Re: [Asterisk-Users] Re: Asterisk SIP architecture question
David Thomas wrote: > Does asterisk have support for SIP session timers? > No. /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
[Asterisk-Users] Re: Asterisk SIP architecture question
Does asterisk have support for SIP session timers? David On 11/24/05, Olle E. Johansson <[EMAIL PROTECTED]> wrote: > 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 > ___ --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] Re: Asterisk SIP architecture question
Kevin P. Fleming wrote: > David Thomas wrote: > >> Is the CDR accounting done based on SIP signaling? If a UA is talking >> (RTP) to a third party PSTN gateway, isn't it at risk if say the UA >> loses power. How will asterisk know the call has ended if it is not >> involved in the media path. The idea is this.. I want to use >> canreinvite =yes to force users to talk end-to-end to preserve >> bandwidth, but I can see the potential for hung calls if asterisk >> never get the BYE from a UA in the event the ATA gets unplugged or >> somehow loses power. > > > That is the case in every SIP network, Asterisk is not unique in that > regard. > > I would suggest that you could make a modification to chan_sip so that > if the peer goes 'unreachable' (as determined by using qualify=yes) than > any existing calls involved with that peer are immediately hung up; that > would take care of this problem. That would be a good addition. Optional of course. /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] Re: Asterisk SIP architecture question
If you use qualify=yes to determine whether that device is alive or not, then it won't be very accurate as every now and then, the device may fail to reply to the SIP OPTIONS packet due to reasons other than it is really offline. If you are linked to a PSTN GW, I would believe that GW will monitor the RTP stream and then initiate a BYE if it sees no RTP packets coming in. That way Asterisk will receive the proper disconnect signal in a canreinvite=yes scenario. David Kevin P. Fleming wrote: David Thomas wrote: Is the CDR accounting done based on SIP signaling? If a UA is talking (RTP) to a third party PSTN gateway, isn't it at risk if say the UA loses power. How will asterisk know the call has ended if it is not involved in the media path. The idea is this.. I want to use canreinvite =yes to force users to talk end-to-end to preserve bandwidth, but I can see the potential for hung calls if asterisk never get the BYE from a UA in the event the ATA gets unplugged or somehow loses power. That is the case in every SIP network, Asterisk is not unique in that regard. I would suggest that you could make a modification to chan_sip so that if the peer goes 'unreachable' (as determined by using qualify=yes) than any existing calls involved with that peer are immediately hung up; that would take care of this problem. ___ --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 ___ --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] Re: Asterisk SIP architecture question
David Thomas wrote: Is the CDR accounting done based on SIP signaling? If a UA is talking (RTP) to a third party PSTN gateway, isn't it at risk if say the UA loses power. How will asterisk know the call has ended if it is not involved in the media path. The idea is this.. I want to use canreinvite =yes to force users to talk end-to-end to preserve bandwidth, but I can see the potential for hung calls if asterisk never get the BYE from a UA in the event the ATA gets unplugged or somehow loses power. That is the case in every SIP network, Asterisk is not unique in that regard. I would suggest that you could make a modification to chan_sip so that if the peer goes 'unreachable' (as determined by using qualify=yes) than any existing calls involved with that peer are immediately hung up; that would take care of this problem. ___ --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
[Asterisk-Users] Re: Asterisk SIP architecture question
Kevin, Is the CDR accounting done based on SIP signaling? If a UA is talking (RTP) to a third party PSTN gateway, isn't it at risk if say the UA loses power. How will asterisk know the call has ended if it is not involved in the media path. The idea is this.. I want to use canreinvite =yes to force users to talk end-to-end to preserve bandwidth, but I can see the potential for hung calls if asterisk never get the BYE from a UA in the event the ATA gets unplugged or somehow loses power. regards David On 11/23/05, Kevin P. Fleming <[EMAIL PROTECTED]> 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 :-( > > 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 > ___ --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
[Asterisk-Users] Re: Asterisk SIP architecture question
Thanks for the information Matt! Does asterisk store any SIP dialog cdr info in mysql like Call-ID & Cseq? With This info I could at least detect runaway calls and fake a BYE to the pstn gateway with an external app. regards, David On 11/23/05, Matt Riddell <[EMAIL PROTECTED]> wrote: > 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 > ___ --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
