Warum Pickup bei ausgehenden Gesprächen????

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MarcelDeLuX

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Hab hier mal ne ganz komische Situation:

Das Pickup mittels *8 funzt soweit super, offensichtlich aber zu gut.

Ein Telefon (GXP2000) wählt also eine externe Nummer, die über Capi vermittelt wird. Der Anrufer hört das Freizeichen und auch die Gegenstelle klingelt (getestet mit Handy).
Nun kommt ein anderes Telefon und macht den Pickup mittels *8 und bekommt das eigentlich extern anrufende Gespräch. Die angerufene Gegenstelle wird dabei natürlich unterbrochen...

Wasn hier los?

Hier mal nen Auszug aus dem Logfile:
Code: 
Nov  6 15:27:22 DEBUG[29057] chan_sip.c: Setting NAT on RTP to 0
Nov  6 15:27:22 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Response 27508: Match Found
Nov  6 15:27:22 DEBUG[29057] chan_sip.c: Setting NAT on RTP to 0
Nov  6 15:27:22 DEBUG[29057] chan_sip.c: Checking SIP call limits for device 18
Nov  6 15:27:22 DEBUG[29057] chan_sip.c: build_route: Contact hop: <sip:[email protected]:5060;transport=udp>
Nov  6 15:27:22 DEBUG[29050] channel.c: Avoiding initial deadlock for 'SIP/18-09508a00'
Nov  6 15:27:22 VERBOSE[27916] logger.c:     -- Executing SetCallerPres("SIP/18-09508a00", "prohib_not_screened") in new stack
Nov  6 15:27:22 VERBOSE[27916] logger.c:     -- Executing Dial("SIP/18-09508a00", "CAPI/g1/XXXXXX/b") in new stack
Nov  6 15:27:22 VERBOSE[27916] logger.c:        > data = g1/XXXXXX/b format=8
Nov  6 15:27:22 VERBOSE[27916] logger.c:        > capi request group = 2
Nov  6 15:27:22 VERBOSE[27916] logger.c:   == dokom#01: Call CAPI/dokom#01/XXXXXX-bad with B3  (pres=0x20, ton=0x00)
Nov  6 15:27:22 VERBOSE[29052] logger.c:     -- dokom#01: received CONNECT_CONF PLCI = 0x204
Nov  6 15:27:22 VERBOSE[27916] logger.c:     -- Called g1/XXXXXX/b
Nov  6 15:27:22 VERBOSE[29050] logger.c:  Extension Changed 18 new state InUse for Notify User 14
Nov  6 15:27:22 VERBOSE[29050] logger.c:  Extension Changed 18 new state InUse for Notify User 27
Nov  6 15:27:22 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Request 776: Match Found
Nov  6 15:27:22 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Request 815: Match Found
Nov  6 15:27:23 VERBOSE[29052] logger.c:     -- CAPI queue frame: [ TYPE: Control (4) SUBCLASS: Unknown control '15' (15) ] [dokom#01]
Nov  6 15:27:23 VERBOSE[27916] logger.c:     -- CAPI/dokom#01/XXXXXX-bad is proceeding passing it to SIP/18-09508a00
Nov  6 15:27:24 VERBOSE[29052] logger.c:     -- CAPI queue frame: [ TYPE: Control (4) SUBCLASS: Unknown control '14' (14) ] [dokom#01]
Nov  6 15:27:24 VERBOSE[27916] logger.c:     -- CAPI/dokom#01/XXXXXX-bad is making progress passing it to SIP/18-09508a00
Nov  6 15:27:24 VERBOSE[29052] logger.c:     -- CAPI queue frame: [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [dokom#01]
Nov  6 15:27:24 VERBOSE[27916] logger.c:     -- CAPI/dokom#01/XXXXXX-bad is ringing
Nov  6 15:27:29 DEBUG[29057] chan_sip.c: Setting NAT on RTP to 0
Nov  6 15:27:29 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Response 32495: Match Found
Nov  6 15:27:29 DEBUG[29057] chan_sip.c: Setting NAT on RTP to 0
Nov  6 15:27:29 DEBUG[29057] chan_sip.c: Checking SIP call limits for device 24
Nov  6 15:27:29 DEBUG[29057] chan_sip.c: build_route: Contact hop: <sip:[email protected]:5060;transport=udp>
Nov  6 15:27:29 DEBUG[29050] channel.c: Avoiding initial deadlock for 'SIP/24-094f0c00'
Nov  6 15:27:29 DEBUG[29057] channel.c: Planning to masquerade channel SIP/24-094f0c00 into the structure of CAPI/dokom#01/XXXXXX-bad
Nov  6 15:27:29 DEBUG[29057] channel.c: Done planning to masquerade channel SIP/24-094f0c00 into the structure of CAPI/dokom#01/XXXXXX-bad
Nov  6 15:27:29 DEBUG[27916] channel.c: Got clone lock for masquerade on 'SIP/24-094f0c00' at 0x94e1a3c
Nov  6 15:27:29 VERBOSE[27916] logger.c:     -- dokom#01: activehangingup (cause=0) for PLCI=0x204
Nov  6 15:27:29 DEBUG[27916] channel.c: Putting channel SIP/24-094f0c00 in 64/64 formats
Nov  6 15:27:29 DEBUG[27916] channel.c: Destroying channel clone 'CAPI/dokom#01/XXXXXX-bad<ZOMBIE>'
Nov  6 15:27:29 DEBUG[27916] channel.c: Done Masquerading SIP/24-094f0c00 (0)
Nov  6 15:27:29 VERBOSE[27916] logger.c:     -- SIP/24-094f0c00 answered SIP/18-09508a00
Nov  6 15:27:29 VERBOSE[27916] logger.c:     -- Attempting native bridge of SIP/18-09508a00 and SIP/24-094f0c00
Nov  6 15:27:29 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Response 27509: Match Found
Nov  6 15:27:29 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Response 32496: Match Found
Nov  6 15:27:29 VERBOSE[29052] logger.c:   == dokom#01: Interface cleanup PLCI=0x204
Nov  6 15:27:34 DEBUG[27916] channel.c: Didn't get a frame from channel: SIP/24-094f0c00
Nov  6 15:27:34 DEBUG[27916] channel.c: Bridge stops bridging channels SIP/18-09508a00 and SIP/24-094f0c00
Nov  6 15:27:34 DEBUG[27916] chan_sip.c: update_call_counter(24) - decrement call limit counter
Nov  6 15:27:34 DEBUG[27916] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Nov  6 15:27:34 VERBOSE[27916] logger.c:   == Spawn extension (from-internal, XXXXXX, 2) exited non-zero on 'SIP/18-09508a00'
Nov  6 15:27:34 VERBOSE[27916] logger.c:     -- Executing Macro("SIP/18-09508a00", "hangupcall") in new stack
Nov  6 15:27:34 VERBOSE[27916] logger.c:     -- Executing ResetCDR("SIP/18-09508a00", "w") in new stack
Nov  6 15:27:34 DEBUG[27916] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Nov  6 15:27:34 DEBUG[27916] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES ('2007-11-06 15:27:22','\"PM\" <18>','18','XXXXXX','from-internal', 'SIP/18-09508a00','SIP/24-094f0c00','ResetCDR','w',12,5,'ANSWERED',3,'','1194359242.5393')
Nov  6 15:27:34 DEBUG[27916] app_macro.c: Executed application: ResetCDR
Nov  6 15:27:34 VERBOSE[27916] logger.c:     -- Executing NoCDR("SIP/18-09508a00", "") in new stack
Nov  6 15:27:34 NOTICE[27916] cdr.c: CDR on channel 'SIP/18-09508a00' not posted
Nov  6 15:27:34 NOTICE[27916] cdr.c: CDR on channel 'SIP/18-09508a00' lacks end
Nov  6 15:27:34 DEBUG[27916] app_macro.c: Executed application: NoCDR
Nov  6 15:27:34 DEBUG[27916] pbx.c: Expression result is '1'
Nov  6 15:27:34 VERBOSE[27916] logger.c:     -- Executing GotoIf("SIP/18-09508a00", "1?skiprg") in new stack
Nov  6 15:27:34 VERBOSE[27916] logger.c:     -- Goto (macro-hangupcall,s,6)
Nov  6 15:27:34 DEBUG[27916] app_macro.c: Executed application: GotoIf
Nov  6 15:27:34 DEBUG[27916] pbx.c: Expression result is '1'
Nov  6 15:27:34 VERBOSE[27916] logger.c:     -- Executing GotoIf("SIP/18-09508a00", "1?skipblkvm") in new stack
Nov  6 15:27:34 VERBOSE[27916] logger.c:     -- Goto (macro-hangupcall,s,9)
Nov  6 15:27:34 DEBUG[27916] app_macro.c: Executed application: GotoIf
Nov  6 15:27:34 DEBUG[27916] pbx.c: Expression result is '1'
Nov  6 15:27:34 VERBOSE[27916] logger.c:     -- Executing GotoIf("SIP/18-09508a00", "1?theend") in new stack
Nov  6 15:27:34 VERBOSE[27916] logger.c:     -- Goto (macro-hangupcall,s,11)
Nov  6 15:27:34 DEBUG[27916] app_macro.c: Executed application: GotoIf
Nov  6 15:27:34 VERBOSE[27916] logger.c:     -- Executing Hangup("SIP/18-09508a00", "") in new stack
Nov  6 15:27:34 VERBOSE[27916] logger.c:   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/18-09508a00' in macro 'hangupcall'
Nov  6 15:27:34 VERBOSE[27916] logger.c:   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/18-09508a00'
Nov  6 15:27:34 DEBUG[27916] chan_sip.c: update_call_counter(18) - decrement call limit counter
Nov  6 15:27:34 VERBOSE[29050] logger.c:  Extension Changed 18 new state Idle for Notify User 14
Nov  6 15:27:34 VERBOSE[29050] logger.c:  Extension Changed 18 new state Idle for Notify User 27
 
