Re: [asterisk-users] Asterisk Queues problem- URGENT
Hi all, Okay I have solved the problem. Actually the asterisk detected 24 Port FXO and numbered its ports. Since it has previously detcted ports 1-4 for FXS and ports 58 for FXO for my initial 8-port card. When I installed 24 port second card it numbered the new fxo ports from 9-32. uptill now fair enough. Now problem was when I physically inserted lines in to the Patch Panel 24 port of the new card I inserted the lines from port 1 - 10 (right now only ten lines added). The problem was solved by reinserting the lines in the patch panel from 9-18 since the ports from 1-8 are already detected by asterisk for the previous card. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Syed Nasruddin Sent: Monday, August 04, 2008 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem- URGENT Hi, Can anyone help me on this. I am really stuck.again defining the problem briefly.: 1. Second New card TDM240P added to machine. 2. Only FXO modules i.e 24 FXO. 3. Asterisk detected all the ports successfully and when I run module reload chan_zap.so it list allthe FXO ports correctly. 4. when I can on any of the newly added lines there is a clear ring on the orginators phone while no activity detetcted by asterisk. It just keep quiet. It looks like call is not being detected by the card to my asterisk. 5. 4 port FXO card which was previously installed is functioning properly only this new added card is causing problem. 6. I have 12 new lines and only one of the lines is generating below mentioned logs in asterisk: == Starting post polarity CID detection on channel 18 -- Starting simple switch on 'Zap/18-1' [Aug 4 11:09:29] NOTICE[12255]: chan_zap.c:6169 ss_thread: Got event 17 (Polarity Reversal)... [Aug 4 11:09:31] WARNING[12255]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/18-1' == Starting post polarity CID detection on channel 17 -- Starting simple switch on 'Zap/17-1' [Aug 4 11:09:35] ERROR[12256]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-1) [Aug 4 11:09:35] WARNING[12256]: chan_zap.c:6202 ss_thread: CallerID feed failed: Success [Aug 4 11:09:35] WARNING[12256]: chan_zap.c:6215 ss_thread: CallerID returned with error on channel 'Zap/17-1' [Aug 4 11:09:37] WARNING[12256]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/17-1' Can anyone decipher this code??? What is happening?? Please give me some cluess to work on. In my Zapata.conf I have following two lines related to above logs: Cidsignalling= v23 Cidstart = polarity Please help./ Syed nasr (MONDAY 04/08/2008) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Syed Nasruddin Sent: Friday, August 01, 2008 8:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem Thanks, Yes that was the problem I have added joinempty=yes. It is now working,. Right now another critical problem has come up which I have mentioned in my previous email. I am copying the problem here again: was initially running only with one TDM800P card having 4FXO and 4 FXS port then I later added another 24 port FXO card. So now in total I have now 32 FXO ports for in coming calls. Card was successfully integerated and all the ports were detected by asterisk. Just few minutes back the POT lines were also ready and now I am getting additional errors which I am pasting here. starting simple switch on 'Zap/17-1'[Aug 1 19:00:26] ERROR[3416]: callerid.c:564 callerid_feed: fsk_s erie made mylen 0 (-1)[Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6202 ss_thread: Caller ID feed failed: Success [Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6215 ss_thread: Caller ID returned with error on channel 'Zap/17-1' [Aug 1 19:00:28] WARNING[3416]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch-- Hungup 'Zap/17-1' -- Saved useragent X-Lite release 1002tx stamp 29712 for pee r 1001[Aug 1 19:18:29] NOTICE[3162]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 17 == Starting post polarity CID detection on channel 17-- Starting simple switch on 'Zap/17-1' [Aug 1 19:18:29] NOTICE[3582]: chan_zap.c:6169 ss_thread: Got event 4 (Alarm)... [Aug 1 19:18:31] WARNING[3582]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch Hungup 'Zap/17-1' Please help on this urgent. I cant upgrade right now since I am not confident abt upgrade procedure and any other problems occuring after that. This is my only production machine. thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson Sent: Friday, August 01, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re
Re: [asterisk-users] Asterisk Queues problem- URGENT
