[Asterisk-Users] sip.conf user entry for ViaTalk

[Ben Wern]

Try as I might, I can not get incoming calls from ViaTalk to match 
against my user entry. I have both peer and user entries, and incoming 
and outgoing calls work, but incoming calls do not move to my in-viatalk 
context (they stay in the default context.) Has anyone else managed to 
get this to work? My user entry looks like:
[viatalk-in]
username=1407965XXXX
context=viatalk-in
type=user
host=965.407.1.switch.vtnoc.net

I've also tried username=+1407965XXXX, host=67.15.74.73, 
host=67.15.74.73:5060, and host=dynamic. SIP debug from an incoming call 
shows:


<-- SIP read from 67.15.74.73:5060:
INVITE sip:s at 10.1.42.254 SIP/2.0
Via: SIP/2.0/UDP 67.15.74.73:5060;branch=z9hG4bK6169ed4e;rport
From: "Wern Ben" <sip:+1407694XXXX at 67.15.74.73>;tag=as7366fb31
To: <sip:s at 10.1.42.254>
Contact: <sip:+14076947004 at 67.15.74.73>
Call-ID: 0cc843c02ecebe49523ed5512e6d49bc at 67.15.74.73
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 18 Aug 2005 03:48:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 214
charon*CLI>
v=0
o=root 16334 16334 IN IP4 67.15.74.73
s=session
c=IN IP4 67.15.74.73
t=0 0
m=audio 21762 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

The next message indicates that it can't find a user or peer to match 
"67.15.74.73:5060", and moves to the default context. In the above 
examples, I've XXXX'd the last four numbers of live phone numbers. 
ViaTalk appears to be sending the incoming caller info (including plus 
sign) in the From: part, and not my userid.

Does anyone have any suggestions?

Ben Wern