Könntest du bitte mal ein log mit
set verbose 5
capi debug
machen?

Armin
 
here u are:

Ich hoffe, dass waren alle einträge von diesem Vorgang, musste zwischendrin was rausnehmen, was nicht dazu gehörte.

Code: 
Nov  7 09:17:55 VERBOSE[29057] logger.c: --- (11 headers 0 lines) ---
Nov  7 09:17:55 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Found
Nov  7 09:17:55 VERBOSE[29057] logger.c: Destroying call '[email protected]'
Nov  7 09:17:56 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.118:5060: 
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.118:5060;branch=z9hG4bK03bc23f32c75ac61
From: "Daniel Dommisch" <sip:[email protected]>;tag=a9cab45464e7c758
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060;transport=udp>
Supported: replaces, timer, path
Call-ID: [email protected]
CSeq: 548 INVITE
User-Agent: Grandstream GXP2000 1.1.4.18
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 256

v=0
o=17 8000 8000 IN IP4 192.168.100.118
s=SIP Call
c=IN IP4 192.168.100.118
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=ptime:20

Nov  7 09:17:56 VERBOSE[29057] logger.c: --- (13 headers 13 lines) ---
Nov  7 09:17:56 VERBOSE[29057] logger.c: Using INVITE request as basis request - [email protected]
Nov  7 09:17:56 VERBOSE[29057] logger.c: Sending to 192.168.100.118 : 5060 (non-NAT)
Nov  7 09:17:56 DEBUG[29057] chan_sip.c: Setting NAT on RTP to 0
Nov  7 09:17:56 VERBOSE[29057] logger.c: Reliably Transmitting (no NAT) to 192.168.100.118:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.118:5060;branch=z9hG4bK03bc23f32c75ac61;received=192.168.100.118
From: "Daniel Dommisch" <sip:[email protected]>;tag=a9cab45464e7c758
To: <sip:[email protected]>;tag=as2e5cf427
Call-ID: [email protected]
CSeq: 548 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="783daeba"
Content-Length: 0


---
Nov  7 09:17:56 VERBOSE[29057] logger.c: Scheduling destruction of call '[email protected]' in 15000 ms
Nov  7 09:17:56 VERBOSE[29057] logger.c: Found user '17'
Nov  7 09:17:56 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.118:5060: 
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.118:5060;branch=z9hG4bK03bc23f32c75ac61
From: "Daniel Dommisch" <sip:[email protected]>;tag=a9cab45464e7c758
To: <sip:[email protected]>;tag=as2e5cf427
Contact: <sip:[email protected]:5060;transport=udp>
Supported: path
Call-ID: [email protected]
CSeq: 548 ACK
User-Agent: Grandstream GXP2000 1.1.4.18
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


Nov  7 09:17:56 VERBOSE[29057] logger.c: --- (12 headers 0 lines) ---
Nov  7 09:17:56 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Response 548: Match Found
Nov  7 09:17:56 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.118:5060: 
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.118:5060;branch=z9hG4bKd3bc96acff559345
From: "Daniel Dommisch" <sip:[email protected]>;tag=a9cab45464e7c758
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060;transport=udp>
Supported: replaces, timer, path
Proxy-Authorization: Digest username="17", realm="asterisk", algorithm=MD5, uri="sip:[email protected]", nonce="783daeba", response="f3021a978f17907ac6c99ab91c2a1a57"
Call-ID: [email protected]
CSeq: 549 INVITE
User-Agent: Grandstream GXP2000 1.1.4.18
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 256

v=0
o=17 8000 8001 IN IP4 192.168.100.118
s=SIP Call
c=IN IP4 192.168.100.118
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=ptime:20

