Re: [asterisk-users] Call Queue advise
yes but I believe that least recent would ring one agent at a time? If my understanding is incorrect please correct it. We are wanting to keep with multiple phones ring to ensure coverage. From what I've seen, I don't think this is possible. But maybe ask in the #asterisk channel on Freenode IRC. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Queue advise
yes but I believe that least recent would ring one agent at a time? If my understanding is incorrect please correct it. We are wanting to keep with multiple phones ring to ensure coverage. From what I've seen, I don't think this is possible. But maybe ask in the #asterisk channel on Freenode IRC. Could ring the least-recent a couple times to provide workload balance, then ring everyone to make sure someone answers. --Don -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with upgrading - RBS T1
I have a system with two Sangoma A104D cards running Asterisk 1.8.11-cert10, Dahdi 2.5.0.1, LibPRI 1.4.12, and Wanpipe 3.5.23. The PRI spans are configured with esf,b8zs. Everything has been working great, which is why I haven't updated it further. You might try an older Dahdi version just to see. Although this might be tricky depending on the OS version. Ryan On Mon, Dec 9, 2013 at 5:56 PM, Jeff LaCoursiere j...@jeff.net wrote: Upgrading an ancient customer installation... was running 1.4.23.1 (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running fine for 5+ years. Customer getting anxious about hardware failure, so we built a new box and installed 1.8.24.0, Dahdi 2.7.0.1, and a new Sangoma A104D. The single active span is an RBS T1 B8ZS/ESF/EM Wink. I tried to move one span over one night which was working fine on the old box. Once plugged in there were no alarms, Sangoma wanpipemon utility showed connected. I tried calling in on a DID number, and in the 'full' log, with debug and verbose set to 100: [Dec 5 00:51:37] VERBOSE[5283] sig_analog.c: -- Starting simple switch on 'DAHDI/9-1' [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=1.17E+04, Et=1.45E+06, s/n= 0.01 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=2.76E+03, Et=1.10E+06, s/n= 0.00 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=2.06E+04, Et=1.39E+06, s/n= 0.01 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=4.68E+03, Et=1.40E+06, s/n= 0.00 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00, s/n= -nan [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00, s/n= -nan [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00, s/n= -nan [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00, s/n= -nan [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00, s/n= -nan [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=1.30E+10, Et=2.11E+12, s/n= 0.01 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: analog_exception 9 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: Exception on 15, channel 9 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: __analog_handle_event 9 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: Got event UNKNOWN/OTHER(131127) on channel 9 (index 0) [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: DTMF Down '7' [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: Begin DTMF digit: 0x37 '7' on DAHDI/9-1 [Dec 5 00:51:38] DEBUG[5283] chan_dahdi.c: Begin DTMF digit: 0x37 '7' on DAHDI/9-1 [Dec 5 00:51:38] DTMF[5283] channel.c: DTMF begin '7' received on DAHDI/9-1 [Dec 5 00:51:38] DTMF[5283] channel.c: DTMF begin ignored '7' on DAHDI/9-1 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=2.02E+10, Et=4.01E+12, s/n= 0.01 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=1.88E+10, Et=3.89E+12, s/n= 0.00 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=4.78E+10, Et=1.17E+12, s/n= 0.04 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=5.10E+03, Et=6.26E+06, s/n= 0.00 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: analog_exception 9 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: Exception on 15, channel 9 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: __analog_handle_event 9 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: Got event UNKNOWN/OTHER(262199) on channel 9 (index 0) [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: Detected digit '7' [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: End DTMF digit: 0x37 '7' on DAHDI/9-1 [Dec 5 00:51:38] DEBUG[5283] chan_dahdi.c: End DTMF digit: 0x37 '7' on DAHDI/9-1 [Dec 5 00:51:38] DTMF[5283] channel.c: DTMF end '7' received on DAHDI/9-1, duration 0 ms [Dec 5 00:51:38] DTMF[5283] channel.c: DTMF end accepted without begin '7' on DAHDI/9-1 [Dec 5 00:51:38] DTMF[5283] channel.c: DTMF end passthrough '7' on DAHDI/9-1 [Dec 5 00:51:38] DEBUG[5283] chan_dahdi.c: Enabled echo cancellation on channel 9 [Dec 5 00:51:38] VERBOSE[5283] sig_analog.c: -- Unknown extension '7' in context 'from-pstn' requested ... At this point I hear 'invalid extension' and get hung up on, but if you grep out all the DTMF events from this call, you get: root@astsouth:/var/log/asterisk# grep 'DTMF end' /tmp/foo | grep received [Dec 5 00:51:38] DTMF[5283] channel.c: DTMF end '7' received on DAHDI/9-1, duration 0 ms [Dec 5 00:51:41] DTMF[5283] channel.c: DTMF end '1' received on DAHDI/9-1, duration 80 ms [Dec 5 00:51:41] DTMF[5283] channel.c: DTMF end '5' received on DAHDI/9-1, duration 80 ms [Dec 5 00:51:41] DTMF[5283] channel.c: DTMF end '7' received on DAHDI/9-1, duration 80 ms [Dec 5 00:51:41] DTMF[5283] channel.c: DTMF end '6' received on DAHDI/9-1, duration 80 ms [Dec 5 00:51:41] DTMF[5283] channel.c: DTMF end '0' received on DAHDI/9-1, duration 80 ms [Dec 5 00:51:41] DTMF[5283] channel.c: DTMF end '0' received on DAHDI/9-1, duration 80 ms And '715-7600' is the
