      hex:08 01 0F 05 04 03 90 90 A3 1E 02 85 83 6C 05 00 80 34 38 36 70 0D 80 30 30 34 30 34 33 32 30 34 34
08:56:47  ISDN  > #1203      IE: Bearer
08:56:47  ISDN  > #1204      IE: ProgressInd
08:56:47  ISDN  > #1205      IE: CallingPartyNbr  486
08:56:47  ISDN  > #1206      IE: CalledPartyNbr  00404xxx
08:56:47  ISDN  > #1207      new call state : L3PcSt01U
08:56:47  ISDN  > #1208 p: 0 S: sapi: 0 cr=0 ea=0 tei:  0 ea=1 INFO Nr(106) Ns(  1) pf=0
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Endpoint-Is-Isdn -> true
08:56:47  CC    > [Call 00cae348] Set call property: Hops -> 0x0000000f
08:56:47  ISDN  > #1209 p: 0 R: sapi: 0 cr=0 ea=0 tei:  0 ea=1 RR   Nr(  2)         pf=0
08:56:47  SIP_TR> [STACK] > Stack: to 192.168.216.5
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport=5060;received=192.168.216.5
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < Peer : Datapath context: 00000022
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < Peer : Inband info: no
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < Peer : State: TRYING
08:56:47  CC    > [EP router-00caed90/incoming] Set call-leg property: Allows Push-Back -> false
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Network -> router
08:56:47  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Call-Leg-ID -> 0x00cabc38
08:56:47  ISDN  > #1210 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr(  2) Ns(106) pf=0
08:56:47  ISDN  > #1211 p: 0 <--- Layer2  tei:  0 sapi:  0 SetupAck
08:56:47  ISDN  > #1212      hex:08 01 8F 0D 18 01 89
08:56:47  ISDN  > #1213      IE: ChannelId  BCh0 excl  otherIf
08:56:47  ISDN  > #1214      ALLOC bchan BCh0 cref : 0x0f
08:56:47  ISDN  > #1215      to  CC key: 8388623 L3MoreInfoInd
08:56:47  ISDN  > #1216      new call state : L3PcSt02U
08:56:47  ISDN  > #1217 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(107)         pf=0
08:56:47  DP    > TDM-00/00/00: registerEventCallback(0xa2d484)
08:56:47  DP    > TDM-00/00/00: addToContext(00000022)
08:56:47  DP    > TDM-00/00/00: configure(Pstn-Config)
08:56:47  DP    > TDM-00/00/00: The following config package could not be converted:
SEQUENCE {
  packageId UUIDpackageId UUID = CfgToneDetection
  argument SEQUENCE {
    profile GeneralString IMPLICIT [CONTEXT 1] = 'default' OPTIONAL
    detectDtmf BOOLEAN IMPLICIT [CONTEXT 2] = false OPTIONAL
  }
}
08:56:47  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: State -> ADDRESS-INCOMPLETE
08:56:47  DP    > TDM-00/00/00: setOperationMode(send/receive)
08:56:47  TDM   > TDM-00/00/00: Acquiring DSP resource.
08:56:47  TDM   > TDM-00/00/00: Reconfigure resource.
08:56:47  TDM   > [DSP 0xe5af58] Received RTP Media Config.
08:56:47  TDM   > [DSP 0xe5af58] voice config.
08:56:47  TDM   > [DSP 0xe5af58] dejitter config.
08:56:47  TDM   > [DSP 0xe5af58] pstn config.
08:56:47  TDM   > [DSP 0xe5af58] caller-id config (None).
08:56:47  TDM   > [DSP 0xe5af58] Admin operation mode inactive -> send/receive
08:56:47  TDM   > [DSP 0xe5af58] Dsp operation mode inactive -> send/receive
08:56:47  TDM   > [DSP 0xe5af58]: Connected port 0 bChannel 0 to DSP timeslot 0
08:56:47  DSP   > [00000000][160185970] State=closed: Event=openVoice
08:56:47  DSP   > [00000000][160185970] Set tx plugin to voice.
08:56:47  DSP   > [00000000][160185970] Opening
08:56:47  DSP   > [00000000][160185970] Dsp 0 Channel 0 configured for G.711aLaw on timeslot 0:
    Silence compression OFF
    Echo canceller ON (NLPM Adaptive, HybridLoss 6)
    Post filter: ON, HighPass filter: ON
    Output gain: 0dB
    Input gain:  0dB
    Transmission: Fax None / Modem: None
    Max bit rate (relay): Fax 14400 bit/s / Modem 9600 bit/s
    Fax/modem gain (relay): -9.5dB
    Fax/modem bypass codec: G.711aLaw
    Modem Dejitter buffer size: 200ms
    Fax Dejitter buffer size: 200ms
    T.38 Error correction: ON
    T.38 HDLC image tx: ON
    Fax protocol mode: T.38 UDP
    DTMF signal gain: lf -2dB, hf -1dB, mute encoder OFF
    Fax detection forced: OFF
    Caller-ID: Disabled

08:56:47  DSP   > [Channel 0 0] ACTIVATING CHANNEL
08:56:47  DSP   > [00000000][160185970] Set tx media type 2.
08:56:47  DSP   > [00000000][160185970] Set rx media type 2.
08:56:47  DSP   > [00000000][160185970] New state=voice
08:56:47  TDM   > [DSP 0xe5af58] Activate VOICE dejitter configuration
08:56:47  Dejit > Input length: 80
08:56:47  Dejit > [0xe5b230] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

08:56:47  Dejit > Input length: 80
08:56:47  Dejit > [0xe5b6e0] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

08:56:47  Dejit > Input length: 80
08:56:47  Dejit > [0xe5b9a0] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