Nov  7 09:17:56 VERBOSE[29057] logger.c: --- (14 headers 13 lines) ---
Nov  7 09:17:56 VERBOSE[29057] logger.c: Using INVITE request as basis request - [email protected]
Nov  7 09:17:56 VERBOSE[29057] logger.c: Sending to 192.168.100.118 : 5060 (non-NAT)
Nov  7 09:17:56 DEBUG[29057] chan_sip.c: Setting NAT on RTP to 0
Nov  7 09:17:56 VERBOSE[29057] logger.c: Found user '17'
Nov  7 09:17:56 VERBOSE[29057] logger.c: Found RTP audio format 0
Nov  7 09:17:56 VERBOSE[29057] logger.c: Found RTP audio format 8
Nov  7 09:17:56 VERBOSE[29057] logger.c: Found RTP audio format 4
Nov  7 09:17:56 VERBOSE[29057] logger.c: Found RTP audio format 18
Nov  7 09:17:56 VERBOSE[29057] logger.c: Found RTP audio format 3
Nov  7 09:17:56 VERBOSE[29057] logger.c: Peer audio RTP is at port 192.168.100.118:5004
Nov  7 09:17:56 DEBUG[29057] chan_sip.c: Peer audio RTP is at port 192.168.100.118:5004
Nov  7 09:17:56 VERBOSE[29057] logger.c: Found description format PCMU
Nov  7 09:17:56 VERBOSE[29057] logger.c: Found description format PCMA
Nov  7 09:17:56 VERBOSE[29057] logger.c: Found description format G723
Nov  7 09:17:56 VERBOSE[29057] logger.c: Found description format G729
Nov  7 09:17:56 VERBOSE[29057] logger.c: Found description format GSM
Nov  7 09:17:56 VERBOSE[29057] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Nov  7 09:17:56 VERBOSE[29057] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Nov  7 09:17:56 DEBUG[29057] chan_sip.c: Checking SIP call limits for device 17
Nov  7 09:17:56 VERBOSE[29057] logger.c: Looking for 0172XXXXXXX in from-internal (domain 192.168.100.200)
Nov  7 09:17:56 DEBUG[29057] chan_sip.c: build_route: Contact hop: <sip:[email protected]:5060;transport=udp>
Nov  7 09:17:56 VERBOSE[29057] logger.c: list_route: hop: <sip:[email protected]:5060;transport=udp>
Nov  7 09:17:56 VERBOSE[29057] logger.c: Transmitting (no NAT) to 192.168.100.118:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.118:5060;branch=z9hG4bKd3bc96acff559345;received=192.168.100.118
From: "Daniel Dommisch" <sip:[email protected]>;tag=a9cab45464e7c758
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 549 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0

---
...
...
---

Nov  7 09:17:56 VERBOSE[29057] logger.c: --- (11 headers 0 lines) ---
Nov  7 09:17:56 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Request 872: Match Found
Nov  7 09:18:03 VERBOSE[30411] logger.c:     -- Executing Dial("SIP/17-094c8580", "CAPI/g1/0172XXXXXXX/b") in new stack
Nov  7 09:18:03 VERBOSE[30411] logger.c:        > data = g1/0172XXXXXXX/b format=8
Nov  7 09:18:03 VERBOSE[30411] logger.c:        > parsed dialstring: 'g1' 'NULL' '0172XXXXXXX' 'b'
Nov  7 09:18:03 VERBOSE[30411] logger.c:        > capi request group = 2
Nov  7 09:18:03 VERBOSE[30411] logger.c:   == dokom#02: setting format alaw - 0x8 (alaw)
Nov  7 09:18:03 VERBOSE[30411] logger.c:        > parsed dialstring: 'g1' 'NULL' '0172XXXXXXX' 'b'
Nov  7 09:18:03 VERBOSE[30411] logger.c:   == dokom#02: Call CAPI/dokom#02/0172XXXXXXX-c6e with B3  (pres=0x20, ton=0x00)
Nov  7 09:18:03 VERBOSE[29052] logger.c: CONNECT_CONF ID=002 #0x65fb LEN=0014

Nov  7 09:18:03 VERBOSE[30411] logger.c: CONNECT_CONF ID=002 #0x65fb LEN=0014

Nov  7 09:18:03 VERBOSE[30411] logger.c:     -- Called g1/0172XXXXXXX/b
Nov  7 09:18:03 VERBOSE[29050] logger.c:        > CAPI devicestate requested for dokom#02/0172XXXXXXX
Nov  7 09:18:03 VERBOSE[29052] logger.c:     -- dokom#02: received CONNECT_CONF PLCI = 0x104
Nov  7 09:18:03 VERBOSE[29050] logger.c:        > CAPI devicestate requested for dokom#02/0172XXXXXXX
Nov  7 09:18:03 VERBOSE[29052] logger.c: INFO_IND ID=002 #0xf72d LEN=0016
  Controller/PLCI/NCCI            = 0x104
  InfoNumber                      = 0xb
  InfoElement                     = 7

Nov  7 09:18:03 VERBOSE[29052] logger.c: INFO_RESP ID=002 #0xf72d LEN=0012
  Controller/PLCI/NCCI            = 0x104

Nov  7 09:18:03 VERBOSE[29052] logger.c:     -- dokom#02: unhandled INFO_IND 0xb (PLCI=0x104)
Nov  7 09:18:03 VERBOSE[29052] logger.c: INFO_IND ID=002 #0xf72e LEN=0016
  Controller/PLCI/NCCI            = 0x104
  InfoNumber                      = 0xa
  InfoElement                     = <00>

Nov  7 09:18:03 VERBOSE[29052] logger.c: INFO_RESP ID=002 #0xf72e LEN=0012
  Controller/PLCI/NCCI            = 0x104

Nov  7 09:18:03 VERBOSE[29052] logger.c:     -- dokom#02: unhandled INFO_IND 0xa (PLCI=0x104)
Nov  7 09:18:04 VERBOSE[29052] logger.c: INFO_IND ID=002 #0xf72f LEN=0016
  Controller/PLCI/NCCI            = 0x104
  InfoNumber                      = 0x18
  InfoElement                     = <89>

Nov  7 09:18:04 VERBOSE[29052] logger.c: INFO_RESP ID=002 #0xf72f LEN=0012
  Controller/PLCI/NCCI            = 0x104

Nov  7 09:18:04 VERBOSE[29052] logger.c:     -- dokom#02: info element CHANNEL IDENTIFICATION 89
Nov  7 09:18:04 VERBOSE[29052] logger.c: INFO_IND ID=002 #0xf730 LEN=0015
  Controller/PLCI/NCCI            = 0x104
  InfoNumber                      = 0x800d
  InfoElement                     = default

Nov  7 09:18:04 VERBOSE[29052] logger.c: INFO_RESP ID=002 #0xf730 LEN=0012
  Controller/PLCI/NCCI            = 0x104

Nov  7 09:18:04 VERBOSE[29052] logger.c:     -- dokom#02: info element SETUP ACK
Nov  7 09:18:10 VERBOSE[29052] logger.c: INFO_IND ID=002 #0xf731 LEN=0015
  Controller/PLCI/NCCI            = 0x104
  InfoNumber                      = 0x8001
  InfoElement                     = default

Nov  7 09:18:10 VERBOSE[29052] logger.c: INFO_RESP ID=002 #0xf731 LEN=0012
  Controller/PLCI/NCCI            = 0x104

Nov  7 09:18:10 VERBOSE[29052] logger.c:     -- dokom#02: info element ALERTING
Nov  7 09:18:10 VERBOSE[29052] logger.c: CONNECT_B3_REQ ID=002 #0x65fc LEN=0013
  Controller/PLCI/NCCI            = 0x104
  NCPI                            = default

Nov  7 09:18:10 VERBOSE[29052] logger.c:     -- dokom#02: sent CONNECT_B3_REQ PLCI=0x104
Nov  7 09:18:10 VERBOSE[29052] logger.c:     -- CAPI queue frame: [ TYPE: Control (4) SUBCLASS: Unknown control '14' (14) ] [dokom#02]
Nov  7 09:18:10 VERBOSE[29052] logger.c:     -- CAPI queue frame: [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [dokom#02]
Nov  7 09:18:10 VERBOSE[29052] logger.c: CONNECT_B3_CONF ID=002 #0x65fc LEN=0014
  Controller/PLCI/NCCI            = 0x10104
  Info                            = 0x0

Nov  7 09:18:10 VERBOSE[30411] logger.c:     -- CAPI/dokom#02/0172XXXXXXX-c6e is making progress passing it to SIP/17-094c8580
Nov  7 09:18:10 VERBOSE[30411] logger.c: We're at 192.168.100.200 port 15054
Nov  7 09:18:10 VERBOSE[30411] logger.c: Adding codec 0x4 (ulaw) to SDP
Nov  7 09:18:10 VERBOSE[30411] logger.c: Adding codec 0x8 (alaw) to SDP
Nov  7 09:18:10 VERBOSE[30411] logger.c: Transmitting (no NAT) to 192.168.100.118:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.118:5060;branch=z9hG4bKd3bc96acff559345;received=192.168.100.118
From: "Daniel Dommisch" <sip:[email protected]>;tag=a9cab45464e7c758
To: <sip:[email protected]>;tag=as7d2474b7
Call-ID: [email protected]
CSeq: 549 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 190

v=0
o=root 29046 29046 IN IP4 192.168.100.200
s=session
c=IN IP4 192.168.100.200
t=0 0
m=audio 15054 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

---
Nov  7 09:18:10 VERBOSE[30411] logger.c:     -- CAPI/dokom#02/0172XXXXXXX-c6e is ringing
Nov  7 09:18:10 VERBOSE[30411] logger.c: Transmitting (no NAT) to 192.168.100.118:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.118:5060;branch=z9hG4bKd3bc96acff559345;received=192.168.100.118
From: "Daniel Dommisch" <sip:[email protected]>;tag=a9cab45464e7c758
To: <sip:[email protected]>;tag=as7d2474b7
Call-ID: [email protected]
CSeq: 549 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
Nov  7 09:18:10 VERBOSE[29050] logger.c:        > CAPI devicestate requested for dokom#02/0172XXXXXXX
Nov  7 09:18:10 VERBOSE[29052] logger.c: CONNECT_B3_ACTIVE_IND ID=002 #0xf732 LEN=0013
  Controller/PLCI/NCCI            = 0x10104
  NCPI                            = default

Nov  7 09:18:10 VERBOSE[29052] logger.c: CONNECT_B3_ACTIVE_RESP ID=002 #0xf732 LEN=0012
  Controller/PLCI/NCCI            = 0x10104

Nov  7 09:18:12 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.180:5060: 
INVITE sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK301bd1088cb7fd87
From: "Gabi Drescher" <sip:[email protected]>;tag=6aa7aad3d7ac10bc
To: <sip:*[email protected]>
Contact: <sip:[email protected]:5060;transport=udp>
Supported: replaces, timer, path
Call-ID: [email protected]
CSeq: 28930 INVITE
User-Agent: Grandstream GXP2000 1.1.4.18
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 256

v=0
o=13 8000 8000 IN IP4 192.168.100.180
s=SIP Call
c=IN IP4 192.168.100.180
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=ptime:20

Nov  7 09:18:12 VERBOSE[29057] logger.c: --- (13 headers 13 lines) ---
Nov  7 09:18:12 VERBOSE[29057] logger.c: Using INVITE request as basis request - [email protected]
Nov  7 09:18:12 VERBOSE[29057] logger.c: Sending to 192.168.100.180 : 5060 (non-NAT)
Nov  7 09:18:12 DEBUG[29057] chan_sip.c: Setting NAT on RTP to 0
Nov  7 09:18:12 VERBOSE[29057] logger.c: Reliably Transmitting (no NAT) to 192.168.100.180:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK301bd1088cb7fd87;received=192.168.100.180
From: "Gabi Drescher" <sip:[email protected]>;tag=6aa7aad3d7ac10bc
To: <sip:*[email protected]>;tag=as246f6e2b
Call-ID: [email protected]
CSeq: 28930 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="254e6c14"
Content-Length: 0


---
Nov  7 09:18:12 VERBOSE[29057] logger.c: Scheduling destruction of call '[email protected]' in 15000 ms
Nov  7 09:18:12 VERBOSE[29057] logger.c: Found user '13'
Nov  7 09:18:12 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.180:5060: 
ACK sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK301bd1088cb7fd87
From: "Gabi Drescher" <sip:[email protected]>;tag=6aa7aad3d7ac10bc
To: <sip:*[email protected]>;tag=as246f6e2b
Contact: <sip:[email protected]:5060;transport=udp>
Supported: path
Call-ID: [email protected]
CSeq: 28930 ACK
User-Agent: Grandstream GXP2000 1.1.4.18
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


Nov  7 09:18:12 VERBOSE[29057] logger.c: --- (12 headers 0 lines) ---
Nov  7 09:18:12 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Response 28930: Match Found
Nov  7 09:18:12 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.180:5060: 
INVITE sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK42a4bb2837982bd7
From: "Gabi Drescher" <sip:[email protected]>;tag=6aa7aad3d7ac10bc
To: <sip:*[email protected]>
Contact: <sip:[email protected]:5060;transport=udp>
Supported: replaces, timer, path
Proxy-Authorization: Digest username="13", realm="asterisk", algorithm=MD5, uri="sip:*[email protected]", nonce="254e6c14", response="5f1233cbd40f64dcd0ca19a5566e5775"
Call-ID: [email protected]
CSeq: 28931 INVITE
User-Agent: Grandstream GXP2000 1.1.4.18
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 256

v=0
o=13 8000 8001 IN IP4 192.168.100.180
s=SIP Call
c=IN IP4 192.168.100.180
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=ptime:20

