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

[Glenn Dalgliesh]

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..