08:56:47  TDM   > [DSP 0xe5af58] Activate VOICE tx-buffer configuration
08:56:47  TDM   > [DSP 0xe5af58] Activating TX.
08:56:47  TDM   > [DSP 0xe5af58] Activating RX.
08:56:47  DSP   > Scheduler: added source processor 0xe5b3ac
08:56:47  CC    > [EP HG_2_PBX-00caba80] Hunting tentatively succeeded
08:56:47  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: State -> ADDRESS-INCOMPLETE
08:56:47  CC    > [EP router-00caed90/incoming] Set call-leg property: State -> ADDRESS-INCOMPLETE
08:56:47  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < Peer : State: ADDRESS-INCOMPLETE
08:56:47  DSP   > [Channel 0 0] FIRST_PACKET_RECEIVED
08:56:51  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Quality-Of-Service -> MOS 4.40, RTP, (undefined) (0ms), Local: Rx 0 pkts, 0 bytes, 0 lost, jitter 0 ms, Tx 0 pkts, 0 bytes, rtt 0 ms, Remote: Rx 0 pkts, 0 bytes, 0 lost, jitter 0 ms, Tx 0 pkts, 0 bytes, rtt 0 ms
08:56:51  CC    > [EP router-00caed90/outgoing] Set call-leg property: Quality-Of-Service -> MOS 4.40, RTP, (undefined) (0ms), Local: Rx 0 pkts, 0 bytes, 0 lost, jitter 0 ms, Tx 0 pkts, 0 bytes, rtt 0 ms, Remote: Rx 0 pkts, 0 bytes, 0 lost, jitter 0 ms, Tx 0 pkts, 0 bytes, rtt 0 ms
08:56:51  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Quality-Of-Service -> MOS 4.40, RTP, (undefined) (0ms), Local: Rx 0 pkts, 0 bytes, 0 lost, jitter 0 ms, Tx 0 pkts, 0 bytes, rtt 0 ms, Remote: Rx 0 pkts, 0 bytes, 0 lost, jitter 0 ms, Tx 0 pkts, 0 bytes, rtt 0 ms
08:56:52  ISDN  > #1218 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr(  2) Ns(107) pf=0
08:56:52  ISDN  > #1219 p: 0 <--- Layer2  tei:  0 sapi:  0 CallProc
08:56:52  ISDN  > #1220      hex:08 01 8F 02
08:56:52  ISDN  > #1221      to  CC key: 8388623 L3ProceedingInd
08:56:52  ISDN  > #1222      new call state : L3PcSt03U
08:56:52  ISDN  > #1223 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(108)         pf=0
08:56:52  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: State -> PROCEEDING
08:56:52  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: State -> PROCEEDING
08:56:52  CC    > [EP router-00caed90/incoming] Set call-leg property: State -> PROCEEDING
08:56:52  ISDN  > #1224 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr(  2) Ns(108) pf=0
08:56:52  ISDN  > #1225 p: 0 <--- Layer2  tei:  0 sapi:  0 Alerting
08:56:52  ISDN  > #1226      hex:08 01 8F 01 1E 02 81 88
08:56:52  ISDN  > #1227      IE: ProgressInd
08:56:52  ISDN  > #1228      to  CC key: 8388623 L3AlertingInd
08:56:52  ISDN  > #1229      new call state : L3PcSt04U
08:56:52  ISDN  > #1230 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(109)         pf=0
08:56:52  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: State -> ALERTING
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < Peer : State: PROCEEDING
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < SM   : Event 'CC-Proceeding' (Active State 'Peer-Trying', Passive State 'Idle')
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > SM   : New active state 'Peer-Proceeding' (Code: 0)
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > SM   : Action 'Session-Proceeding'
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] < SM   : Event 'App-Proceeding' (State 'Establishing-Incoming')
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : Action 'Stk-Status(183)'
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] < SM   : Event 'TX-Info-NoSDP' (State 'Final-Offer-Received')
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > Stack: 183 Session Progress
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Update packet for destination: 192.168.216.5:5060
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] ---------------------------------------------------
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Traffic-Class for ID    : sip:[email protected]
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Traffic-Class           : local-default
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Update invite transaction timeout     : 32
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Update non-invite transaction timeout : 32
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Contact Header          : 192.168.216.7:5060
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Local UDP Address       : 192.168.216.7:5060
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Datapath Controllers    : 192.168.216.7
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : New state 'Establishing-Incoming-Proceeding'
08:56:52  SIP_TR> [STACK] > Stack: to 192.168.216.5
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport=5060;received=192.168.216.5
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


08:56:52  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: State -> ALERTING
08:56:52  CC    > [EP router-00caed90] Routing succeeded
08:56:52  CC    > [EP router-00caed90/incoming] Transfer call 00cabb40 to 00cae348 ==> conference
08:56:52  CC    > [EP router-00caed90/incoming] Drop call 00cae348
08:56:52  CC    > [EP router-00caed90/incoming] Set call-leg property: Cause -> Normal call clearing
08:56:52  CC    > [EP router-00caed90/incoming] Set call-leg property: State -> RELEASED
08:56:52  CC    > [EP router-00caed90/outgoing] Drop call 00cabb40
08:56:52  CC    > [EP router-00caed90/outgoing] Set call-leg property: Cause -> Normal call clearing
08:56:52  CC    > [EP router-00caed90/outgoing] Set call-leg property: State -> RELEASED
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < Peer : State: ALERTING
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < SM   : Event 'CC-Alerting' (Active State 'Peer-Proceeding', Passive State 'Idle')
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > SM   : New active state 'Peer-Alerting' (Code: 0)
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > SM   : Action 'Session-Alerting'
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] < SM   : Event 'App-Alerting' (State 'Establishing-Incoming-Proceeding')
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : Action 'Stk-Status(180)'
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] < SM   : Event 'TX-Info-NoSDP' (State 'Final-Offer-Received')
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > Stack: 180 Ringing
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Update packet for destination: 192.168.216.5:5060
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] ---------------------------------------------------
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Traffic-Class           : local-default
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Update invite transaction timeout     : 32
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Update non-invite transaction timeout : 32
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Contact Header          : 192.168.216.7:5060
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Local UDP Address       : 192.168.216.7:5060
08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Datapath Controllers    : 192.168.216.7
08:56:52  SIP_TR> [STACK] > Stack: to 192.168.216.5
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport=5060;received=192.168.216.5
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


