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Fuso

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Ich habe meinen Asterisk erfolgreich an eine Avaya Telefonanlage in der Firma registriert. Ich kann durch den Tunnel telefonate sowohl innerhalb als auch ausserhalb führen. Auch das Angerufen werden klappt. Allerdings muß ich mich immer sehr kurz fassen :) Offensichtlich schickt die Anlage in der Firma an meinen Asterisk ein BYE was die Verbindung dann ganz Ordnungsgemäß auf beiden Seiten trennt. Danach gibt es dann ein Re-Register und ich kann wiedertelefoniern. Registriere ich aber mein SNOM-320 direkt an der Anlage in der Firma kann ich ewig sprechen.

Hier noch ein Protokoll:
Code: 
Asterisk Ready.
*CLI> [Jan 26 19:59:48] NOTICE[18594]: chan_sip.c:9726 sip_reregister:    -- Re-registration for  [email protected]
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 172.25.28.241:5060:
REGISTER sip:172.25.28.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK3acf379b;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as6f6a5131
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: AVAYA_I55 VOIP1P68_rel_0227
Expires: 120
Contact: <sip:[email protected]>
Content-Length: 0


---

<--- SIP read from UDP://172.25.28.241:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK3acf379b;rport
From: <sip:[email protected]:5060>;tag=as6f6a5131
To: <sip:[email protected]:5060>;tag=00006ccb
Call-ID: [email protected]
CSeq: 102 REGISTER
WWW-Authenticate: Digest realm="172.25.28.241", domain="172.25.28.241", nonce="4d4f4e204a414e2032362031393a35393a33342032303039e7e9b6791b6a80881643a7bb66719406", algorithm=MD5
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Responding to challenge, registration to domain/host name 172.25.28.241
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 172.25.28.241:5060:
REGISTER sip:172.25.28.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK732b1cee;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as6287758a
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 REGISTER
User-Agent: AVAYA_I55 VOIP1P68_rel_0227
Authorization: Digest username="14", realm="172.25.28.241", algorithm=MD5, uri="172.25.28.241", nonce="4d4f4e204a414e2032362031393a35393a33342032303039e7e9b6791b6a80881643a7bb66719406", response="d3859594db9357a7d03c0b4eab7530e2"
Expires: 120
Contact: <sip:[email protected]>
Content-Length: 0


---

<--- SIP read from UDP://172.25.28.241:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK732b1cee;rport
From: <sip:[email protected]:5060>;tag=as6287758a
To: <sip:[email protected]:5060>;tag=00003d30
Call-ID: [email protected]
CSeq: 103 REGISTER
Contact: <sip:[email protected]:5060>;expires=100
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[Jan 26 19:59:48] NOTICE[18594]: chan_sip.c:15934 handle_response_register: Outbound Registration: Expiry for 172.25.28.241 is 120 sec (Scheduling reregistration in 105 s)

*CLI> 
<--- SIP read from UDP://172.25.28.241:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.25.28.241:5060;branch=z9hG4bK-80073E88-51EA-DD11-847C-00073B01ADA7
From: "003xxxxx67841" <sip:[email protected]:5060>;tag=000043b3
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1232999975 INVITE
Contact: "003xxxxx67841" <sip:[email protected]:5060;transport=udp>
Max-Forwards: 70
User-Agent: AVAYA_I55 VOIP1P68_rel_0227
Supported: 100rel
Allow: INVITE, ACK, CANCEL, BYE, REFER
Content-Type: application/sdp
Alert-Info: <cid:[email protected]>;avaya-cm-alert-type=external
Content-Length:   169

v=0
o=- 0 0 IN IP4 172.25.28.241
s=-
t=0 0
m=audio 20264 RTP/AVP 8 101
c=IN IP4 172.25.28.241
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (14 headers 9 lines) ---
Sending to 172.25.28.241 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '14' for '003xxxxx67841' from 172.25.28.241:5060
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 172.25.28.241:20264
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100e (gsm|ulaw|alaw|g722), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.25.28.241:20264
Looking for 14 in hls_in (domain 192.168.110.124)
list_route: hop: <sip:[email protected]:5060;transport=udp>