Nov  7 09:18:12 VERBOSE[29057] logger.c: --- (14 headers 13 lines) ---
Nov  7 09:18:12 VERBOSE[29057] logger.c: Using INVITE request as basis request - [email protected]
Nov  7 09:18:12 VERBOSE[29057] logger.c: Sending to 192.168.100.180 : 5060 (non-NAT)
Nov  7 09:18:12 DEBUG[29057] chan_sip.c: Setting NAT on RTP to 0
Nov  7 09:18:12 VERBOSE[29057] logger.c: Found user '13'
Nov  7 09:18:12 VERBOSE[29057] logger.c: Found RTP audio format 0
Nov  7 09:18:12 VERBOSE[29057] logger.c: Found RTP audio format 8
Nov  7 09:18:12 VERBOSE[29057] logger.c: Found RTP audio format 4
Nov  7 09:18:12 VERBOSE[29057] logger.c: Found RTP audio format 18
Nov  7 09:18:12 VERBOSE[29057] logger.c: Found RTP audio format 3
Nov  7 09:18:12 VERBOSE[29057] logger.c: Peer audio RTP is at port 192.168.100.180:5004
Nov  7 09:18:12 DEBUG[29057] chan_sip.c: Peer audio RTP is at port 192.168.100.180:5004
Nov  7 09:18:12 VERBOSE[29057] logger.c: Found description format PCMU
Nov  7 09:18:12 VERBOSE[29057] logger.c: Found description format PCMA
Nov  7 09:18:12 VERBOSE[29057] logger.c: Found description format G723
Nov  7 09:18:12 VERBOSE[29057] logger.c: Found description format G729
Nov  7 09:18:12 VERBOSE[29057] logger.c: Found description format GSM
Nov  7 09:18:12 VERBOSE[29057] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Nov  7 09:18:12 VERBOSE[29057] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Nov  7 09:18:12 DEBUG[29057] chan_sip.c: Checking SIP call limits for device 13
Nov  7 09:18:12 VERBOSE[29057] logger.c: Looking for *8 in from-internal (domain 192.168.100.200)
Nov  7 09:18:12 DEBUG[29057] chan_sip.c: build_route: Contact hop: <sip:[email protected]:5060;transport=udp>
Nov  7 09:18:12 VERBOSE[29057] logger.c: list_route: hop: <sip:[email protected]:5060;transport=udp>
Nov  7 09:18:12 VERBOSE[29057] logger.c: Transmitting (no NAT) to 192.168.100.180:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK42a4bb2837982bd7;received=192.168.100.180
From: "Gabi Drescher" <sip:[email protected]>;tag=6aa7aad3d7ac10bc
To: <sip:*[email protected]>
Call-ID: [email protected]
CSeq: 28931 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*[email protected]>
Content-Length: 0


---
Nov  7 09:18:12 VERBOSE[29057] logger.c: We're at 192.168.100.200 port 14834
Nov  7 09:18:12 VERBOSE[29057] logger.c: Adding codec 0x4 (ulaw) to SDP
Nov  7 09:18:12 VERBOSE[29057] logger.c: Adding codec 0x8 (alaw) to SDP
Nov  7 09:18:12 VERBOSE[29057] logger.c: Reliably Transmitting (no NAT) to 192.168.100.180:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK42a4bb2837982bd7;received=192.168.100.180
From: "Gabi Drescher" <sip:[email protected]>;tag=6aa7aad3d7ac10bc
To: <sip:*[email protected]>;tag=as62b5189b
Call-ID: [email protected]
CSeq: 28931 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*[email protected]>
Content-Type: application/sdp
Content-Length: 190

v=0
o=root 29046 29046 IN IP4 192.168.100.200
s=session
c=IN IP4 192.168.100.200
t=0 0
m=audio 14834 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

---
Nov  7 09:18:12 DEBUG[29057] channel.c: Planning to masquerade channel SIP/13-094fd2d0 into the structure of CAPI/dokom#02/0172XXXXXXX-c6e
Nov  7 09:18:12 DEBUG[29057] channel.c: Done planning to masquerade channel SIP/13-094fd2d0 into the structure of CAPI/dokom#02/0172XXXXXXX-c6e
Nov  7 09:18:12 VERBOSE[29050] logger.c: Reliably Transmitting (no NAT) to 192.168.100.106:5060:
NOTIFY sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK66593524;rport
From: <sip:[email protected]>;tag=as6d9c78e9
To: "Nicole Killing" <sip:[email protected]>;tag=6989058941620062
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 822 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="720" state="full" entity="sip:[email protected]">
<dialog id="13">
<state>confirmed</state>
</dialog>
</dialog-info>

---
Nov  7 09:18:12 VERBOSE[29050] logger.c:  Extension Changed 13 new state InUse for Notify User 14
Nov  7 09:18:12 VERBOSE[29050] logger.c: Reliably Transmitting (no NAT) to 192.168.100.218:5060:
NOTIFY sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK18222860;rport
From: <sip:[email protected]>;tag=as7ebc6b44
To: "Peter Montero" <sip:[email protected]>;tag=30345aa7c96095a7
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 865 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="763" state="full" entity="sip:[email protected]">
<dialog id="13">
<state>confirmed</state>
</dialog>
</dialog-info>

---
Nov  7 09:18:12 VERBOSE[29050] logger.c:  Extension Changed 13 new state InUse for Notify User 18
Nov  7 09:18:12 VERBOSE[29050] logger.c: Reliably Transmitting (no NAT) to 192.168.100.227:5060:
NOTIFY sip:[email protected]:5060;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK32a96e01;rport
From: <sip:[email protected];user=phone>;tag=as1d9addff
To: "Marcel Linzer" <sip:[email protected];user=phone>;tag=d53f782fe57b0ab5
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 862 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 205

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="760" state="full" entity="sip:[email protected]">
<dialog id="13">
<state>confirmed</state>
</dialog>
</dialog-info>

---
Nov  7 09:18:12 VERBOSE[29050] logger.c:  Extension Changed 13 new state InUse for Notify User 27
Nov  7 09:18:12 DEBUG[30411] channel.c: Got clone lock for masquerade on 'SIP/13-094fd2d0' at 0x94e331c
Nov  7 09:18:12 VERBOSE[30411] logger.c:   == dokom#02: SIP/13-094fd2d0 fixup now SIP/13-094fd2d0<MASQ>
Nov  7 09:18:12 VERBOSE[30411] logger.c:   == dokom#02: CAPI Hangingup for PLCI=0x104 in state 5
Nov  7 09:18:12 VERBOSE[30411] logger.c:     -- dokom#02: activehangingup (cause=0) for PLCI=0x104
Nov  7 09:18:12 VERBOSE[30411] logger.c: DISCONNECT_B3_REQ ID=002 #0x6663 LEN=0013
  Controller/PLCI/NCCI            = 0x10104
  NCPI                            = default

Nov  7 09:18:12 VERBOSE[29052] logger.c: DISCONNECT_B3_CONF ID=002 #0x6663 LEN=0014
  Controller/PLCI/NCCI            = 0x10104
  Info                            = 0x0

Nov  7 09:18:12 DEBUG[29050] channel.c: Avoiding initial deadlock for 'SIP/13-094fd2d0'
Nov  7 09:18:12 DEBUG[30411] channel.c: Putting channel SIP/13-094fd2d0 in 64/64 formats
Nov  7 09:18:12 DEBUG[30411] channel.c: Destroying channel clone 'CAPI/dokom#02/0172XXXXXXX-c6e<ZOMBIE>'
Nov  7 09:18:12 DEBUG[30411] channel.c: Done Masquerading SIP/13-094fd2d0 (0)
Nov  7 09:18:12 VERBOSE[30411] logger.c:     -- SIP/13-094fd2d0 answered SIP/17-094c8580
Nov  7 09:18:12 VERBOSE[29052] logger.c: DISCONNECT_B3_IND ID=002 #0xf79b LEN=0015
  Controller/PLCI/NCCI            = 0x10104
  Reason_B3                       = 0x0
  NCPI                            = default

