Aubrey Wells
2007-Aug-29 17:25 UTC
[asterisk-users] Queue Agents on Remote Asterisk server?
Hi, I have a main Asterisk server, and a server at a branch location connected via a IAX2 trunk. I want to have a queue at the main location that has people from both locations as members. I got this working, but the trouble comes when the round-robin logic selects a member at the branch office to call. If that user is unavailable, their voicemail answers the call, and the main server detects this as an answered call and assumes the agent answered. This is obviously not what I want, as I would like for the call to roll to one of the other agents. Has anyone come across this before? Solutions? Thanks! ------------------ Aubrey Wells -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070829/93d9da26/attachment.htm
James FitzGibbon
2007-Aug-29 17:50 UTC
[asterisk-users] Queue Agents on Remote Asterisk server?
On 8/29/07, Aubrey Wells <aubrey at sheltonjohns.com> wrote:> I have a main Asterisk server, and a server at a branch location connected > via a IAX2 trunk. I want to have a queue at the main location that has > people from both locations as members. I got this working, but the trouble > comes when the round-robin logic selects a member at the branch office to > call. If that user is unavailable, their voicemail answers the call, and the > main server detects this as an answered call and assumes the agent answered. > This is obviously not what I want, as I would like for the call to roll to > one of the other agents. Has anyone come across this before? Solutions? >Don't contact the remote agents using a context that includes a call to VoiceMail(). Contact a remote context that dials the agent using Dial() with the appropriate timeout and hangs up if the agent is unavailable. Then app_queue () will do the right thing. -- j. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070829/d9c6a0ff/attachment.htm
Steve Totaro
2007-Aug-29 17:52 UTC
[asterisk-users] Queue Agents on Remote Asterisk server?
Aubrey Wells wrote:> Hi, > I have a main Asterisk server, and a server at a branch location > connected via a IAX2 trunk. I want to have a queue at the main > location that has people from both locations as members. I got this > working, but the trouble comes when the round-robin logic selects a > member at the branch office to call. If that user is unavailable, > their voicemail answers the call, and the main server detects this as > an answered call and assumes the agent answered. This is obviously not > what I want, as I would like for the call to roll to one of the other > agents. Has anyone come across this before? Solutions? > > Thanks! > * > * > *------------------* > *Aubrey Wells* > >Why do your agents have voicemail? Thanks, Steve
How about sending a SipHeader to the second box and then on the second box look for the header. If the header does not exist then ring the extension normally. If the header is there than send back congestion (basically have a gotoif before it hits the Exten => Foo,1,Voicemail) ----- Original Message ----- From: Aubrey Wells To: asterisk-users at lists.digium.com Sent: Wednesday, August 29, 2007 8:25 PM Subject: [asterisk-users] Queue Agents on Remote Asterisk server? Hi, I have a main Asterisk server, and a server at a branch location connected via a IAX2 trunk. I want to have a queue at the main location that has people from both locations as members. I got this working, but the trouble comes when the round-robin logic selects a member at the branch office to call. If that user is unavailable, their voicemail answers the call, and the main server detects this as an answered call and assumes the agent answered. This is obviously not what I want, as I would like for the call to roll to one of the other agents. Has anyone come across this before? Solutions? Thanks! ------------------ Aubrey Wells ------------------------------------------------------------------------------ _______________________________________________ --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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070831/a99864bc/attachment.htm