You have to understand that this mailing list is not free instant support. Even more, you are using an unsupported Asterisk feature for 1.4. I will check it when I have some spare time to try to reproduce and fix it. If you are too much in a hurry you can always contact me off-list for paid support for this feature.
Moy On Mon, Apr 6, 2009 at 3:15 AM, Jose Arias <cyr2...@gmail.com> wrote: > Hi, > I was asked for the patch and I sent it. Does anybody have any news about > this subject? > I'm willing to try a fix for 1.4 but I'd need any guidelines to do it. > Thanks in advanced > Jose > 2009/4/2 Moises Silva <moises.si...@gmail.com> >> >> Async AGI was never released for Asterisk 1.4.X, so probably the patch >> you used has a bug or something, do you still have the patch around? >> >> Moy >> >> On Thu, Apr 2, 2009 at 5:44 AM, <cyr2...@gmail.com> wrote: >> > Hi Henrik, >> > >> > I would like to do the same thing you are doing here. I want to >> > implement an external queue functionality so I need to stop a play file >> > launched previously with an async agi command on caller's channel, sending >> > the call to agent's extension. >> > >> > I'm redirecting caller's channel with REDIRECT while playing is taking >> > place but I'm always getting a hang up on caller's channel. >> > >> > I'm using: >> > >> > asterisk-1.4.18 >> > asterisk-addons-1.4.7 >> > async agi patch 2007-12-11 10:34:12 (the one back-ported to 1.4) >> > >> > Both caller and agent are using 501 and 500 extensions and the async agi >> > loop is waiting on 800, for example. The caller is dialing 800 where a play >> > file is commanded through and async agi stream file command by the >> > application. >> > >> > The relevant part of extensions.conf follows: >> > >> > exten => _5.,1,Noop(SIP call on 'sip_sercom' a ${EXTEN}); >> > exten => _5.,n,Wait(1); >> > exten => _5.,n,Dial(SIP/${EXTEN},${TIMEOUTDIAL},Tto); >> > exten => _5.,n,Hangup(); >> > >> > exten => _8.,1,Noop(every thing starting 8 ${EXTEN}); >> > exten => _8.,n,AGI(agi:async); >> > exten => _8.,n,Hangup(); >> > >> > And the redirect command the application is sending to is: >> > >> > Action: Redirect >> > Channel: SIP/501-081f0730 >> > Exten: 500 >> > Context: sip_sercom >> > Priority: 1 >> > >> > Therefore, Henrik, could you show me your related dial plan and the >> > redirect command you are sending? I wasn't able to see what I'm getting >> > wrong. >> > >> > thanks in advanced >> > Jose M Arias >> > >> > -- >> > This message was sent on behalf of cyr2...@gmail.com at >> > openSubscriber.com >> > >> > http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/10933120.html >> > >> > _______________________________________________ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> >> >> >> -- >> "I do not agree with what you have to say, but I’ll defend to the >> death your right to say it." Voltaire > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- "I do not agree with what you have to say, but I’ll defend to the death your right to say it." Voltaire _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users