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@200.200.200.200:5060;user=phone>;tag=4fdc9d0e-1e600f94-ed7e6 23f To: <sip:8014377860@67.137.28.10:5060;user=phone>;tag=as4fc8aa8a Call-ID: 3a8530f4-43cb1-1e600f94@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
Olle E Johansson
2006-Feb-24 07:52 UTC
[Asterisk-Users] Re: [asterisk-dev] Possible Bug in SIP Stack.
Chris Modesitt wrote:> 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@200.200.200.200:5060;user=phone>;tag=4fdc9d0e-1e600f94-ed7e623f > > To: <sip:8014377860@67.137.28.10:5060;user=phone>;tag=as4fc8aa8a > > Call-ID: 3a8530f4-43cb1-1e600f94@200.200.200.200 > > CSeq: 5466974 INVITE > > User-Agent: Asterisk PBX > > > > I still can make outbound calls with no-problems, any ideas? > >Can you get SIP debug logs from a call setup with Asterisk 1.0.10 and 1.2.4 so we can compare them and see what happened? Thanks /Olle
Olle E Johansson
2006-Feb-24 08:30 UTC
[Asterisk-Users] Re: [asterisk-dev] Possible Bug in SIP Stack.
Chris Modesitt wrote:> 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@200.200.200.200:5060;user=phone>;tag=4fdc9d0e-1e600f94-ed7e623f > > To: <sip:8014377860@67.137.28.10:5060;user=phone>;tag=as4fc8aa8a > > Call-ID: 3a8530f4-43cb1-1e600f94@200.200.200.200 > > CSeq: 5466974 INVITE > > User-Agent: Asterisk PBX > > > > I still can make outbound calls with no-problems, any ideas? > >Can you get SIP debug logs from a call setup with Asterisk 1.0.10 and 1.2.4 so we can compare them and see what happened? Thanks /Olle _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev