Re: [asterisk-users] Asterisk Queue Problem - Automatic Call Distribution
If i call the extension number directing to the queue, i would get the following warning: WARNING[31250]: app_queue.c:3605 queue_exec: Unable to join queue queue1 If i replace the MGCP queue member with a SIP, it is working fine too. i tried using the local channel as adviced with the following configuration: queues.conf strategy=roundrobin member=Local/[EMAIL PROTECTED] extension.conf exten=601,1,Queue(queue1) [test] exten=200,1,Dial(MGCP/[EMAIL PROTECTED]) I would still get the same warning and the call doesn't seems to direct to the test context. Thanks for the reply. Regards, Chong On 5/17/07, Lenz [EMAIL PROTECTED] wrote: Hi Chong, I have no experience with MGCP, but do you see anything in the Asterisk CLI or the full log while the terminal is supposedly being called by the ACD? Another thing you could do is to create a Local/[EMAIL PROTECTED] channel, whee as the queue memebre you have Local/[EMAIL PROTECTED] and that does the actual calling to the MGCP terminal - you should see something in the CLI at this point. Hope this helps, l. On Wed, 16 May 2007 13:45:19 +0200, TienSen Chong [EMAIL PROTECTED] wrote: Hi all, I am seeing a strange problem with Asterisk queue. I am not sure if it's my configuration which is wrong or there's something with Asterisk. I am using Asterisk 1.4.2 and i have a queue with one MGCP member. When i tried to call the extension number directing to the queue, the MGCP phone is not ringing. However, it is fine to call the MGCP phone directly. The strange thing is after i've called the MGCP phone directly, calling the extension number directing to the queue works fine. I wonder what could be wrong. Any comment and help is very much appreciated. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Queue MOH
Hi, I have a queue with a few members. When i tried to call the number directing to the queue, one of the member is ringing and the music on hold is being played at the caller. Is there any way if i want the caller to hear dial tone rather than the MOH? Regards, Chong ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue MOH
Thank you very much for sharing this. Chong On 5/17/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TienSen Chong Sent: 17. maí 2007 10:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Queue MOH Is there any way if i want the caller to hear dial tone rather than the MOH? Perhaps you could use something like Queue(yourqueuename|rt|||60); in extensions.conf or extension.ael? The r is defined as ring instead of playing MOH. Baldvin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Queue Problem - Automatic Call Distribution
Hi all, I am seeing a strange problem with Asterisk queue. I am not sure if it's my configuration which is wrong or there's something with Asterisk. I am using Asterisk 1.4.2 and i have a queue with one MGCP member. When i tried to call the extension number directing to the queue, the MGCP phone is not ringing. However, it is fine to call the MGCP phone directly. The strange thing is after i've called the MGCP phone directly, calling the extension number directing to the queue works fine. I wonder what could be wrong. Any comment and help is very much appreciated. The following is the configuration: queues.conf [queue1] strategy=roundrobin member=MGCP/[EMAIL PROTECTED] mgcp.conf [101] host=dynamic context=default canreinvite=no callerid=101101 line=101 extension.conf exten=601,1,Queue(queue1) Regards, Chong ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Conference with G.722
I haven't tried the app_conference yet. I want to know if the conference is consisting of 3 users with G.722, does the app_conference perform transcoding? If it is not, then app_conference will solve the issue of having conference consists of only G.722 user since no transcoding is needed. Is my understanding correct? Regards, chong On 4/26/07, Thomas Kenyon [EMAIL PROTECTED] wrote: TienSen Chong wrote: Hi all, I am having problem with conference call (meetme feature) using G.722 phone. G.722 phone to phone is working fine. I suspect this is due to the fact that Asterisk 1.4 only support G.722 passthrough. This will be the case, Meetme transcodes the audio (to slin iirc), where it mixes it. Any ideas how this problem can be fixed. Have you tried using app_conference? To be honest, I don't know how you would be able to have more than 2 people in a call without some transcoding going on. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Conference with G.722
Hi all, I am having problem with conference call (meetme feature) using G.722 phone. G.722 phone to phone is working fine. I suspect this is due to the fact that Asterisk 1.4 only support G.722 passthrough. Any ideas how this problem can be fixed. Thanks. Regards, Chong ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
