asterisk@frameweb.it
2005-Sep-23 08:15 UTC
[Asterisk-Users] Problems with queue and remote agents
I all. I have configured a pair of * servers, sip connected each other Mi problem is the following If on the first * i configure a queue containing phone number of the second * (i.e with a round robin strategy) I have non problem as far as all phones are online. If one of the remote phone number is unavailable, when the round-robin strategy touch that phone the call is answered by the voicemail (the extension is onthephone or is unavailable....) I think that the problem could be the first * pass the call to the second, and has no way to decide if the remote extension is available or not Could be an improvement to iax interconnect the two asterisk ? Or is there any othe solution ? I already removed static agent from the queue, but the problem is the same if one remote extensions is loggd in but is busy Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it
Use agent callback login so the voicemail answering does not result in the call going to vmail, agentcallback login, along with ackcall=yes in agents.conf requires the # key to take the call from the queue.> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of asterisk@frameweb.it > Sent: Friday, September 23, 2005 9:16 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Problems with queue and remote agents > > I all. > I have configured a pair of * servers, sip connected each other > > Mi problem is the following > > If on the first * i configure a queue containing phone number of the > second > * (i.e with a round robin strategy) > I have non problem as far as all phones are online. > > If one of the remote phone number is unavailable, when the round-robin > strategy touch that phone the call is answered > by the voicemail (the extension is onthephone or is unavailable....) > > I think that the problem could be the first * pass the call to thesecond,> and has no way to decide > if the remote extension is available or not > > Could be an improvement to iax interconnect the two asterisk ? > > Or is there any othe solution ? > > I already removed static agent from the queue, but the problem is thesame> if one remote extensions is loggd in but is busy > > Andrea > > Chi ricevesse questa mail per errore e' gentilmente pregato di > cancellarla. > > Visitate il sito http://www.frameweb.it > > _______________________________________________ > --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
You should either use Agents (standard or callback) or disable voicemail on the second server, with a straight dial instead of the dial+voicemail macro you'll likely be using. bye l. In data Fri, 23 Sep 2005 17:15:38 +0200, <asterisk@frameweb.it> ha scritto:> I all. > I have configured a pair of * servers, sip connected each other > > Mi problem is the following > > If on the first * i configure a queue containing phone number of the > second > * (i.e with a round robin strategy) > I have non problem as far as all phones are online. > > If one of the remote phone number is unavailable, when the round-robin > strategy touch that phone the call is answered > by the voicemail (the extension is onthephone or is unavailable....) > > I think that the problem could be the first * pass the call to the > second, > and has no way to decide > if the remote extension is available or not > > Could be an improvement to iax interconnect the two asterisk ? > > Or is there any othe solution ? > > I already removed static agent from the queue, but the problem is the > same > if one remote extensions is loggd in but is busy > > Andrea > > Chi ricevesse questa mail per errore e' gentilmente pregato di > cancellarla. > > Visitate il sito http://www.frameweb.it > > _______________________________________________ > --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-- Assum est, versa et manduca.