[Asterisk-Users] sip.conf and SIP client host= not recognized in some cases

[Andres]

the problem is B has a different source port (it is not 5060).  Notice 
that A does have source port=5060.   sip.conf is not flexible enough to 
recognize users with dynamic source ports (unless they are behind NAT 
and you have nat=yes)


Glenn Dalgliesh wrote:

>I am seeing an issue with getting certain sip devices to be recognized as
>defined SIP clients host= in the sip.conf and the only deference that I can
>find btw sources that work and don't work is that devices that send packets
>with an Initial Via header of themselves appears to work and pick the
>context correctly but those that don't have the Via just get dropped in the
>context of the [General] context in sip.conf. Anyone have any similar
>experiences?
>
>Call comes from ccc.ccc.ccc.ccc to Asterisk from Invite in Example A in ends
>up in [inbound] context but in Example B it ends up in [default]. The only
>difference I can find btw these two examples is the fact that A has a VIA
>record and B doesn't. Can anyone confirm this behavior or at least explain
>it? (Used today's CVS)
>
>/etc/asterisk/sip.conf
>[general]
>port = 5060                     ; Port to bind to
>bindaddr = aaa.aaa.aaa.aaa               ; Address to bind to
>context = default            ; Default for incoming calls
>
>[carriera]
>type=friend
>host=ccc.ccc.ccc.ccc
>context=inbound
>
>[carrierb]
>type=friend
>host=bbb.bbb.bbb.bbb
>context=inbound
>
>/etc/asterisk/extensions.conf
>[inbound]
>exten => _.,1,Playback,tt-monkeysintro
>
>[default]
>exten => _.,1,Congestion
>
>
>Example A:
>U ccc.ccc.ccc.ccc:5060 -> aaa.aaa.aaa.aaa:5060
>  INVITE sip:4445552574 at aaa.aaa.aaa.aaa SIP/2.0..
>Via: SIP/2.0/UDP ccc.ccc.ccc.ccc:5060;branch=z9hG4bK7ab24dcc..
>From: "asterisk" <sip:asterisk at ccc.ccc.ccc.ccc>;tag=as3a541e32..
>To: <sip:4445552574 at aaa.aaa.aaa.aaa>..Contact:
><sip:asterisk at ccc.ccc.ccc.ccc>..
>Call-ID: 75adb4aa7e9ff711120b14f518b44a1b at ccc.ccc.ccc.ccc..
>CSeq: 102 INVITE..
>User-Agent: Asterisk PBX..Date: Wed, 05 May 2004 21:08:44 GMT..Allow:
>INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..
>Content-Type: application/sdp..
>Content-Length: 211..
>..
>v=0..
>o=root 13122 13122 IN IP4 ccc.ccc.ccc.ccc..
>s=session..
>c=IN IP4 ccc.ccc.ccc.ccc..
>t=0 0..m=audio 18980 RTP/AVP 0 3 8..
>a=rtpmap:0 PCMU/8000..
>a=rtpmap:3 GSM/8000..
>a=rtpmap:8 PCMA/8000..
>a=silenceSupp:off - - - -..
>#
>
>Example B:
>U bbb.bbb.bbb.bbb:44151 -> aaa.aaa.aaa.aaa:5060
>  INVITE sip:4445552574 at aaa.aaa.aaa.aaa:5060 SIP/2.0..
>Call-ID: 7007601020188505154-1083791562 at bbb.bbb.bbb.bbb..
>From: sip:8889992264 at bbb.bbb.bbb.bbb:5060;tag=12436..
>To: sip:4445552574 at aaa.aaa.aaa.aaa:5060..
>Content-Length: 251..
>Content-Type: application/sdp..
>CSeq: 1 INVITE..
>Via: SIP/2.0/UDP bbb.bbb.bbb.bbb:5060;branch=z9hG4bK-61400000000
>  03442-414d9af3..
>Contact: sip:8889992264 at bbb.bbb.bbb.bbb:5060..
>Supported: 100rel..
>Max-Forwards: 70..
>..
>v=0..
>o=MG4000|1.0 111 12345 IN IP4 65.77.154.6..
>s=-..
>c=IN IP4 65.77.154.6..
>t=0 0..
>m=audio 7824 RTP/AVP 18 0 102 103..
>a=rtpmap:102 G.723.1a-L/8000..
>a=rtpmap:103 telephone-event/8000..
>a=fmtp:103 0-15..
>a=X-sqn: 0..a=X-cap: 1
>image udptl t38..
>a=ptime:10..
>
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