<--- Transmitting (no NAT) to 172.25.28.241:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.25.28.241:5060;branch=z9hG4bK-80073E88-51EA-DD11-847C-00073B01ADA7;received=172.25.28.241
From: "003xxxxx67841" <sip:[email protected]:5060>;tag=000043b3
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1232999975 INVITE
Server: AVAYA_I55 VOIP1P68_rel_0227
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.25.28.241:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.25.28.241:5060;branch=z9hG4bK-80073E88-51EA-DD11-847C-00073B01ADA7;received=172.25.28.241
From: "003xxxxx67841" <sip:[email protected]:5060>;tag=000043b3
To: <sip:[email protected]:5060>;tag=as53aca27d
Call-ID: [email protected]
CSeq: 1232999975 INVITE
Server: AVAYA_I55 VOIP1P68_rel_0227
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Length: 0


<------------>
Audio is at 192.168.110.124 port 19816
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.110.60:2190:
INVITE sip:[email protected]:2190;line=5c3rwcqn SIP/2.0
Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK2e0dd3b3;rport
Max-Forwards: 70
From: "HLS: 003xxxxx67841" <sip:[email protected]>;tag=as309f0f1b
To: <sip:[email protected]:2190;line=5c3rwcqn>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: AVAYA_I55 VOIP1P68_rel_0227
Date: Mon, 26 Jan 2009 18:59:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 355

v=0
o=root 257282876 257282876 IN IP4 192.168.110.124
s=Asterisk PBX SVN-branch-1.6.1-r158687
c=IN IP4 192.168.110.124
t=0 0
m=audio 19816 RTP/AVP 8 9 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
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

---

<--- SIP read from UDP://192.168.110.60:2190 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK2e0dd3b3;rport=5060
From: "HLS: 003xxxxx67841" <sip:[email protected]>;tag=as309f0f1b
To: <sip:[email protected]:2190;line=5c3rwcqn>;tag=cj8144it9i
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:2190;line=5c3rwcqn>;flow-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---

<--- Transmitting (no NAT) to 172.25.28.241:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.25.28.241:5060;branch=z9hG4bK-80073E88-51EA-DD11-847C-00073B01ADA7;received=172.25.28.241
From: "003xxxxx67841" <sip:[email protected]:5060>;tag=000043b3
To: <sip:[email protected]:5060>;tag=as53aca27d
Call-ID: [email protected]
CSeq: 1232999975 INVITE
Server: AVAYA_I55 VOIP1P68_rel_0227
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Length: 0


<------------>

<--- SIP read from UDP://192.168.110.60:2190 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK2e0dd3b3;rport=5060
From: "HLS: 003xxxxx67841" <sip:[email protected]>;tag=as309f0f1b
To: <sip:[email protected]:2190;line=5c3rwcqn>;tag=cj8144it9i
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:2190;line=5c3rwcqn>;flow-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP://192.168.110.60:2190 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK2e0dd3b3;rport=5060
From: "HLS: 003xxxxx67841" <sip:[email protected]>;tag=as309f0f1b
To: <sip:[email protected]:2190;line=5c3rwcqn>;tag=cj8144it9i
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:2190;line=5c3rwcqn>;flow-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP://192.168.110.60:2190 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK2e0dd3b3;rport=5060
From: "HLS: 003xxxxx67841" <sip:[email protected]>;tag=as309f0f1b
To: <sip:[email protected]:2190;line=5c3rwcqn>;tag=cj8144it9i
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:2190;line=5c3rwcqn>;flow-id=1
User-Agent: snom320/7.1.17
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, join, callerid
Content-Type: application/sdp
Content-Length: 398

v=0
o=root 20148482 20148483 IN IP4 192.168.110.60
s=call
c=IN IP4 192.168.110.60
t=0 0
m=audio 55148 RTP/AVP 8 9 3 0 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:g+9rX6XrweyfQyTeYZXejI3NmMrKbyQ7rtOTH6JO
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:3 gsm/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