08:56:52  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : New state 'Establishing-Incoming-Alerting'
08:56:52  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Network -> GW_SIP_HA/192.168.216.7
08:56:52  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Call-Leg-ID -> 0x00cab470
08:56:59  ISDN  > #1231 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr(  2) Ns(109) pf=0
08:56:59  ISDN  > #1232 p: 0 <--- Layer2  tei:  0 sapi:  0 Connect
08:56:59  ISDN  > #1233      hex:08 01 8F 07 29 05 09 09 03 0A 39 4C 0D 00 81 30 34 30 34 33 32 30 34 34 38 36
08:56:59  ISDN  > #1234      IE: DateTime
08:56:59  ISDN  > #1235      IE: ConnectedNbr  0404xxx
08:56:59  ISDN  > #1236      to  CC key: 8388623 L3SetupConf
08:56:59  ISDN  > #1237      Connects bchan BCh0  cref : 0x0f
08:56:59  ISDN  > #1238      ---> Layer2  tei:  0 sapi:  0 ConnectAck
08:56:59  ISDN  > #1239      hex:08 01 0F 0F
08:56:59  ISDN  > #1240      new call state : L3PcSt10U
08:56:59  ISDN  > #1241 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(110)         pf=0
08:56:59  ISDN  > #1242 p: 0 S: sapi: 0 cr=0 ea=0 tei:  0 ea=1 INFO Nr(110) Ns(  2) pf=0
08:56:59  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Screening-Indicator -> User provided, verified and passed
08:56:59  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Presentation-Indicator -> Presentation allowed
08:56:59  ISDN  > #1243 p: 0 R: sapi: 0 cr=0 ea=0 tei:  0 ea=1 RR   Nr(  3)         pf=0
08:56:59  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: E164-Number -> 0404xxx
08:56:59  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: State -> CONNECTED
08:56:59  DP    > TDM-00/00/00: The following config package could not be converted:
SEQUENCE {
  packageId UUIDpackageId UUID = CfgToneDetection
  argument SEQUENCE {
    profile GeneralString IMPLICIT [CONTEXT 1] = 'default' OPTIONAL
    detectDtmf BOOLEAN IMPLICIT [CONTEXT 2] = true OPTIONAL
  }
}
08:56:59  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Provides Data -> true
08:56:59  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: Provides Data -> true
08:56:59  CC    > [EP HG_2_PBX-00caba80] Hunting definitely succeeded
08:56:59  CC    > [EP HG_2_PBX-00caba80/incoming] Transfer call 00cad810 to 00cae348 ==> conference
08:56:59  CC    > [EP HG_2_PBX-00caba80/incoming] Drop call 00cae348
08:56:59  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: Provides Data -> false
08:56:59  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: Cause -> Normal call clearing
08:56:59  CC    > [EP HG_2_PBX-00caba80/incoming] Set call-leg property: State -> RELEASED
08:56:59  CC    > [EP HG_2_PBX-00caba80/outgoing] Drop call 00cad810
08:56:59  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: Cause -> Normal call clearing
08:56:59  CC    > [EP HG_2_PBX-00caba80/outgoing] Set call-leg property: State -> RELEASED
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < Peer : Inband info: yes
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < Peer : State: CONNECTED
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < SM   : Event 'CC-Connected' (Active State 'Peer-Alerting', Passive State 'Idle')
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > SM   : New active state 'Connected' (Code: 0)
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > SM   : Action 'Session-Connect'
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] < SM   : Event 'App-Connect' (State 'Establishing-Incoming-Alerting')
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : Action 'Stk-Status(200)'
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] < SM   : Event 'TX-Success' (State 'Final-Offer-Received')
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] > SM   : Action 'Create-Answer'
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] > Stack: Answer
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] > SM   : New state 'Negotiated'
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] > SM   : Action 'Update-Datapath'
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] Stream 0: Added
08:56:59  DP    > RTP-00/0022: registerEventCallback(0xa18a5c)
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] Created datapath controller: Audio
08:56:59  DP    > RTP-00/0022: configure(Voice-Config)
08:56:59  DP    > RTP-00/0022: event(General/ConfigChanged-Event)
08:56:59  DP    > RTP-00/0022: configure(Dejitter-Config)
08:56:59  DP    > RTP-00/0022: event(General/ConfigChanged-Event)
08:56:59  DP    > RTP-00/0022: addToContext(00000022)
08:56:59  DP    > RTP-00/0022: addConnector(ffffffff)
08:56:59  DP    > TDM-00/00/00: addConnector(ffffffff)
08:56:59  TDM   > [DSP 0xe5af58] Add connector notify. ID 0
08:56:59  DSP   > Scheduler: removed source processor 0xe5b3ac
08:56:59  DSP   > Scheduler: added source processor 0xe5b3ac
08:56:59  DP    > RTP-00/0022: getConfiguration(00010300)
08:56:59  DP    > RTP-00/0022: getConfiguration(00010b00)
08:56:59  DP    > TDM-00/00/00: configure(Voice-Config)
08:56:59  TDM   > [DSP 0xe5af58] voice config.
08:56:59  TDM   > [DSP 0xe5af58] reconfiguring DSP.
08:56:59  DSP   > [00000000][160197880] State=voice: Event=close
08:56:59  DSP   > [00000000][160197880] Set tx plugin to none.
08:56:59  DSP   > [00000000][160197880] Set tx media type 2.
08:56:59  DSP   > [00000000][160197880] Set rx media type 2.
08:56:59  DSP   > [00000000][160197880] Closing (dsp 0, channel 0)
08:56:59  DSP   > [Channel 0 0] SENDING IDLE PACKET
08:56:59  DSP   > [00000000][160197880] New state=closed
08:56:59  DSP   > [00000000][160197880] State=closed: Event=openVoice
08:56:59  DSP   > [00000000][160197880] Set tx plugin to voice.
08:56:59  DSP   > [00000000][160197880] Opening
08:56:59  DSP   > [00000000][160197880] Dsp 0 Channel 0 configured for G.711aLaw on timeslot 0:
    Silence compression OFF
    Echo canceller ON (NLPM Adaptive, HybridLoss 6)
    Post filter: ON, HighPass filter: ON
    Output gain: 0dB
    Input gain:  0dB
    Transmission: Fax Relay / Modem: None
    Max bit rate (relay): Fax 14400 bit/s / Modem 9600 bit/s
    Fax/modem gain (relay): -9.5dB
    Fax/modem bypass codec: G.711aLaw
    Modem Dejitter buffer size: 200ms
    Fax Dejitter buffer size: 200ms
    T.38 Error correction: ON
    T.38 HDLC image tx: ON
    Fax protocol mode: T.38 UDP
    DTMF signal gain: lf -2dB, hf -1dB, mute encoder ON
    Fax detection forced: OFF
    Caller-ID: Disabled

