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

Karl Brose khb at brose.com
Wed May 5 15:32:14 MST 2004


Yes, this is a bug.   At least I call it a BUG.
We had a similar issue in IAX and I finally got it acknowledged and fixed.
It's been reported before for SIP, I believe, but it's apparently not 
acknowledged
as a bug again.
In your example you have two unauthenticated friends, meaning you have 
no secret
to authenticate against. The host ip address does not authenticate.
The call that gets accepted into the intended context will be the one 
to  only the last
friend in the list.   I think you got that the other way around, but 
it's probably not so.
Please check again and test by switching the sequence of your friends.

It's a mind boggling bug for starters particularly, if you are build up 
a dial plan and
add clients and all of a sudden what worked before stops for no obvious 
reason.
The only way out is to read (and understand!) the source code.


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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