On Mon, Aug 04, 2008 at 11:34:06AM +0500, Syed Nasruddin wrote: Hi, Can anyone help me on this. I am really stuck.again defining the problem briefly.: 1. Second New card TDM240P added to machine. 2. Only FXO modules i.e 24 FXO. 3. Asterisk detected all the ports successfully and when I run module reload chan_zap.so it list allthe FXO ports correctly. 4. when I can on any of the newly added lines there is a clear ring on the orginators phone while no activity detetcted by asterisk. It just keep quiet. It looks like call is not being detected by the card to my asterisk. 5. 4 port FXO card which was previously installed is functioning properly only this new added card is causing problem. 6. I have 12 new lines and only one of the lines is generating below mentioned logs in asterisk: == Starting post polarity CID detection on channel 18 -- Starting simple switch on 'Zap/18-1' [Aug 4 11:09:29] NOTICE[12255]: chan_zap.c:6169 ss_thread: Got event 17 (Polarity Reversal)... [Aug 4 11:09:31] WARNING[12255]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/18-1' == Starting post polarity CID detection on channel 17 -- Starting simple switch on 'Zap/17-1' [Aug 4 11:09:35] ERROR[12256]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-1) [Aug 4 11:09:35] WARNING[12256]: chan_zap.c:6202 ss_thread: CallerID feed failed: Success [Aug 4 11:09:35] WARNING[12256]: chan_zap.c:6215 ss_thread: CallerID returned with error on channel 'Zap/17-1' [Aug 4 11:09:37] WARNING[12256]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/17-1' If you connected a phone on the same line, at which point did it ring? Can anyone decipher this code??? The messages here are basically: The line polarity reversed. But it is still on-hook. So it must be the telco signalling me that the caller ID starts now (before the first ring). ... Hey, I don't see any caller ID. ... And there's no ring coming either. I guess no call's going to start now. Something is terribly wrong. Goodbye! What is happening?? Please give me some cluess to work on. In my Zapata.conf I have following two lines related to above logs: Cidsignalling= v23 Cidstart = polarity Does this combination make sense? Do you use it for all the channels? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queues problem- URGENT
Hi, Can anyone help me on this. I am really stuck.again defining the problem briefly.: 1. Second New card TDM240P added to machine. 2. Only FXO modules i.e 24 FXO. 3. Asterisk detected all the ports successfully and when I run module reload chan_zap.so it list allthe FXO ports correctly. 4. when I can on any of the newly added lines there is a clear ring on the orginators phone while no activity detetcted by asterisk. It just keep quiet. It looks like call is not being detected by the card to my asterisk. 5. 4 port FXO card which was previously installed is functioning properly only this new added card is causing problem. 6. I have 12 new lines and only one of the lines is generating below mentioned logs in asterisk: == Starting post polarity CID detection on channel 18 -- Starting simple switch on 'Zap/18-1' [Aug 4 11:09:29] NOTICE[12255]: chan_zap.c:6169 ss_thread: Got event 17 (Polarity Reversal)... [Aug 4 11:09:31] WARNING[12255]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/18-1' == Starting post polarity CID detection on channel 17 -- Starting simple switch on 'Zap/17-1' [Aug 4 11:09:35] ERROR[12256]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-1) [Aug 4 11:09:35] WARNING[12256]: chan_zap.c:6202 ss_thread: CallerID feed failed: Success [Aug 4 11:09:35] WARNING[12256]: chan_zap.c:6215 ss_thread: CallerID returned with error on channel 'Zap/17-1' [Aug 4 11:09:37] WARNING[12256]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/17-1' Can anyone decipher this code??? What is happening?? Please give me some cluess to work on. In my Zapata.conf I have following two lines related to above logs: Cidsignalling= v23 Cidstart = polarity Please help./ Syed nasr (MONDAY 04/08/2008) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Syed Nasruddin Sent: Friday, August 01, 2008 8:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem Thanks, Yes that was the problem I have added joinempty=yes. It is now working,. Right now another critical problem has come up which I have mentioned in my previous email. I am copying the problem here again: was initially running only with one TDM800P card having 4FXO and 4 FXS port then I later added another 24 port FXO card. So now in total I have now 32 FXO ports for in coming calls. Card was successfully integerated and all the ports were detected by asterisk. Just few minutes back the POT lines were also ready and now I am getting additional errors which I am pasting here. starting simple switch on 'Zap/17-1'[Aug 1 19:00:26] ERROR[3416]: callerid.c:564 callerid_feed: fsk_s erie made mylen 0 (-1)[Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6202 ss_thread: Caller ID feed failed: Success [Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6215 ss_thread: Caller ID returned with error on channel 'Zap/17-1' [Aug 1 19:00:28] WARNING[3416]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch-- Hungup 'Zap/17-1' -- Saved useragent X-Lite release 1002tx stamp 29712 for pee r 1001[Aug 1 19:18:29] NOTICE[3162]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 17 == Starting post polarity