08:56:59  DSP   > [00000000][160197880] Set tx media type 2.
08:56:59  DSP   > [00000000][160197880] Set rx media type 2.
08:56:59  DSP   > [00000000][160197880] New state=voice
08:56:59  TDM   > [DSP 0xe5af58] Activate VOICE dejitter configuration
08:56:59  Dejit > Input length: 80
08:56:59  Dejit > [0xe5b230] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

08:56:59  Dejit > Input length: 80
08:56:59  Dejit > [0xe5b6e0] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

08:56:59  Dejit > Input length: 80
08:56:59  Dejit > [0xe5b9a0] Reinitialized dejitter buffer:
    mode                : static
    max delay           : 40
    max queue fill level: 4
    average fill level  : 2

08:56:59  TDM   > [DSP 0xe5af58] Activate VOICE tx-buffer configuration
08:56:59  DP    > RTP-00/0022: getConfiguration(00010e00)
08:56:59  DP    > TDM-00/00/00: configure(Dejitter-Config)
08:56:59  TDM   > [DSP 0xe5af58] dejitter config.
08:56:59  TDM   > [DSP 0xe5af58] Reconfiguring dejitter.
08:56:59  TDM   > [DSP 0xe5af58] Activate VOICE dejitter configuration
08:56:59  Dejit > Input length: 80
08:56:59  Dejit > [0xe5b230] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 6

08:56:59  Dejit > Input length: 80
08:56:59  Dejit > [0xe5b6e0] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 6

08:56:59  Dejit > Input length: 80
08:56:59  Dejit > [0xe5b9a0] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 6

08:56:59  DP    > RTP-00/0022: registerEventCallback(0xda5a18)
08:56:59  DP    > TDM-00/00/00: registerEventCallback(0xda5a18)
08:56:59  DSP   > [Channel 0 0] IDLE_PACKET_RECEIVED
08:56:59  DSP   > [Channel 0 0] ACTIVATING CHANNEL
08:56:59  DP    > RTP-00/0022: configure(RTP-Config)
08:56:59  RTP   > [02000022] Configure local source:  0xc0a8d807/4932/13322368
08:56:59  RTP   > [02000022] Configure remote source: 0xc0a8d805/12484/14949080
08:56:59  RTP   > [02000022] Next hop gateway is:     0x00000000
08:56:59  RTP   > [TERM 2000022] Config changed (codec=G.711 A-law | media=audio)
08:56:59  DP    > RTP-00/0022: event(General/ConfigChanged-Event)
08:56:59  DP    > RTP-00/0022: getConfiguration(00010300)
08:56:59  DP    > TDM-00/00/00: configure(MediaType/RTP-Config)
08:56:59  TDM   > TDM-00/00/00: Received Media Config.
08:56:59  TDM   > TDM-00/00/00: Resource can be re-used.
08:56:59  TDM   > [DSP 0xe5af58] Received RTP Media Config.
08:56:59  DSP   > [00000000][160197900] Try codec update for G.711aLaw.
08:56:59  TDM   > [DSP 0xe5af58] Reconfiguring dejitter.
08:56:59  TDM   > [DSP 0xe5af58] Activate VOICE dejitter configuration
08:56:59  Dejit > Input length: 80
08:56:59  Dejit > [0xe5b230] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 6

08:56:59  Dejit > Input length: 80
08:56:59  Dejit > [0xe5b6e0] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 6

08:56:59  Dejit > Input length: 80
08:56:59  Dejit > [0xe5b9a0] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 6

