Glenn Dalgliesh
2004-May-05 15:03 UTC
[Asterisk-Users] sip.conf and SIP client host= not recognized in some cases
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@aaa.aaa.aaa.aaa SIP/2.0.. Via: SIP/2.0/UDP ccc.ccc.ccc.ccc:5060;branch=z9hG4bK7ab24dcc.. From: "asterisk" <sip:asterisk@ccc.ccc.ccc.ccc>;tag=as3a541e32.. To: <sip:4445552574@aaa.aaa.aaa.aaa>..Contact: <sip:asterisk@ccc.ccc.ccc.ccc>.. Call-ID: 75adb4aa7e9ff711120b14f518b44a1b@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@aaa.aaa.aaa.aaa:5060 SIP/2.0.. Call-ID: 7007601020188505154-1083791562@bbb.bbb.bbb.bbb.. From: sip:8889992264@bbb.bbb.bbb.bbb:5060;tag=12436.. To: sip:4445552574@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@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..
Karl Brose
2004-May-05 15:32 UTC
[Asterisk-Users] sip.conf and SIP client host= not recognized in some cases
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@aaa.aaa.aaa.aaa SIP/2.0.. >Via: SIP/2.0/UDP ccc.ccc.ccc.ccc:5060;branch=z9hG4bK7ab24dcc.. >From: "asterisk" <sip:asterisk@ccc.ccc.ccc.ccc>;tag=as3a541e32.. >To: <sip:4445552574@aaa.aaa.aaa.aaa>..Contact: ><sip:asterisk@ccc.ccc.ccc.ccc>.. >Call-ID: 75adb4aa7e9ff711120b14f518b44a1b@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@aaa.aaa.aaa.aaa:5060 SIP/2.0.. >Call-ID: 7007601020188505154-1083791562@bbb.bbb.bbb.bbb.. >From: sip:8889992264@bbb.bbb.bbb.bbb:5060;tag=12436.. >To: sip:4445552574@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@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.. > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
Andres
2004-May-05 18:51 UTC
[Asterisk-Users] sip.conf and SIP client host= not recognized in some cases
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@aaa.aaa.aaa.aaa SIP/2.0.. >Via: SIP/2.0/UDP ccc.ccc.ccc.ccc:5060;branch=z9hG4bK7ab24dcc.. >From: "asterisk" <sip:asterisk@ccc.ccc.ccc.ccc>;tag=as3a541e32.. >To: <sip:4445552574@aaa.aaa.aaa.aaa>..Contact: ><sip:asterisk@ccc.ccc.ccc.ccc>.. >Call-ID: 75adb4aa7e9ff711120b14f518b44a1b@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@aaa.aaa.aaa.aaa:5060 SIP/2.0.. >Call-ID: 7007601020188505154-1083791562@bbb.bbb.bbb.bbb.. >From: sip:8889992264@bbb.bbb.bbb.bbb:5060;tag=12436.. >To: sip:4445552574@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@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.. > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
brian k. west
2004-May-05 20:53 UTC
[Asterisk-Users] sip.conf and SIP client host= not recognized in some cases
I think he finally got it working... butin the process of me trying to recreate this bug I did find another one... that one has been fixed. bkw ----- Original Message ----- From: "Karl Brose" <khb@brose.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, May 05, 2004 7:29 PM Subject: Re: [Asterisk-Users] sip.conf and SIP client host= not recognized in some cases> If that were the case life be so much easier > > brian k. west wrote: > > >also in my backwards thinking type=user someone that calls us.. and > >type=peer is someone we call... > > > > > > > > Fact is the user/peer/friend & auth business is a mess, obscure > parameters, no documentation and BUGS > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >