Hi All, I'm looking for how to enable SIP Markers, or specifically, how to have the TIME reset when a call route changes. I'm debugging an issue, where a sip client we have switching to one-way-audio, when an asterisk server fruther down the call path dials out to the PSTN. Scenario is: SIP Client -> A*k1 -> A*k2 -> PSTN Provider/Gradwell -> O2 -> Mobile - the SIP client dials on O2 mobile, call goes out to A*1. - A*1 Dials out to A*k2 as A*k2 is the gateway to PSTN providers and normal office phones. - A*k2 dials some local Cisco phones, then on no answer plays an audio file, so call is ANSWERED. - A*k2 then Dials out to gradwell, to a mobile phone number. - Gradwell takes the call, routes it via PSTN. My problem, is that at the point where the O2 mobile accepts the call, I get one-way audio. (SIP Client outbound, nothing inbound). Tracing the RTP stream all the way back, I can see that audio makes it all the way to the SIP Client. However, we notice that at the point where the O2 mobile answers, the TIME= value of the packet jumps significantly, say from 119248 to 1518324408. Talking to the sip client developer, they say that I need to enable SIP Markers on the server (I guess A*k2), so that if the stream source changes then the timers are reset. Does this sound right, and if so, how do I do that ? I am running an older load on A*K2 of 1.4.18, and 1.4.15svn (privately compiled to add an extra codec) on A*k1. I can look into upgrading these, but the developer thinks it's just a missing config on Asterisk. Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090610/2b9a231e/attachment.htm
(resend as apparently I was blocked) Hi All, I'm looking for how to enable SIP Markers, or specifically, how to have the TIME reset when a call route changes. I'm debugging an issue, where a sip client we have switching to one-way-audio, when an asterisk server fruther down the call path dials out to the PSTN. Scenario is: SIP Client -> A*k1 -> A*k2 -> PSTN Provider/Gradwell -> O2 -> Mobile - the SIP client dials on O2 mobile, call goes out to A*1. - A*1 Dials out to A*k2 as A*k2 is the gateway to PSTN providers and normal office phones. - A*k2 dials some local Cisco phones, then on no answer plays an audio file, so call is ANSWERED. - A*k2 then Dials out to gradwell, to a mobile phone number. - Gradwell takes the call, routes it via PSTN. My problem, is that at the point where the O2 mobile accepts the call, I get one-way audio. (SIP Client outbound, nothing inbound). Tracing the RTP stream all the way back, I can see that audio makes it all the way to the SIP Client. However, we notice that at the point where the O2 mobile answers, the TIME= value of the packet jumps significantly, say from 119248 to 1518324408. Talking to the sip client developer, they say that I need to enable SIP Markers on the server (I guess A*k2), so that if the stream source changes then the timers are reset. Does this sound right, and if so, how do I do that ? I am running an older load on A*K2 of 1.4.18, and 1.4.15svn (privately compiled to add an extra codec) on A*k1. I can look into upgrading these, but the developer thinks it's just a missing config on Asterisk. Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090610/f22b7832/attachment.htm