08:56:59  TDM   > [DSP 0xe5af58] Reconfiguring tx-buffer.
08:56:59  TDM   > [DSP 0xe5af58] Activate VOICE tx-buffer configuration
08:56:59  DP    > RTP-00/0022: setOperationMode(send/receive)
08:56:59  RTP   > [02000022] Set mode: INACTIVE -> TX/RX
08:56:59  DPMUX > Changed next processor of port 0x0022face: 0xe41e98
08:56:59  RTP   > [02000022] (BCD) Event=enable | New State=broken
08:56:59  TDM   > DATA_BUFF_TX: Update Connection. Processor: e41f58
08:56:59  DPMUX > Activating directpath port: 0x0022face
08:56:59  DPMUX >   Protocol:                 RTP
08:56:59  DPMUX >   Local transport address:  0xc0a8d807/4932 (2 port(s))
08:56:59  DPMUX >   Remote transport address: 0xc0a8d805/12484 (2 port(s))
08:56:59  DPMUX >   IP Interface:             3
08:56:59  RTP   > [02000022] Next hop gateway is:     0xc0a8d805
08:56:59  NTE   > [00e42370] Tx activation request.
08:56:59  DSP   > [Channel 0 0] FIRST_PACKET_RECEIVED
08:56:59  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Provides Data -> true
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > Stack: 200 OK
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Update packet for destination: 192.168.216.5:5060
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] ---------------------------------------------------
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Traffic-Class           : local-default
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Update invite transaction timeout     : 32
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Update non-invite transaction timeout : 32
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Contact Header          : 192.168.216.7:5060
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating SDP Origin              : 192.168.216.7
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating SDP Contact             : 192.168.216.7
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Local UDP Address       : 192.168.216.7:5060
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Datapath Controllers    : 192.168.216.7
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : New state 'Establishing-Incoming-OK'
08:56:59  SIP_TR> [STACK] > Stack: to 192.168.216.5
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport=5060;received=192.168.216.5
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Server: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Supported: replaces
Content-Type: application/sdp
Content-Length: 197

v=0
o=MxSIP 0 59 IN IP4 192.168.216.7
s=SIP Call
c=IN IP4 192.168.216.7
t=0 0
m=audio 4932 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

08:56:59  SIP_TR> [STACK] < Stack: from 192.168.216.5
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK799ef6e3;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.1
Content-Length: 0


08:56:59  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] < Stack: ACK
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] < SM   : Event 'Stk-Acked' (State 'Establishing-Incoming-OK')
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] < SM   : Event 'RX-Ack' (State 'Negotiated')
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : New state 'Established'
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : Action 'App-Connected'
08:56:59  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < SM   : Event 'Active-Session-Connected' (Active State 'Connected', Passive State 'Idle')
08:57:00  DP    > RTP-00/0022: event(Event/Mediatype-RTP/Connection-Established)
08:57:00  RTP   > [02000022] (BCD) Event=rx-rtp | New State=established
08:57:00  RTP   > [02000022] Rx first packet (seq=29533)
08:57:00  Dejit > Input length: 160
08:57:00  Dejit > [0xe5b230] Reinitialized dejitter buffer:
    mode                : adaptive
    max delay           : 60
    max queue fill level: 3

08:57:00  MEDIA > [00000000][160198250] Codec=G711A | Media=VOICE | Ecan=ON | Vad=OFF
08:57:00  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] Connection Established
08:57:00  Dejit > [0xe5b230] underrun phase detected, freezed packet measurements
08:57:00  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

08:57:01  DP    > RTP-00/0022: event(Event/Mediatype-RTP/Connection-Broken)
08:57:01  RTP   > [02000022] (BCD) Event=timeout | New State=broken
08:57:01  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] Connection Broken
08:57:01  RTP   > [02000022] RTCP TX (SR) Timestamp=160199360
08:57:01  RTP   > [02000022]  Local TX-Info: packets=71, octets=11360
08:57:01  RTP   > [02000022]  Local RX-Info: packets=5, octets=800, lost=0, jitter=17, since last=0mS
08:57:01  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

08:57:01  DP    > RTP-00/0022: getStatistics(00020900)
08:57:01  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Quality-Of-Service -> MOS 2.58, RTP, G.711 A-law (20ms), Local: Rx 5 pkts, 800 bytes, 0 lost, jitter 273 ms, Tx 103 pkts, 16480 bytes, rtt 0 ms, Remote: Rx 0 pkts, 0 bytes, 0 lost, jitter 0 ms, Tx 0 pkts, 0 bytes, rtt 0 ms
08:57:03  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

08:57:04  RTP   > [02000022] RTCP RX (SR) Timestamp=160202980
08:57:04  RTP   > [02000022]  Remote processing delay=3621ms
08:57:04  RTP   > [02000022]  RTT 0ms
08:57:04  RTP   > [02000022]  Remote TX-Info: packets=5, octets=800
08:57:04  RTP   > [02000022]  Remote RX-Info: jitter=1, lost=0
08:57:06  RTP   > [02000022] RTCP TX (SR) Timestamp=160204360
08:57:06  RTP   > [02000022]  Local TX-Info: packets=321, octets=51360
08:57:06  RTP   > [02000022]  Local RX-Info: packets=5, octets=800, lost=0, jitter=17, since last=1380mS
08:57:06  DP    > RTP-00/0022: getStatistics(00020900)
08:57:06  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Quality-Of-Service -> MOS 2.58, RTP, G.711 A-law (20ms), Local: Rx 5 pkts, 800 bytes, 0 lost, jitter 273 ms, Tx 353 pkts, 56480 bytes, rtt 0 ms, Remote: Rx 0 pkts, 0 bytes, 0 lost, jitter 1 ms, Tx 5 pkts, 800 bytes, rtt 0 ms
08:57:07  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