<------------->
--- (13 headers 16 lines) ---
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.110.60:55148
Found audio description format pcma for ID 8
Found audio description format g722 for ID 9
Found audio description format gsm for ID 3
Found audio description format pcmu for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100e (gsm|ulaw|alaw|g722), peer - audio=0x100e (gsm|ulaw|alaw|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100e (gsm|ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.110.60:55148
list_route: hop: <sip:[email protected]:2190;line=5c3rwcqn>
set_destination: Parsing <sip:[email protected]:2190;line=5c3rwcqn> for address/port to send to
set_destination: set destination to 192.168.110.60, port 2190
Transmitting (no NAT) to 192.168.110.60:2190:
ACK sip:[email protected]:2190;line=5c3rwcqn SIP/2.0
Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK5467c14f;rport
Max-Forwards: 70
From: "HLS: 003xxxxx67841" <sip:[email protected]>;tag=as309f0f1b
To: <sip:[email protected]:2190;line=5c3rwcqn>;tag=cj8144it9i
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: AVAYA_I55 VOIP1P68_rel_0227
Content-Length: 0


---

Audio is at 192.168.110.124 port 12640
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.25.28.241:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.25.28.241:5060;branch=z9hG4bK-80073E88-51EA-DD11-847C-00073B01ADA7;received=172.25.28.241
From: "003xxxxx67841" <sip:[email protected]:5060>;tag=000043b3
To: <sip:[email protected]:5060>;tag=as53aca27d
Call-ID: [email protected]
CSeq: 1232999975 INVITE
Server: AVAYA_I55 VOIP1P68_rel_0227
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1163344232 1163344232 IN IP4 192.168.110.124
s=Asterisk PBX SVN-branch-1.6.1-r158687
c=IN IP4 192.168.110.124
t=0 0
m=audio 12640 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

<------------>

<--- SIP read from UDP://172.25.28.241:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.25.28.241:5060;branch=z9hG4bK-80073E88-51EA-DD11-847C-00073B01ADA7
From: "003xxxxx67841" <sip:[email protected]:5060>;tag=000043b3
To: <sip:[email protected]:5060>;tag=as53aca27d
Call-ID: [email protected]
CSeq: 1232999975 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: REGISTER

<--- SIP read from UDP://172.25.28.241:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.25.28.241:5060;branch=z9hG4bK-00E1DDC0-51EA-DD11-847C-00073B01ADA7
From: "003xxxxx67841" <sip:[email protected]:5060>;tag=000043b3
To: <sip:[email protected]:5060>;tag=as53aca27d
Call-ID: [email protected]
CSeq: 1232999976 BYE
Max-Forwards: 70
User-Agent: AVAYA_I55 VOIP1P68_rel_0227
Allow: INVITE, ACK, CANCEL, BYE, REFER
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 172.25.28.241 : 5060 (no NAT)

<--- Transmitting (no NAT) to 172.25.28.241:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.25.28.241:5060;branch=z9hG4bK-00E1DDC0-51EA-DD11-847C-00073B01ADA7;received=172.25.28.241
From: "003xxxxx67841" <sip:[email protected]:5060>;tag=000043b3
To: <sip:[email protected]:5060>;tag=as53aca27d
Call-ID: [email protected]
CSeq: 1232999976 BYE
Server: AVAYA_I55 VOIP1P68_rel_0227
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:[email protected]:2190;line=5c3rwcqn> for address/port to send to
set_destination: set destination to 192.168.110.60, port 2190
Reliably Transmitting (no NAT) to 192.168.110.60:2190:
BYE sip:[email protected]:2190;line=5c3rwcqn SIP/2.0
Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK2f6292ab;rport
Max-Forwards: 70
From: "HLS: 003xxxxx67841" <sip:[email protected]>;tag=as309f0f1b
To: <sip:[email protected]:2190;line=5c3rwcqn>;tag=cj8144it9i
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: AVAYA_I55 VOIP1P68_rel_0227
Content-Length: 0