Nov  7 09:18:12 VERBOSE[30411] logger.c: We're at 192.168.100.200 port 15054
Nov  7 09:18:12 VERBOSE[30411] logger.c: Adding codec 0x4 (ulaw) to SDP
Nov  7 09:18:12 VERBOSE[30411] logger.c: Adding codec 0x8 (alaw) to SDP
Nov  7 09:18:12 VERBOSE[30411] logger.c: Reliably Transmitting (no NAT) to 192.168.100.118:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.118:5060;branch=z9hG4bKd3bc96acff559345;received=192.168.100.118
From: "Daniel Dommisch" <sip:[email protected]>;tag=a9cab45464e7c758
To: <sip:[email protected]>;tag=as7d2474b7
Call-ID: [email protected]
CSeq: 549 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 190

v=0
o=root 29046 29047 IN IP4 192.168.100.200
s=session
c=IN IP4 192.168.100.200
t=0 0
m=audio 15054 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

---
Nov  7 09:18:12 VERBOSE[30411] logger.c:     -- Attempting native bridge of SIP/17-094c8580 and SIP/13-094fd2d0
Nov  7 09:18:12 VERBOSE[29052] logger.c: DISCONNECT_B3_RESP ID=002 #0xf79b LEN=0012
  Controller/PLCI/NCCI            = 0x10104

Nov  7 09:18:12 VERBOSE[29050] logger.c:        > CAPI devicestate requested for dokom#02/0172XXXXXXX
Nov  7 09:18:12 VERBOSE[29052] logger.c: DISCONNECT_REQ ID=002 #0x6664 LEN=0017
  Controller/PLCI/NCCI            = 0x104
  AdditionalInfo                 
   BChannelinformation            = default
   Keypadfacility                 = default
   Useruserdata                   = default
   Facilitydataarray              = default

Nov  7 09:18:12 VERBOSE[29052] logger.c: DISCONNECT_CONF ID=002 #0x6664 LEN=0014
  Controller/PLCI/NCCI            = 0x104
  Info                            = 0x0

Nov  7 09:18:12 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.218:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK18222860;rport
From: <sip:[email protected]>;tag=as7ebc6b44
To: "Peter Montero" <sip:[email protected]>;tag=30345aa7c96095a7
Call-ID: [email protected]
CSeq: 865 NOTIFY
User-Agent: Grandstream GXP2000 1.1.4.18
Contact: <sip:[email protected]:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


Nov  7 09:18:12 VERBOSE[29057] logger.c: --- (11 headers 0 lines) ---
Nov  7 09:18:12 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Request 865: Match Found
Nov  7 09:18:12 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.180:5060: 
ACK sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK7f56af231d4a17eb
From: "Gabi Drescher" <sip:[email protected]>;tag=6aa7aad3d7ac10bc
To: <sip:*[email protected]>;tag=as62b5189b
Contact: <sip:[email protected]:5060;transport=udp>
Supported: path
Proxy-Authorization: Digest username="13", realm="asterisk", algorithm=MD5, uri="sip:*[email protected]", nonce="254e6c14", response="5f1233cbd40f64dcd0ca19a5566e5775"
Call-ID: [email protected]
CSeq: 28931 ACK
User-Agent: Grandstream GXP2000 1.1.4.18
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


Nov  7 09:18:12 VERBOSE[29057] logger.c: --- (13 headers 0 lines) ---
Nov  7 09:18:12 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Response 28931: Match Found
Nov  7 09:18:12 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.118:5060: 
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.118:5060;branch=z9hG4bK5f2673a660e53e16
From: "Daniel Dommisch" <sip:[email protected]>;tag=a9cab45464e7c758
To: <sip:[email protected]>;tag=as7d2474b7
Contact: <sip:[email protected]:5060;transport=udp>
Supported: path
Proxy-Authorization: Digest username="17", realm="asterisk", algorithm=MD5, uri="sip:[email protected]", nonce="783daeba", response="f3021a978f17907ac6c99ab91c2a1a57"
Call-ID: [email protected]
CSeq: 549 ACK
User-Agent: Grandstream GXP2000 1.1.4.18
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


Nov  7 09:18:12 VERBOSE[29057] logger.c: --- (13 headers 0 lines) ---
Nov  7 09:18:12 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Response 549: Match Found
Nov  7 09:18:12 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.227:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK32a96e01;rport
From: <sip:[email protected];user=phone>;tag=as1d9addff
To: "Marcel Linzer" <sip:[email protected];user=phone>;tag=d53f782fe57b0ab5
Call-ID: [email protected]
CSeq: 862 NOTIFY
User-Agent: Grandstream GXP2000 1.1.4.22
Contact: <sip:[email protected]:5060;transport=udp;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


Nov  7 09:18:12 VERBOSE[29057] logger.c: --- (11 headers 0 lines) ---
Nov  7 09:18:12 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Request 862: Match Found
Nov  7 09:18:12 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.106:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK66593524;rport
From: <sip:[email protected]>;tag=as6d9c78e9
To: "Nicole Killing" <sip:[email protected]>;tag=6989058941620062
Call-ID: [email protected]
CSeq: 822 NOTIFY
User-Agent: Grandstream GXP2000 1.1.4.18
Contact: <sip:[email protected]:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


Nov  7 09:18:12 VERBOSE[29057] logger.c: --- (11 headers 0 lines) ---
Nov  7 09:18:12 DEBUG[29057] chan_sip.c: Stopping retransmission on '[email protected]' of Request 822: Match Found
Nov  7 09:18:12 VERBOSE[29052] logger.c: INFO_IND ID=002 #0xf79c LEN=0019
  Controller/PLCI/NCCI            = 0x104
  InfoNumber                      = 0x4000
  InfoElement                     = <00 00 00 00>

Nov  7 09:18:12 VERBOSE[29052] logger.c: INFO_RESP ID=002 #0xf79c LEN=0012
  Controller/PLCI/NCCI            = 0x104

Nov  7 09:18:12 VERBOSE[29052] logger.c:     -- dokom#02: info element CHARGE in UNITS
Nov  7 09:18:12 VERBOSE[29052] logger.c: INFO_IND ID=002 #0xf79d LEN=0038
  Controller/PLCI/NCCI            = 0x104
  InfoNumber                      = 0x1c
  InfoElement                     = <91 a1 14 02 01 00 02 01 24>0<0c>0<0a a1 05>0<03 02 01 00 82 01 00>

Nov  7 09:18:12 VERBOSE[29052] logger.c: INFO_RESP ID=002 #0xf79d LEN=0012
  Controller/PLCI/NCCI            = 0x104

Nov  7 09:18:12 VERBOSE[29052] logger.c:     -- dokom#02: info element FACILITY
Nov  7 09:18:12 VERBOSE[29052] logger.c: INFO_IND ID=002 #0xf79e LEN=0015
  Controller/PLCI/NCCI            = 0x104
  InfoNumber                      = 0x804d
  InfoElement                     = default

Nov  7 09:18:12 VERBOSE[29052] logger.c: INFO_RESP ID=002 #0xf79e LEN=0012
  Controller/PLCI/NCCI            = 0x104

Nov  7 09:18:12 VERBOSE[29052] logger.c:     -- dokom#02: info element RELEASE
Nov  7 09:18:12 VERBOSE[29052] logger.c: DISCONNECT_IND ID=002 #0xf79f LEN=0014
  Controller/PLCI/NCCI            = 0x104
  Reason                          = 0x3490