08:57:09  ISDN  > #1244 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(  3)         pf=1
08:57:09  ISDN  > #1245 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(110)         pf=1
08:57:09  RTP   > [02000022] RTCP RX (RR)
08:57:09  RTP   > [02000022]  Remote RX-Info: jitter=1, lost=0
08:57:11  RTP   > [02000022] RTCP TX (SR) Timestamp=160209360
08:57:11  RTP   > [02000022]  Local TX-Info: packets=571, octets=91360
08:57:11  RTP   > [02000022]  Local RX-Info: packets=5, octets=800, lost=0, jitter=17, since last=6380mS
08:57:11  DP    > RTP-00/0022: getStatistics(00020900)
08:57:11  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Quality-Of-Service -> MOS 2.58, RTP, G.711 A-law (20ms), Local: Rx 5 pkts, 800 bytes, 0 lost, jitter 273 ms, Tx 603 pkts, 96480 bytes, rtt 0 ms, Remote: Rx 0 pkts, 0 bytes, 0 lost, jitter 1 ms, Tx 5 pkts, 800 bytes, rtt 0 ms
08:57:14  RTP   > [02000022] RTCP RX (RR)
08:57:14  RTP   > [02000022]  Remote RX-Info: jitter=1, lost=0
08:57:15  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

08:57:16  RTP   > [02000022] RTCP TX (SR) Timestamp=160214360
08:57:16  RTP   > [02000022]  Local TX-Info: packets=821, octets=131360
08:57:16  RTP   > [02000022]  Local RX-Info: packets=5, octets=800, lost=0, jitter=17, since last=11380mS
08:57:16  DP    > RTP-00/0022: getStatistics(00020900)
08:57:16  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Quality-Of-Service -> MOS 2.58, RTP, G.711 A-law (20ms), Local: Rx 5 pkts, 800 bytes, 0 lost, jitter 273 ms, Tx 853 pkts, 136480 bytes, rtt 0 ms, Remote: Rx 0 pkts, 0 bytes, 0 lost, jitter 1 ms, Tx 5 pkts, 800 bytes, rtt 0 ms
08:57:19  ISDN  > #1246 p: 0 S: sapi: 0 cr=0 ea=0 tei:  0 ea=1 RR   Nr(110)         pf=1
08:57:19  ISDN  > #1247 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(  3)         pf=1
08:57:19  ISDN  > #1248 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(110)         pf=1
08:57:19  ISDN  > #1249 p: 0 R: sapi: 0 cr=0 ea=0 tei:  0 ea=1 RR   Nr(  3)         pf=1
08:57:19  RTP   > [02000022] RTCP RX (RR)
08:57:19  RTP   > [02000022]  Remote RX-Info: jitter=1, lost=0
08:57:21  RTP   > [02000022] RTCP TX (SR) Timestamp=160219360
08:57:21  RTP   > [02000022]  Local TX-Info: packets=1071, octets=171360
08:57:21  RTP   > [02000022]  Local RX-Info: packets=5, octets=800, lost=0, jitter=17, since last=16380mS
08:57:21  DP    > RTP-00/0022: getStatistics(00020900)
08:57:21  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Quality-Of-Service -> MOS 2.58, RTP, G.711 A-law (20ms), Local: Rx 5 pkts, 800 bytes, 0 lost, jitter 273 ms, Tx 1104 pkts, 176640 bytes, rtt 0 ms, Remote: Rx 0 pkts, 0 bytes, 0 lost, jitter 1 ms, Tx 5 pkts, 800 bytes, rtt 0 ms
08:57:24  RTP   > [02000022] RTCP RX (RR)
08:57:24  RTP   > [02000022]  Remote RX-Info: jitter=1, lost=0
08:57:26  RTP   > [02000022] RTCP TX (SR) Timestamp=160224360
08:57:26  RTP   > [02000022]  Local TX-Info: packets=1321, octets=211360
08:57:26  RTP   > [02000022]  Local RX-Info: packets=5, octets=800, lost=0, jitter=17, since last=21380mS
08:57:26  DP    > RTP-00/0022: getStatistics(00020900)
08:57:26  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Quality-Of-Service -> MOS 2.58, RTP, G.711 A-law (20ms), Local: Rx 5 pkts, 800 bytes, 0 lost, jitter 273 ms, Tx 1354 pkts, 216640 bytes, rtt 0 ms, Remote: Rx 0 pkts, 0 bytes, 0 lost, jitter 1 ms, Tx 5 pkts, 800 bytes, rtt 0 ms
08:57:29  ISDN  > #1250 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(  3)         pf=1
08:57:29  ISDN  > #1251 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(110)         pf=1
08:57:29  RTP   > [02000022] RTCP RX (RR)
08:57:29  RTP   > [02000022]  Remote RX-Info: jitter=1, lost=0
08:57:31  RTP   > [02000022] RTCP TX (SR) Timestamp=160229360
08:57:31  RTP   > [02000022]  Local TX-Info: packets=1571, octets=251360
08:57:31  RTP   > [02000022]  Local RX-Info: packets=5, octets=800, lost=0, jitter=17, since last=26380mS
08:57:31  SIP_TR> [STACK] < Stack: from 192.168.216.5
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.216.5:5060;branch=z9hG4bK11b87a01;rport
Max-Forwards: 70
From: "xxx" <sip:[email protected]>;tag=as41f51440
To: <sip:00404xxx@sn4638-ha1>;tag=2210277577
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 395895816 395895817 IN IP4 192.168.210.70
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.210.70
t=0 0
m=audio 55984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