CID detection on channel 17-- Starting simple switch on 'Zap/17-1' [Aug 1 19:18:29] NOTICE[3582]: chan_zap.c:6169 ss_thread: Got event 4 (Alarm)... [Aug 1 19:18:31] WARNING[3582]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch Hungup 'Zap/17-1' Please help on this urgent. I cant upgrade right now since I am not confident abt upgrade procedure and any other problems occuring after that. This is my only production machine. thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson Sent: Friday, August 01, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem Syed Nasruddin wrote: Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: _app_queue.c:3939 queue_exec: unable to join queue myqueue_ In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It's a production machine. Thanks Syed nasr
[asterisk-users] Asterisk Queues problem
Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue myqueue In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It's a production machine. Thanks Syed nasr ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queues problem
Hi, I was initially running only with one TDM800P card having 4FXO and 4 FXS port then I later added another 24 port FXO card. So now in total I have now 32 FXO ports for in coming calls. Card was successfully integerated and all the ports were detected by asterisk. Just few minutes back the POT lines were also ready and now I am getting additional errors which I am pasting here. starting simple switch on 'Zap/17-1'[Aug 1 19:00:26] ERROR[3416]: callerid.c:564 callerid_feed: fsk_s erie made mylen 0 (-1)[Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6202 ss_thread: Caller ID feed failed: Success [Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6215 ss_thread: Caller ID returned with error on channel 'Zap/17-1' [Aug 1 19:00:28] WARNING[3416]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch-- Hungup 'Zap/17-1' -- Saved useragent X-Lite release 1002tx stamp 29712 for pee r 1001[Aug 1 19:18:29] NOTICE[3162]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 17 == Starting post polarity CID detection on channel 17-- Starting simple switch on 'Zap/17-1' [Aug 1 19:18:29] NOTICE[3582]: chan_zap.c:6169 ss_thread: Got eve nt 4 (Alarm)... [Aug 1 19:18:31] WARNING[3582]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch * Hungup 'Zap/17-1' Kindly give me a hint abt what is happening. And also why my agents are not getting in the queues. Thanks for quick reply. Syed nasr From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Syed Nasruddin Sent: Friday, August 01, 2008 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Queues problem Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue myqueue In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It's a production machine. Thanks Syed nasr ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queues problem
Syed Nasruddin wrote: Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: _app_queue.c:3939 queue_exec: unable to join queue “myqueue”_ In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It’s a production machine. Thanks Syed nasr When diagnosing this sort of issue, it is a good idea to check the value of QUEUESTATUS to see why the caller could not enter the queue. The most common reason for a caller to not join the queue is because joinempty=no is set in queues.conf (if you do not have joinempty set at all, then it defaults to no). This setting causes callers attempting to join a queue to not be able to if the queue is empty or if all the queue members are paused or have an invalid device state. Another possibility is that you have a maximum length set on the queue and so no more callers can join because the queue is full. My suggestion is to see what the QUEUESTATUS is. If the status is JOINEMPTY, then you can issue a queue show command on the CLI to see what the current states of your queue members are. It may be as easy to fix as setting joinempty=yes in queues.conf. If the status is something else, though, then a different fix may be in order instead. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queues problem
Syed Nasruddin a écrit : Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: _app_queue.c:3939 queue_exec: unable to join queue “myqueue”_ In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It’s a production machine. Thanks Syed nasr I would recommend upgrading your asterisk to at least 14.20.1 I have had many troubles with queues, SIP and IAX with asterisk 1.4.18 that have been fixed in the following releases ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queues problem
