[asterisk-dev] 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-dev/attachments/20060224/454530b1/attachment.htm


More information about the asterisk-dev mailing list