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

Wed May 5 18:32:34 MST 2004

also in my backwards thinking type=user someone that calls us.. and
type=peer is someone we call...

So friend and insecure=yes might fix it.

bkw

----- Original Message ----- 
From: "brian k. west" <brian at bkw.org>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, May 05, 2004 7:18 PM
Subject: Re: [Asterisk-Users] sip.conf and SIP client host= not recognized
in some cases

> 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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