[Asterisk-Dev] Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly

[James Sizemore]

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 at 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 at 192.168.7.250>;tag=19D8A640-5E9
> To: <sip:6155555917 at 192.168.53.68>
> Date: Wed, 06 Mar 2002 00:27:08 GMT
> Call-ID: BA75B677-2FCF11D6-806AB8FC-77734AF8 at 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 at 192.168.7.250>;party=calling;screen=yes;privacy=off
> Timestamp: 1015374428
> Contact: <sip:6155552115 at 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 at 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 at 192.168.7.250>;tag=19D8A640-5E9
> To: <sip:6155555917 at 192.168.53.68>;tag=as43478a8a
> Call-ID: BA75B677-2FCF11D6-806AB8FC-77734AF8 at 192.168.7.250
> CSeq: 101 INVITE
> User-Agent: Memphis ISDN-NET PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:6155555917 at 192.168.53.68>
> Content-Length: 0
> 
> 
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