Re: [asterisk-users] Mobile answer machine cut off
On 25/08/10 7:35 PM, Julian Lyndon-Smith wrote: 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 ? Just something happening when the call gets cut off. Is there any DTMF being transmitted, why was the call disconnected etc. Or just take a snippet and put it up on pastebin/post here -- 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
Re: [asterisk-users] Mobile answer machine cut off
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-Smithaster...@dotr.com 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
Re: [asterisk-users] Mobile answer machine cut off
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 li...@venturevoip.com 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-Smithaster...@dotr.com 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
[asterisk-users] Mobile answer machine cut off
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 ? Thanks Julian -- 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
Re: [asterisk-users] Mobile answer machine cut off
Crap, sorry, meant to add that we are running 1.4 svn head Julian On 21 August 2010 23:38, Julian Lyndon-Smith aster...@dotr.com 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 ? Thanks Julian -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- 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