08:57:31  DP    > RTP-00/0022: getStatistics(00020900)
08:57:31  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Quality-Of-Service -> MOS 2.58, RTP, G.711 A-law (20ms), Local: Rx 5 pkts, 800 bytes, 0 lost, jitter 273 ms, Tx 1604 pkts, 256640 bytes, rtt 0 ms, Remote: Rx 0 pkts, 0 bytes, 0 lost, jitter 1 ms, Tx 5 pkts, 800 bytes, rtt 0 ms
08:57:32  ISDN  > #1252 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr(  3) Ns(110) pf=0
08:57:32  ISDN  > #1253 p: 0 <--- Layer2  tei:  0 sapi:  0 Disconnect
08:57:32  ISDN  > #1254      hex:08 01 8F 45 08 02 80 90 1E 02 81 88
08:57:32  ISDN  > #1255      IE: Cause  NormalCallClearing
08:57:32  ISDN  > #1256      IE: ProgressInd
08:57:32  ISDN  > #1257      to  CC key: 8388623 L3DisconnectInd
08:57:32  ISDN  > #1258      new call state : L3PcSt12U
08:57:32  ISDN  > #1259 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(111)         pf=0
08:57:32  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Cause -> Normal call clearing
08:57:32  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: State -> DISCONNECTING
08:57:32  DP    > TDM-00/00/00: The following config package could not be converted:
SEQUENCE {
  packageId UUIDpackageId UUID = CfgToneDetection
  argument SEQUENCE {
    profile GeneralString IMPLICIT [CONTEXT 1] = 'default' OPTIONAL
    detectDtmf BOOLEAN IMPLICIT [CONTEXT 2] = false OPTIONAL
  }
}
08:57:32  DP    > TDM-00/00/00: setOperationMode(receive-only)
08:57:32  TDM   > [DSP 0xe5af58] Admin operation mode send/receive -> receive-only
08:57:32  TDM   > [DSP 0xe5af58] Dsp operation mode send/receive -> receive-only
08:57:32  TDM   > [DSP 0xe5af58] Deactivating TX.
08:57:32  DSP   > Scheduler: removed source processor 0xe5b3ac
08:57:32  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: Provides Data -> false
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < Peer : Inband info: no
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < Peer : State: DISCONNECTING
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < Peer : Cause: Normal call clearing -> Code: 487 (Profile: default)
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < SM   : Event 'CC-Dropped' (Active State 'Connected', Passive State 'Idle')
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > SM   : New active state 'Disconnected' (Code: 0)
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > Peer : Code: 0 -> Cause: Normal call clearing (Profile: default)
08:57:32  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Drop call 00cae348
08:57:32  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Provides Data -> false
08:57:32  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: Cause -> Normal call clearing
08:57:32  CC    > [EP IF_VOIP_GATEWAY_HA-00ca88d0/active] Set call-leg property: State -> RELEASED
08:57:32  DP    > RTP-00/0022: subtractFromContext(00000022)
08:57:32  DP    > TDM-00/00/00: unregisterEventCallback(0xda5a18)
08:57:32  DP    > RTP-00/0022: unregisterEventCallback(0xda5a18)
08:57:32  TDM   > DATA_BUFF_TX: Update Connection. Processor: 0
08:57:32  DP    > TDM-00/00/00: removeConnector(00000000)
08:57:32  TDM   > [DSP 0xe5af58] Remove connector notify. ID: 0
08:57:32  DP    > RTP-00/0022: removeConnector(00000000)
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > SM   : Action 'Session-Drop'
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] < SM   : Event 'App-Drop' (State 'Established')
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : Action 'Stk-Bye'
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > Stack: BYE
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Update packet for destination: 192.168.216.5:5060
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] ---------------------------------------------------
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Traffic-Class           : local-default
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Update invite transaction timeout     : 32
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Update non-invite transaction timeout : 32
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating VIA Header              : 192.168.216.7:5060
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Local UDP Address       : 192.168.216.7:5060
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] Updating Datapath Controllers    : 192.168.216.7
08:57:32  SIP_TR> [STACK] > Stack: to 192.168.216.5
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.216.7:5060;branch=z9hG4bKdc648d9854c70686f
Max-Forwards: 70
From: <sip:00404xxx@sn4638-ha1>;tag=2210277577
To: "xxx" <sip:[email protected]>;tag=as41f51440
Call-ID: [email protected]
CSeq: 18011 BYE
User-Agent: Patton SN4638 5BIS 00A0BA048B52 R5.T 2009-05-28 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0


08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : New state 'Releasing'
08:57:32  DP    > TDM-00/00/00: subtractFromContext(00000022)
08:57:32  SIP_TR> [STACK] < Stack: from 192.168.216.5
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.216.7:5060;branch=z9hG4bKdc648d9854c70686f;received=192.168.216.7
From: <sip:00404xxx@sn4638-ha1>;tag=2210277577
To: "xxx" <sip:[email protected]>;tag=as41f51440
Call-ID: [email protected]
CSeq: 18011 BYE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


