[Asterisk-Users] Asterisk SIP architecture question
When asterisk is setup to allow SIP users to send media end-to-end (canreinvite=yes), can cdr info still be reliable, considering one of the end-user devices could go down leaving the call open. This is assuming you are using a third party pstn and not asterisk for pstn. Does asterisk have any mechanism for detecting and disconnecting hung calls in this type of scenario? regards, David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
David Thomas wrote: When asterisk is setup to allow SIP users to send media end-to-end (canreinvite=yes), can cdr info still be reliable, considering one of the end-user devices could go down leaving the call open. This is assuming you are using a third party pstn and not asterisk for pstn. Does asterisk have any mechanism for detecting and disconnecting hung calls in this type of scenario? No, not accurately. Asterisk may not receive any information in this case. The best bet is that if you are doing reinvite to make an agreement with your VoIP provider to get a copy of their CDRs -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
Matt Riddell wrote: No, not accurately. Asterisk may not receive any information in this case. The best bet is that if you are doing reinvite to make an agreement with your VoIP provider to get a copy of their CDRs Sorry, this advice is bogus :-( SIP re-INVITEs do _not_ affect the CDRs in any way, period. They only affect the media streams. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
Kevin P. Fleming wrote: Matt Riddell wrote: No, not accurately. Asterisk may not receive any information in this case. The best bet is that if you are doing reinvite to make an agreement with your VoIP provider to get a copy of their CDRs Sorry, this advice is bogus :-( So how does Asterisk know that the media stream has been disconnected between the two remote hosts? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
Matt Riddell wrote: So how does Asterisk know that the media stream has been disconnected between the two remote hosts? It doesn't... nor does any other SIP softswitch. See my other reply for a possible solution. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
Kevin P. Fleming wrote: Matt Riddell wrote: So how does Asterisk know that the media stream has been disconnected between the two remote hosts? It doesn't... nor does any other SIP softswitch. See my other reply for a possible solution. I agree that you could code a fix, but saying my advice is bogus because you could code a fix for Asterisk to avoid it is slightly wrong. The fact remains, if you need *very* accurate cdr's then you either don't do canreinvite=yes for the peer or you code something so that Asterisk notices that the rtp has stopped. The fact remains that without these, the most accurate CDR is going to come from the provider. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
Matt Riddell wrote: The fact remains, if you need *very* accurate cdr's then you either don't do canreinvite=yes for the peer or you code something so that Asterisk notices that the rtp has stopped. The fact remains that without these, the most accurate CDR is going to come from the provider. OK, I'll agree with that, except for one thing: if the provider notices that the RTP has stopped and wants to kill the call, it will send BYE to Asterisk and Asterisk will close the channels and update the CDR. The only time this will be an issue is if _both_ ends disappear and never send any signaling to Asterisk. However, in the general case of not being concerned so much about the peer going away and losing CDR information for _one_ call, using reinvites does _not_ impact the quality of the softswitch's (Asterisk) CDRs. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
Kevin P. Fleming wrote: However, in the general case of not being concerned so much about the peer going away and losing CDR information for _one_ call, using reinvites does _not_ impact the quality of the softswitch's (Asterisk) CDRs. Agreed. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
Kevin P. Fleming wrote: Matt Riddell wrote: So how does Asterisk know that the media stream has been disconnected between the two remote hosts? It doesn't... nor does any other SIP softswitch. See my other reply for a possible solution. ...or implement the SIP timer extension. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP architecture question
Matt Riddell wrote: Kevin P. Fleming wrote: Matt Riddell wrote: So how does Asterisk know that the media stream has been disconnected between the two remote hosts? It doesn't... nor does any other SIP softswitch. See my other reply for a possible solution. I agree that you could code a fix, but saying my advice is bogus because you could code a fix for Asterisk to avoid it is slightly wrong. The fact remains, if you need *very* accurate cdr's then you either don't do canreinvite=yes for the peer or you code something so that Asterisk notices that the rtp has stopped. The fact remains that without these, the most accurate CDR is going to come from the provider. If the audio goes through asterisk without re-invites, you could use the rtptimeouts to detect a dead phone. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