Thanks, Yes that was the problem I have added joinempty=yes. It is now working,. Right now another critical problem has come up which I have mentioned in my previous email. I am copying the problem here again: was initially running only with one TDM800P card having 4FXO and 4 FXS port then I later added another 24 port FXO card. So now in total I have now 32 FXO ports for in coming calls. Card was successfully integerated and all the ports were detected by asterisk. Just few minutes back the POT lines were also ready and now I am getting additional errors which I am pasting here. starting simple switch on 'Zap/17-1'[Aug 1 19:00:26] ERROR[3416]: callerid.c:564 callerid_feed: fsk_s erie made mylen 0 (-1)[Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6202 ss_thread: Caller ID feed failed: Success [Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6215 ss_thread: Caller ID returned with error on channel 'Zap/17-1' [Aug 1 19:00:28] WARNING[3416]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch-- Hungup 'Zap/17-1' -- Saved useragent X-Lite release 1002tx stamp 29712 for pee r 1001[Aug 1 19:18:29] NOTICE[3162]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 17 == Starting post polarity CID detection on channel 17-- Starting simple switch on 'Zap/17-1' [Aug 1 19:18:29] NOTICE[3582]: chan_zap.c:6169 ss_thread: Got event 4 (Alarm)... [Aug 1 19:18:31] WARNING[3582]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch Hungup 'Zap/17-1' Please help on this urgent. I cant upgrade right now since I am not confident abt upgrade procedure and any other problems occuring after that. This is my only production machine. thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson Sent: Friday, August 01, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem Syed Nasruddin wrote: Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: _app_queue.c:3939 queue_exec: unable to join queue myqueue_ In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It's a production machine. Thanks Syed nasr When diagnosing this sort of issue, it is a good idea to check the value of QUEUESTATUS to see why the caller could not enter the queue. The most common reason for a caller to not join the queue is because joinempty=no is set in queues.conf (if you do not have joinempty set at all, then it defaults to no). This setting causes callers attempting to join a queue to not be able to if the queue is empty or if all the queue members are paused or have an invalid device state. Another possibility is that you have a maximum length set on the queue and so no more callers can join because the queue is full. My suggestion is to see what the QUEUESTATUS is. If the status is JOINEMPTY, then you can issue a queue show command on the CLI to see what the current states of your queue members are. It may be as easy to fix as setting joinempty=yes in queues.conf. If the status is something else, though, then a different fix may be in order instead. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Queues Problem
Help! I'm (still) having issues with Asterisk Queues. I want to implement a queue so that callers get the 'all our staff are busy at the moment, your call has been placed in a queue and will be answered by the first available operator. You may press 1 at any time to leave a voicemail' announcement, then they can press 1 and leave a voicemail. Documentation with * 1.2.x (I'm using 1.2.15) and in the O'Reilly Asterisk book says I can add a line context=blah to the queue definition and this becomes the 'escape context' where pressing buttons will take you to whilst in the queue. I've done this, and put the relevant context in extensions.conf and put extension 1 in there - and nothing happens - I call into the queue and press 1 and don't go anywhere. Please help if you know how to solve this issue, I have been working on it for a week and it's becoming quite urgent (not to mention causing me to tear my hair out with frustration...) Regards, John Breen ___ --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 Queues Problem
You need operator=yes as well... John Breen wrote: Help! I'm (still) having issues with Asterisk Queues. I want to implement a queue so that callers get the 'all our staff are busy at the moment, your call has been placed in a queue and will be answered by the first available operator. You may press 1 at any time to leave a voicemail' announcement, then they can press 1 and leave a voicemail. Documentation with * 1.2.x (I'm using 1.2.15) and in the O'Reilly Asterisk book says I can add a line context=blah to the queue definition and this becomes the 'escape context' where pressing buttons will take you to whilst in the queue. I've done this, and put the relevant context in extensions.conf and put extension 1 in there - and nothing happens - I call into the queue and press 1 and don't go anywhere. Please help if you know how to solve this issue, I have been working on it for a week and it's becoming quite urgent (not to mention causing me to tear my hair out with frustration...) Regards, John Breen ___ --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
Re: [asterisk-users] Asterisk Queues Problem
Hi John,
It would be helpful if you had posted your dial plan. But anyway, here is a
sample of our dial plan and how we are handling queues.
