[asterisk-dev] Broken SIP dialog if INFO messages receivedbeforefinal OK
Nick Lewis
Nick.Lewis at atltelecom.com
Tue Apr 20 04:26:27 CDT 2010
>U 2010/04/15 09:25:03.396617 192.168.1.201:5060 -> 192.168.1.202:5060
>SIP/2.0 200 OK
>Via: SIP/2.0/UDP
192.168.1.202;branch=z9hG4bKac943680397;received=192.168.1.202
>From: <sip: 1001 at 192.168.1.202>;tag=1c943596716
>To: <sip:*47427645918B at 192.168.1.201>;tag=as51fe15e5
>Call-ID: 943596255154201082455 at 192.168.1.202
>CSeq: 2 INVITE
>Server: Asterisk PBX 1.6.1.12
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
>Supported: replaces, timer
>Content-Type: application/sdp
>Content-Length: 210
>
>v=0
>o=root 1624991794 1624991794 IN IP4 192.168.1.201
>s=Asterisk PBX 1.6.1.12
>c=IN IP4 192.168.1.201
>t=0 0
>m=audio 19594 RTP/AVP 0
>a=rtpmap:0 PCMU/8000
>a=silenceSupp:off - - - -
>a=ptime:20
>a=sendrecv
I see what you mean. Asterisk is getting in a muddle
(I would be interested to see what chan_sip.c:respprep() thinks the
p->method value is at this point)
-- N_L
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