---

<--- SIP read from UDP://192.168.110.60:2190 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.110.124:5060;branch=z9hG4bK2f6292ab;rport=5060
From: "HLS: 003xxxxx67841" <sip:[email protected]>;tag=as309f0f1b
To: <sip:[email protected]:2190;line=5c3rwcqn>;tag=cj8144it9i
Call-ID: [email protected]
CSeq: 103 BYE
Contact: <sip:[email protected]:2190;line=5c3rwcqn>;flow-id=1
User-Agent: snom320/7.1.17
RTP-RxStat: Total_Rx_Pkts=4592,Rx_Pkts=4592,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=4598,Tx_Pkts=4598,Remote_Tx_Pkts=4501
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: INVITE
Really destroying SIP dialog '[email protected]' Method: BYE

und meine SIP.CONF (ein paar 1&1 betreffende Daten habe ich entfernt):
Code: 
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
language=de
externhost=lls.dyndns.org
externrefresh = 15
localnet=192.168.110.0/255.255.255.0
localnet=172.25.28.0/255.255.255.0
;qualify=yes
disallow=all
allow=g722
allow=gsm
allow=ulaw
allow=alaw
;allow=g729
;allow=slinear
nat=no
allowsubscribe = yes
notifyringing = yes
notifyhold = yes
limitonpeers = yes
registerattempts = 0
registertimeout = 0
maxexpirey=3600
defaultexpirey=120
musiconhold=mp3
musicclass=mp3
useragent=AVAYA_I55 VOIP1P68_rel_0227

;--------------------------- SIP DEBUGGING ---------------------------------------------------
sipdebug = on                   ; Turn on SIP debugging by default, from
                                ; the moment the channel loads this configuration
;recordhistory=yes              ; Record SIP history by default
                                ; (see sip history / sip no history)
dumphistory=no                  ; Dump SIP history at end of SIP dialogue
                                ; SIP history is output to the DEBUG logging channel

register => 14:[email protected]/14

[14]
type=friend
defaultuser=14
fromuser=14
secret=4141
host=172.25.28.241
fromdomain=172.25.28.241
context=hls_in
insecure=port,invite
caninvite=no
canreinvite=no
;nat=no

[10]
callerid=Privat <10>
host=dynamic
domain=192.168.110.124
user=10
secret=1313
type=friend
mailbox=10
nat=no
caninvite=no
canreinvite=no
context=default
subscribecontext=default
call-limit = 10
callgroup = 2
pickupgroup = 2

[11]
callerid=Büro <11>
host=dynamic
domain=192.168.110.124
user=11
secret=7990
type=friend
mailbox=11
nat=no
caninvite=no
canreinvite=no
context=default
subscribecontext=default
call-limit = 10
callgroup = 2
pickupgroup = 2

[12]
callerid=Büro Handy <12>
host=dynamic
domain=192.168.110.124
user=12
secret=7990
type=friend
mailbox=11
vmexten=11
nat=no
caninvite=no
canreinvite=no
context=default
subscribecontext=default
call-limit = 10
callgroup = 2
pickupgroup = 2

Ich hoffe ihr könnt mir auch bei diesem Problem wieder helfen.

Viele Grüße
Frank
 
Hat denn keiner eine Idee?
 
Ich habe die Hoffnung noch nicht aufgegeben dass doch noch ein Spezialist auftaucht der mir zumindest einen Tip geben kann.

Viele Grüße
Fuso
 
Hallo!

Versuch es mal mit der option canreinvite=nat in der sip.conf!
 
Das war es leider auch nicht :(
 
ja sorry das ding heisst auch nonat und nicht nat ,-)! Sorry...
 
Kein Problem. Hab es ausprobiert. Leider geht es auch so nicht.

Ist dies währen einer Verbindung eigentlich normal?

Code: 
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER
)

und wenig später:
Code: 
Really destroying SIP dialog '[email protected]' Method: REGISTER


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