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

brian k. west brian at bkw.org
Wed May 5 18:18:46 MST 2004


Its not a bug ... if you aren't sending any auth data but you are sending a
username you might want to added insecure=yes so it won't try to auth. and
thus land in the general context.

bkw

----- Original Message ----- 
From: "Karl Brose" <khb at brose.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, May 05, 2004 4:32 PM
Subject: Re: [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 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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