James Sizemore
2005-Dec-27 11:28 UTC
[Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly
when my Cisco IAD send a call to my Asterisk gateway the gateway treats it as if I don't have a peer statement in sip.conf, when I do. Here are the first two packets, notice the "Found no matching peer or user for '192.168.7.250:50437'" on the second packet. Any one seen this before, or have a clue as to the problem? Asterisk 1.0.9 sip.conf: [bna-vonx-iad] type=friend context=trusted-out host=192.168.7.250 canreinvite=no Sip read: INVITE sip:6155555917@192.168.53.68:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60 From: "James Sizemore" <sip:6155552115@192.168.7.250>;tag=19D8A640-5E9 To: <sip:6155555917@192.168.53.68> Date: Wed, 06 Mar 2002 00:27:08 GMT Call-ID: BA75B677-2FCF11D6-806AB8FC-77734AF8@192.168.7.250 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 3128236623-802099670-2154346748-2004044536 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: "James Sizemore" <sip:6155552115@192.168.7.250>;party=calling;screen=yes;privacy=off Timestamp: 1015374428 Contact: <sip:6155552115@192.168.7.250:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 191 v=0 o=CiscoSystemsSIP-GW-UserAgent 6047 8216 IN IP4 192.168.7.250 s=SIP Call c=IN IP4 192.168.7.250 t=0 0 m=audio 16434 RTP/AVP 0 c=IN IP4 192.168.7.250 a=rtpmap:0 PCMU/8000 a=ptime:20 20 headers, 9 lines Using latest request as basis request Sending to 192.168.7.250 : 5060 (non-NAT) Found no matching peer or user for '192.168.7.250:50437' Found RTP audio format 0 Peer audio RTP is at port 192.168.7.250:16434 Found description format PCMU Capabilities: us - 0x78e (gsm|ulaw|alaw|lpc10|g729|speex|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 6155555917 in default list_route: hop: <sip:6155552115@192.168.7.250:5060> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60 From: "James Sizemore" <sip:6155552115@192.168.7.250>;tag=19D8A640-5E9 To: <sip:6155555917@192.168.53.68>;tag=as43478a8a Call-ID: BA75B677-2FCF11D6-806AB8FC-77734AF8@192.168.7.250 CSeq: 101 INVITE User-Agent: Memphis ISDN-NET PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:6155555917@192.168.53.68> Content-Length: 0
James Sizemore
2005-Dec-27 13:15 UTC
[Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly
I think I found what is munging up the peer lookup: This call from another Asterisk box starts: <-- SIP read from 192.168.69.254:5060: The peer lookup that fail reads: <-- SIP read from 192.168.7.250:52141: Asterisk seem to be thrown off by the fact that the return port is not 5060, and fails the peer lookup. This is a * bug then. I have documented it with both 1.0.9 and 1.2.1. Time to dig through the sip code. James Sizemore wrote:> when my Cisco IAD send a call to my Asterisk gateway the gateway treats > it as if I don't have a peer statement in sip.conf, when I do. Here are > the first two packets, notice the "Found no matching peer or user for > '192.168.7.250:50437'" on the second packet. Any one seen this before, > or have a clue as to the problem? Asterisk 1.0.9 > > sip.conf: > [bna-vonx-iad] > type=friend > context=trusted-out > host=192.168.7.250 > canreinvite=no > > > Sip read: > INVITE sip:6155555917@192.168.53.68:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60 > From: "James Sizemore" <sip:6155552115@192.168.7.250>;tag=19D8A640-5E9 > To: <sip:6155555917@192.168.53.68> > Date: Wed, 06 Mar 2002 00:27:08 GMT > Call-ID: BA75B677-2FCF11D6-806AB8FC-77734AF8@192.168.7.250 > Supported: 100rel,timer > Min-SE: 1800 > Cisco-Guid: 3128236623-802099670-2154346748-2004044536 > User-Agent: Cisco-SIPGateway/IOS-12.x > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER > CSeq: 101 INVITE > Max-Forwards: 6 > Remote-Party-ID: "James Sizemore" > <sip:6155552115@192.168.7.250>;party=calling;screen=yes;privacy=off > Timestamp: 1015374428 > Contact: <sip:6155552115@192.168.7.250:5060> > Expires: 180 > Allow-Events: telephone-event > Content-Type: application/sdp > Content-Length: 191 > > v=0 > o=CiscoSystemsSIP-GW-UserAgent 6047 8216 IN IP4 192.168.7.250 > s=SIP Call > c=IN IP4 192.168.7.250 > t=0 0 > m=audio 16434 RTP/AVP 0 > c=IN IP4 192.168.7.250 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > > 20 headers, 9 lines > Using latest request as basis request > Sending to 192.168.7.250 : 5060 (non-NAT) > Found no matching peer or user for '192.168.7.250:50437' > Found RTP audio format 0 > Peer audio RTP is at port 192.168.7.250:16434 > Found description format PCMU > Capabilities: us - 0x78e (gsm|ulaw|alaw|lpc10|g729|speex|ilbc), peer - > audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) > Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined > - 0x0 (nothing) > Looking for 6155555917 in default > list_route: hop: <sip:6155552115@192.168.7.250:5060> > Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60 > From: "James Sizemore" <sip:6155552115@192.168.7.250>;tag=19D8A640-5E9 > To: <sip:6155555917@192.168.53.68>;tag=as43478a8a > Call-ID: BA75B677-2FCF11D6-806AB8FC-77734AF8@192.168.7.250 > CSeq: 101 INVITE > User-Agent: Memphis ISDN-NET PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:6155555917@192.168.53.68> > Content-Length: 0 > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users