[Asterisk-Users] Possible Bug in SIP Stack.
Chris Modesitt
chris at octelecom.net
Fri Feb 24 07:34:55 MST 2006
I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is APX
8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server. When I use
Asterisk version 10.0.10 everything works perfectly, however when I use
1.2.4 I lose the ability to receive calls from the PSTN. All I get is the
following error in my SIP Proxies error logs:
SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, don't
know where to send responseSIP/2.0 180 Ringing
From: "MODESITT,CHRIS "
<sip:8013793000 at 200.200.200.200:5060;user=phone>;tag=4fdc9d0e-1e600f94-ed7e6
23f
To: <sip:8014377860 at 67.137.28.10:5060;user=phone>;tag=as4fc8aa8a
Call-ID: 3a8530f4-43cb1-1e600f94 at 200.200.200.200
CSeq: 5466974 INVITE
User-Agent: Asterisk PBX
I still can make outbound calls with no-problems, any ideas?
Thanks
Chris
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060224/454530b1/attachment.htm
More information about the asterisk-users
mailing list