[company-inbound-ivr-officehrs]
exten = s,1,Background(company-officehrs-ivr)
exten = s,2,WaitExten
exten = s,3,Hangup
exten = i,1,Playback(pbx-invalid)
exten = t,1,Goto(company-inbound-ivr-officehrs,s,1)
;Press 1 for English
exten = 1,1,Dial(${TO-ABC-ASTERISK}/companyEnglish,,tTo)
exten = 1,2,Hangup
;Press 2 for Mandarin
exten = 2,1,Dial(${TO-ABC-ASTERISK}/companyMandarin,,tTo)
exten = 2,2,Hangup
[company-english-queue] ;companyEnglish
exten = s,1,AGI(call_log.agi,${EXTEN})
exten = s,2,Queue(queue-out-English|thHr|||10)
exten = s,3,Queue(queue-out-English|tThH|||60)
exten = s,4,Voicemail([EMAIL PROTECTED])
exten = s,5,Hangup
exten = h,1,DeadAGI(call_log.agi,${EXTEN})
exten = t,1,Hangup
[company-mandarin-queue] ;companyMandarin
exten = s,1,AGI(call_log.agi,${EXTEN})
exten = s,2,Queue(queue-out-Mandarin|thHr|||10)
exten = s,3,Queue(queue-out-Mandarin|tThH|||60)
exten = s,4,Voicemail([EMAIL PROTECTED])
exten = s,5,Hangup
exten = h,1,DeadAGI(call_log.agi,${EXTEN})
exten = t,1,Hangup
All call passes through an IVR, from the IVR callers can choose language
(English,Mandarin). And then they will be passed to the queues. Take note of
the Queue(queue-out-English|tThH|||60), the option T allows the calling user
to transfer the call when they pressed a single digit, while in the queue.
Hope that helps.
Best Regards,
Joanna Liza Mariazeta
www.mariazeta.com
On 2/16/07, John Breen [EMAIL PROTECTED] wrote:
Help!
I'm (still) having issues with Asterisk Queues.
I want to implement a queue so that callers get the 'all our staff are
busy at the moment, your call has been placed in a queue and will be
answered by the first available operator. You may press 1 at any time
to leave a voicemail' announcement, then they can press 1 and leave a
voicemail.
Documentation with * 1.2.x (I'm using 1.2.15) and in the O'Reilly
Asterisk book says I can add a line context=blah to the queue definition
and this becomes the 'escape context' where pressing buttons will take
you to whilst in the queue.
I've done this, and put the relevant context in extensions.conf and put
extension 1 in there - and nothing happens - I call into the queue and
press 1 and don't go anywhere.
Please help if you know how to solve this issue, I have been working on
it for a week and it's becoming quite urgent (not to mention causing me
to tear my hair out with frustration...)
Regards,
John Breen
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