Hey Matt, thanks for the response. I know it sounds impossible. Hell, I sound like a user :) But it *is* happening. And only on the cisco phones. We're trying to lab it up right now. What should I be looking for in the sip debug ?
Julian On 25 August 2010 08:17, Matt Riddell <[email protected]> wrote: > On 22/08/10 10:38 AM, Julian Lyndon-Smith wrote: >> Crap, sorry, meant to add that we are running 1.4 svn head >> >> Julian >> >> On 21 August 2010 23:38, Julian Lyndon-Smith<[email protected]> wrote: >>> We are having some strange issue where a call from asterisk dials a >>> mobile number. If the number answers, we put the call through to an >>> agent SIP phone. All works fine. >>> >>> If, however, the call goes straight through to the mobiles voicemail >>> service *and* the agent phone is a Cisco 79xx, then the call is >>> dropped (from the mobile end) about 1 second into the call. If the SIP >>> phone is an Aastra9133i, then there is no problem. >>> >>> Has anyone seen anything like this ? > > Heh, seems impossible! > > Um, maybe the voicemail beep is the same tone as a * and * is used to > disconnect a call or something? > > Try doing a SIP debug and see what turns up. Also make sure it's 100% > repeatable :D > > -- > Cheers, > > Matt Riddell > _______________________________________________ > > http://www.venturevoip.com/news.php (Daily Asterisk News) > http://www.venturevoip.com/exchange.php (Full ITSP Solution) > http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Follow "Ode To Politics" by HB Tasker at http://twitter.com/HBTasker -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