Nov  7 09:18:12 VERBOSE[29052] logger.c: DISCONNECT_RESP ID=002 #0xf79f LEN=0012
  Controller/PLCI/NCCI            = 0x104

Nov  7 09:18:12 VERBOSE[29052] logger.c:        > dokom#02: CAPI INFO 0x3490: Normal call clearing
Nov  7 09:18:12 VERBOSE[29052] logger.c:   == dokom#02: Interface cleanup PLCI=0x104
Nov  7 09:18:16 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.118:5060: 
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.118:5060;branch=z9hG4bK1712a3d9714bb19b
From: "Daniel Dommisch" <sip:[email protected]>;tag=a9cab45464e7c758
To: <sip:[email protected]>;tag=as7d2474b7
Supported: path
Proxy-Authorization: Digest username="17", realm="asterisk", algorithm=MD5, uri="sip:[email protected]", nonce="783daeba", response="b3dd8f82234f3add2550118a2db6f3ea"
Call-ID: [email protected]
CSeq: 550 BYE
User-Agent: Grandstream GXP2000 1.1.4.18
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


Nov  7 09:18:16 VERBOSE[29057] logger.c: --- (12 headers 0 lines) ---
Nov  7 09:18:16 VERBOSE[29057] logger.c: Sending to 192.168.100.118 : 5060 (non-NAT)
Nov  7 09:18:16 VERBOSE[29057] logger.c: Transmitting (no NAT) to 192.168.100.118:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.118:5060;branch=z9hG4bK1712a3d9714bb19b;received=192.168.100.118
From: "Daniel Dommisch" <sip:[email protected]>;tag=a9cab45464e7c758
To: <sip:[email protected]>;tag=as7d2474b7
Call-ID: [email protected]
CSeq: 550 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
Nov  7 09:18:16 DEBUG[30411] channel.c: Didn't get a frame from channel: SIP/17-094c8580
Nov  7 09:18:16 DEBUG[30411] channel.c: Bridge stops bridging channels SIP/17-094c8580 and SIP/13-094fd2d0
Nov  7 09:18:16 DEBUG[30411] chan_sip.c: update_call_counter(13) - decrement call limit counter
Nov  7 09:18:16 VERBOSE[30411] logger.c: Scheduling destruction of call '[email protected]' in 32000 ms
Nov  7 09:18:16 VERBOSE[30411] logger.c: set_destination: Parsing <sip:[email protected]:5060;transport=udp> for address/port to send to
Nov  7 09:18:16 VERBOSE[30411] logger.c: set_destination: set destination to 192.168.100.180, port 5060
Nov  7 09:18:16 VERBOSE[30411] logger.c: Reliably Transmitting (no NAT) to 192.168.100.180:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK246624fd;rport
From: <sip:*[email protected]>;tag=as62b5189b
To: "Gabi Drescher" <sip:[email protected]>;tag=6aa7aad3d7ac10bc
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Nov  7 09:18:16 DEBUG[30411] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Nov  7 09:18:16 VERBOSE[30411] logger.c:   == Spawn extension (from-internal, 0172XXXXXXX, 3) exited non-zero on 'SIP/17-094c8580'
Nov  7 09:18:16 VERBOSE[30411] logger.c:     -- Executing Macro("SIP/17-094c8580", "hangupcall") in new stack
Nov  7 09:18:16 VERBOSE[30411] logger.c:     -- Executing ResetCDR("SIP/17-094c8580", "w") in new stack
Nov  7 09:18:16 VERBOSE[29050] logger.c: Reliably Transmitting (no NAT) to 192.168.100.106:5060:
NOTIFY sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK25804c83;rport
From: <sip:[email protected]>;tag=as6d9c78e9
To: "Nicole Killing" <sip:[email protected]>;tag=6989058941620062
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 823 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 206

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="721" state="full" entity="sip:[email protected]">
<dialog id="13">
<state>terminated</state>
</dialog>
</dialog-info>

---
Nov  7 09:18:16 VERBOSE[29050] logger.c:  Extension Changed 13 new state Idle for Notify User 14
Nov  7 09:18:16 VERBOSE[29050] logger.c: Reliably Transmitting (no NAT) to 192.168.100.218:5060:
NOTIFY sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK4fedbd4b;rport
From: <sip:[email protected]>;tag=as7ebc6b44
To: "Peter Montero" <sip:[email protected]>;tag=30345aa7c96095a7
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 866 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 206

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="764" state="full" entity="sip:[email protected]">
<dialog id="13">
<state>terminated</state>
</dialog>
</dialog-info>

---
Nov  7 09:18:16 VERBOSE[29050] logger.c:  Extension Changed 13 new state Idle for Notify User 18
Nov  7 09:18:16 VERBOSE[29050] logger.c: Reliably Transmitting (no NAT) to 192.168.100.227:5060:
NOTIFY sip:[email protected]:5060;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK30e43692;rport
From: <sip:[email protected];user=phone>;tag=as1d9addff
To: "Marcel Linzer" <sip:[email protected];user=phone>;tag=d53f782fe57b0ab5
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 863 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 206

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="761" state="full" entity="sip:[email protected]">
<dialog id="13">
<state>terminated</state>
</dialog>
</dialog-info>

---
Nov  7 09:18:16 VERBOSE[29050] logger.c:  Extension Changed 13 new state Idle for Notify User 27
Nov  7 09:18:16 DEBUG[30411] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Nov  7 09:18:16 DEBUG[30411] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES ('2007-11-07 09:17:56','\"Daniel Dommisch\" <17>','17','0172XXXXXXX','from-internal', 'SIP/17-094c8580','SIP/13-094fd2d0','Dial','CAPI/g1/0172XXXXXXX/b',20,4,'ANSWERED',3,'','1194423476.5821')
Nov  7 09:18:16 DEBUG[30411] app_macro.c: Executed application: ResetCDR
Nov  7 09:18:16 VERBOSE[30411] logger.c:     -- Executing NoCDR("SIP/17-094c8580", "") in new stack
Nov  7 09:18:16 NOTICE[30411] cdr.c: CDR on channel 'SIP/17-094c8580' not posted
Nov  7 09:18:16 NOTICE[30411] cdr.c: CDR on channel 'SIP/17-094c8580' lacks end
Nov  7 09:18:16 DEBUG[30411] app_macro.c: Executed application: NoCDR
Nov  7 09:18:16 DEBUG[30411] pbx.c: Expression result is '1'
Nov  7 09:18:16 VERBOSE[30411] logger.c:     -- Executing GotoIf("SIP/17-094c8580", "1?skiprg") in new stack
Nov  7 09:18:16 VERBOSE[30411] logger.c:     -- Goto (macro-hangupcall,s,6)
Nov  7 09:18:16 DEBUG[30411] app_macro.c: Executed application: GotoIf
Nov  7 09:18:16 DEBUG[30411] pbx.c: Expression result is '1'
Nov  7 09:18:16 VERBOSE[30411] logger.c:     -- Executing GotoIf("SIP/17-094c8580", "1?skipblkvm") in new stack
Nov  7 09:18:16 VERBOSE[30411] logger.c:     -- Goto (macro-hangupcall,s,9)
Nov  7 09:18:16 DEBUG[30411] app_macro.c: Executed application: GotoIf
Nov  7 09:18:16 DEBUG[30411] pbx.c: Expression result is '1'
Nov  7 09:18:16 VERBOSE[30411] logger.c:     -- Executing GotoIf("SIP/17-094c8580", "1?theend") in new stack
Nov  7 09:18:16 VERBOSE[30411] logger.c:     -- Goto (macro-hangupcall,s,11)
Nov  7 09:18:16 DEBUG[30411] app_macro.c: Executed application: GotoIf
Nov  7 09:18:16 VERBOSE[30411] logger.c:     -- Executing Hangup("SIP/17-094c8580", "") in new stack
Nov  7 09:18:16 VERBOSE[30411] logger.c:   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/17-094c8580' in macro 'hangupcall'
Nov  7 09:18:16 VERBOSE[30411] logger.c:   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/17-094c8580'
Nov  7 09:18:16 DEBUG[30411] chan_sip.c: update_call_counter(17) - decrement call limit counter
Nov  7 09:18:16 VERBOSE[29050] logger.c: Reliably Transmitting (no NAT) to 192.168.100.106:5060:
NOTIFY sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK63064369;rport
From: <sip:[email protected]>;tag=as62a30a70
To: "Nicole Killing" <sip:[email protected]>;tag=fde47de786013338
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 873 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 206