08:57:32  CC    > [EP IF_S0_00-00a324d0/active] Drop call 00cae348
08:57:32  CC    > [EP IF_S0_00-00a324d0/active] Set call-leg property: State -> RELEASED
08:57:32  DP    > TDM-00/00/00: setOperationMode(inactive)
08:57:32  TDM   > [DSP 0xe5af58] Admin operation mode receive-only -> inactive
08:57:32  TDM   > [DSP 0xe5af58] Dsp operation mode receive-only -> inactive
08:57:32  TDM   > [DSP 0xe5af58] Deactivating RX.
08:57:32  TDM   > DATA_BUFF_TX: Reset drops pending packet
08:57:32  DSP   > [00000000][160230830] State=voice: Event=close
08:57:32  DSP   > [00000000][160230830] Set tx plugin to none.
08:57:32  DSP   > [00000000][160230830] Set tx media type 2.
08:57:32  DSP   > [00000000][160230830] Set rx media type 2.
08:57:32  DSP   > [00000000][160230830] Closing (dsp 0, channel 0)
08:57:32  DSP   > [Channel 0 0] SENDING IDLE PACKET
08:57:32  DSP   > [00000000][160230840] New state=closed
08:57:32  TDM   > [DSP 0xe5af58] Disconnected port 0 bChannel 0 from DSP timeslot 0
08:57:32  TDM   > TDM-00/00/00 Releasing DSP resource.
08:57:32  ISDN  > #1260 p: 0 from CC key: 8388623 L3ReleaseReq
08:57:32  ISDN  > #1261 cause : NormalCallClearing
08:57:32  ISDN  > #1262      Disconnects bchan BCh0  cref : 0x0f
08:57:32  ISDN  > #1263      ---> Layer2  tei:  0 sapi:  0 Release
08:57:32  ISDN  > #1264      hex:08 01 0F 4D 08 02 80 90
08:57:32  ISDN  > #1265      IE: Cause  NormalCallClearing
08:57:32  ISDN  > #1266      new call state : L3PcSt19U
08:57:32  ISDN  > #1267 p: 0 S: sapi: 0 cr=0 ea=0 tei:  0 ea=1 INFO Nr(111) Ns(  3) pf=0
08:57:32  DSP   > [Channel 0 0] IDLE_PACKET_RECEIVED
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] < Stack: 200 OK (BYE)
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] < SM   : Event 'Stk-Int-Transaction-Finished' (State 'Releasing')
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] < SM   : Event 'Stk-Term-Status' (State 'Releasing')
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0 SDP] < SM   : Event 'Reset' (State 'Negotiated')
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : New state 'Released'
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0 SES 0xcaabd0] > SM   : Action 'App-Finished'
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] < SM   : Event 'Active-Session-Finished' (Active State 'Disconnected', Passive State 'Idle')
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > SM   : New active state 'Released' (Code: 0)
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] > SM   : Action 'App-Finished'
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] Finished
08:57:32  SIP_SI> [PR IF_VOIP_GATEWAY_HA] Removed endpoint IF_VOIP_GATEWAY_HA-00ca88d0
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] Destroying a session
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] Destroying a session
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] Removing a session
08:57:32  ISDN  > #1268 p: 0 R: sapi: 0 cr=0 ea=0 tei:  0 ea=1 RR   Nr(  4)         pf=0
08:57:32  DP    > RTP-00/0022: unregisterEventCallback(0xa18a5c)
08:57:32  DP    > RTP-00/0022: setOperationMode(inactive)
08:57:32  RTP   > [02000022] Set mode: TX/RX -> INACTIVE
08:57:32  DPMUX > Changed next processor of port 0x0022face: 0x0
08:57:32  RTP   > [02000022] (BCD) Event=disable | New State=disabled
08:57:32  DPMUX > Deactivating directpath port: 0x0022face
08:57:32  NTE   > [00e42370] Tx reset request.
08:57:32  NTE   > Scheduler: Unregister element immediately 0x00e42370
08:57:32  NTE   > Scheduler: Unregistered element: 00e42370
08:57:32  RTP   > [TERM 2000022] Tx-State=idle | Tx-Event=reset
08:57:32  RTP   > [TERM 2000022] Rx-State=idle | Rx-Event=reset
08:57:32  RTP   > [TERM 2000022] Stop Rx Fiter-Timeout
08:57:32  ISDN  > #1269 p: 0 R: sapi: 0 cr=1 ea=0 tei:  0 ea=1 INFO Nr(  4) Ns(111) pf=0
08:57:32  ISDN  > #1270 p: 0 <--- Layer2  tei:  0 sapi:  0 ReleaseCom
08:57:32  ISDN  > #1271      hex:08 01 8F 5A
08:57:32  ISDN  > #1272      to  CC key: 8388623 L3ReleaseConf
08:57:32  ISDN  > #1273      DEALLOC bchan BCh0 cref : 0x0f
08:57:32  ISDN  > #1274      new call state : L3PcSt00U
08:57:32  ISDN  > #1275      *delPc U, tei:  0 sapi:  0
08:57:32  ISDN  > #1276 p: 0 S: sapi: 0 cr=1 ea=0 tei:  0 ea=1 RR   Nr(112)         pf=0
08:57:32  SIP_SI> [EP IF_VOIP_GATEWAY_HA-00ca88d0] Destroying a session
08:57:32  SIP_SI> [PR IF_VOIP_GATEWAY_HA] < SM   : Event 'Ep-Finished' (State 'Up')
08:57:32  DP    > TDM-00/00/00: unregisterEventCallback(0xa2d484)

Kurz noch zur Erinnerung... der Anruf aus dem PSTN Netz auf das Sip Telefon funktioniert noch immer Reibungslos. Nur ausgehende Telefonate funktionieren nicht.

Jemand eine Idee?

JoeMoes · 4 Sep 2009

Joa... endlich mal eine schöne Nachricht zum Wochenende hin.

Da alle SIP Pakete ja nun liefen und "nur" rtp nicht lief, hab ich mir alles mit Wireshark einmal angeschaut. Und siehe da: Unknown RTP version 0

Im Detail bedeutet das nichts anderes, das sich die Patton mit meinem Snom nicht verstanden hat. Sip Reinvite raus genommen und alles lief glatt.

Zur Zusammenfassung:
Mein Interface brauchte folgende Option:

Code:

  interface isdn IF_S0_00
    no inband-info accept transparent

Und halt noch das hässliche RTP Problemchen, was man als sterblicher eigentlich nicht unbedingt hat. :spocht:

Thx fürs lesen

Netview · 10 Sep 2009

Ich denke hier liegt der Fehler:

...
canreinvite= sollte auf 'no' gesetzt sein!

JoeMoes · 10 Sep 2009

Jupp genau... weil leider mein Snom Telefon und die Patton sich nicht mochten per RTP... warum auch immer.

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