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="771" state="full" entity="sip:[email protected]">
<dialog id="17">
<state>terminated</state>
</dialog>
</dialog-info>

---
Nov  7 09:18:16 VERBOSE[29050] logger.c:  Extension Changed 17 new state Idle for Notify User 14
Nov  7 09:18:16 VERBOSE[29050] logger.c: Reliably Transmitting (no NAT) to 192.168.100.218:5060:
NOTIFY sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK206aeae8;rport
From: <sip:[email protected]>;tag=as6a3a4b1b
To: "Peter Montero" <sip:[email protected]>;tag=c5e801e8eea1aea1
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 913 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 206

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="811" state="full" entity="sip:[email protected]">
<dialog id="17">
<state>terminated</state>
</dialog>
</dialog-info>

---
Nov  7 09:18:16 VERBOSE[29050] logger.c:  Extension Changed 17 new state Idle for Notify User 18
Nov  7 09:18:16 VERBOSE[29050] logger.c: Reliably Transmitting (no NAT) to 192.168.100.227:5060:
NOTIFY sip:[email protected]:5060;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK7bebe192;rport
From: <sip:[email protected];user=phone>;tag=as1b1af5c5
To: "Marcel Linzer" <sip:[email protected];user=phone>;tag=661f621f20e76fd7
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 911 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 206

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="809" state="full" entity="sip:[email protected]">
<dialog id="17">
<state>terminated</state>
</dialog>
</dialog-info>

---
Nov  7 09:18:16 VERBOSE[29050] logger.c:  Extension Changed 17 new state Idle for Notify User 27
Nov  7 09:18:16 VERBOSE[29057] logger.c: 
<-- SIP read from 192.168.100.180:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK246624fd;rport
From: <sip:*[email protected]>;tag=as62b5189b
To: "Gabi Drescher" <sip:[email protected]>;tag=6aa7aad3d7ac10bc
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: Grandstream GXP2000 1.1.4.18
Contact: <sip:[email protected]:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0
 
sip debug waere nicht notwendig, da wird's nur unübersichtlich.
Anyway, ich sehe da kein Fehler. Das system verhaelt sich korrekt. Es wird der Channel, der gerade am klingeln ist übernommen. Wenn das bei bestimmten nicht passieren soll, dann musst du das entsprechend mit callgroup und pickupgroup einstellen.

Armin
 
Wie habe ich denn Call-Group und Pickup-Group zu verstehen?

ok, Pickup-Group ist mir noch geläufig: Alle User, die in einer genannten Pickup-Group sind, können die Anrufe der Pickup-Gruppe ziehen.

aber was hat das mit der Call-Group zu tun? Die ist doch nur dafür da um z.B. 2 ISDN-Controller als eine Gruppe zusammenzufassen oder nicht?
 
um das Thema nochmal aufzugreifen:

Ich verstehe den sinn nicht, dass man ausgehende gesrpäche innerhalb einer Gruppe pickuppen kann/sollte...

Wie genau sollten die Gruppen denn nun definiert werden, sodass zwar alle Telefone eingehende Gespräche ziehen können, ausgehende Gespräche jedoch nicht???

einfach alle Extensions in der Pickup-Group 1 belassen und dafür alle Extensions in die Call-Group 2 packen??
 
MarcelDeLuX schrieb:
Ich verstehe den sinn nicht, dass man ausgehende gesrpäche innerhalb einer Gruppe pickuppen kann/sollte...
Zum Vergrößern anklicken....

Aber genau darum geht es doch beim pickup. Wenn ein Telefon klingelt (und das ist nun mal ein ausgehender Kanal), dann will man diesen heranholen. Ein ankommendes Gespräch holt man nicht ran, sondern nur von dort, wo es klingelt.

MarcelDeLuX schrieb:
Wie genau sollten die Gruppen denn nun definiert werden, sodass zwar alle Telefone eingehende Gespräche ziehen können, ausgehende Gespräche jedoch nicht???

einfach alle Extensions in der Pickup-Group 1 belassen und dafür alle Extensions in die Call-Group 2 packen??
Zum Vergrößern anklicken....

Wie du das verteilst bleibt dir überlassen. Beim Pickup prüft Asterisk nur ob der Wert von deiner pickupgroup mit callgroup des klingelden Kanals passt.

Armin
 
Also wenn ich eingehende ISDN-Gespäche habe, wird doch die Call-Group genommen, die vom Controller übergeben wird, richtig?

und wenn ich die Extensions nun in eine andere Callgroup (z.B. 2) setze, dann können die Extensions in der Pickup-Gruppe 1 nur Gespräche annehmen, die durch den Controller in die Call-Group 1 übergeben wird, auch richtig??
 
Ich bin mir nicht sicher, ob ich dich jetzt richtig verstanden habe. Denn Du sprichst schon wieder von eingehenden Gespraechen. Pickup wird aber bei abgehenden (klingelden) Kanälen angewendet. Wenn du z.B. alle deine 'Telefone', bzw. dessen Kanäle auf callgroup=1 und pickupgroup=1 stellst, aber die CAPI Kanäle auf callgroup=2, dann dürfen die Telefone sich gegenseitig die ankommenden Gespräche heranholen, aber eben nicht die, die per ISDN/CAPI rausgehen.

Armin
 
ich denke, das ist genau das, was ich meine.
also setze ich bei den Controllern einfach die Callgroup auf 2 und alles sollte gut sein...

Danke für die umfangreiche Hilfe, Armin!

In Sachen Asterisk bin ich noch nicht so bewandert aber das Ding läuft für mittlerweile 25 Telefone doch erstaunlich gut, auch wenn ich bis vor ca. 6 Monaten überhaupt keine Ahnung davon hatte :rolleyes: :D

Gruß Marcel
 
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