Re: [asterisk-users] what is the possible cause of maximum pbx stack exceeded
I'm not a developer, but from comments in the code, it looks like that warning is generated when Asterisk dialplan processing exceeds a certain depth of includes. Seeing as it is possibly a dialplan related issue, and FreePBX is writing your dialplan, you may have the best odds of getting a relevant answer by asking on the FreePBX forums (and giving them access to a copy of your logs to examine) That's all I got! :) On Wed, Dec 4, 2013 at 3:27 AM, cov...@ccs.covici.com wrote: Hi. I am using asterisk 11 svn r401076M and I am getting this warning at times. I can't find much doing a google search, so anyone with any ideas? I have looked at the logs, but can find no particular pattern to indicate where this is happening and the system appears to be otherwise working, but I am still wondering if something is wrong. I am also using freepbx in case there are known issues there -- because some of these occur during their dialout trunk code. Any suggestions would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the possible cause of maximum pbx stack exceeded
OK, thanks. Rusty Newton rnew...@digium.com wrote: I'm not a developer, but from comments in the code, it looks like that warning is generated when Asterisk dialplan processing exceeds a certain depth of includes. Seeing as it is possibly a dialplan related issue, and FreePBX is writing your dialplan, you may have the best odds of getting a relevant answer by asking on the FreePBX forums (and giving them access to a copy of your logs to examine) That's all I got! :) On Wed, Dec 4, 2013 at 3:27 AM, cov...@ccs.covici.com wrote: Hi. I am using asterisk 11 svn r401076M and I am getting this warning at times. I can't find much doing a google search, so anyone with any ideas? I have looked at the logs, but can find no particular pattern to indicate where this is happening and the system appears to be otherwise working, but I am still wondering if something is wrong. I am also using freepbx in case there are known issues there -- because some of these occur during their dialout trunk code. Any suggestions would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Windows
On Wed, Dec 04, 2013 at 02:12:41PM -0500, Ruddy Gbaguidi wrote: I never tought this is become a Linux vs Windows fight. We have been using asterisk on linux from a long time now and happy with it. But some of our customers who has windows in their environment want to use our call center software we developed on top of asterisk. So, the question was : Did anybody ever tried to isolate the asterisk SIP server/module and make it run under Windows ? Since, asterisk 12 is using pjsip (which is cross platform already), I tought it may be possible and wanted advices. I would love that every single customer switch to Linux and Ubuntu tomorrow morning but at the moment, that's not the case. There was an old half-working port of Asterisk to Cygwin which does run on Windows. It has not worked since at least 1.6.0 . Feel free to try to fix it. I suspect it won't be easy. Patches would be welcomed, I guess (look at what odd fixes that were accepted to make Asterisk build and work on OS/X). And for others: the name is [MS-]Windows. Not 'wind-blows or whatever name you find for it. Please respect this list. If you don't have anything useful to add to the thread, please refrain from replying. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi show channels no such command
On Thu, Dec 05, 2013 at 07:19:56PM -0800, Joseph Towery wrote: Here are the commands asteriskpbx@pbxmain:/etc/dahdi$ dahdi_cfg -vvv DAHDI Tools Version - 2.7.0.2 DAHDI Version: 2.7.0.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 08) 8 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 Setting echocan for channel 5 to mg2 Setting echocan for channel 6 to mg2 Setting echocan for channel 7 to mg2 Setting echocan for channel 8 to mg2 asteriskpbx@pbxmain:/etc/dahdi$ pbxmain*CLI module show like chan_dahdi.so Module Description Use Count chan_dahdi.so DAHDI Telephony Driver w/PRI 0 1 modules loaded What's the output of lsdahdi? Note channels there that are open (by Asterisk). Try running in the asterisk CLI: module unload chan_dahdi.so module load chan_dahdi.so The 'load' command will probably fail and give you the real error. Though you should probably already have it in the logs. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with upgrading - RBS T1
Looks like DTMF tone generation is too short. That first digit '7' only lasted 0ms. The rest of the digit tone durations look about right (80ms). Does this happen on every dial sequence, or just the first? On Tue, Dec 10, 2013 at 8:35 AM, Ryan Wagoner rswago...@gmail.com wrote: I have a system with two Sangoma A104D cards running Asterisk 1.8.11-cert10, Dahdi 2.5.0.1, LibPRI 1.4.12, and Wanpipe 3.5.23. The PRI spans are configured with esf,b8zs. Everything has been working great, which is why I haven't updated it further. You might try an older Dahdi version just to see. Although this might be tricky depending on the OS version. Ryan On Mon, Dec 9, 2013 at 5:56 PM, Jeff LaCoursiere j...@jeff.net wrote: Upgrading an ancient customer installation... was running 1.4.23.1 (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running fine for 5+ years. Customer getting anxious about hardware failure, so we built a new box and installed 1.8.24.0, Dahdi 2.7.0.1, and a new Sangoma A104D. The single active span is an RBS T1 B8ZS/ESF/EM Wink. I tried to move one span over one night which was working fine on the old box. Once plugged in there were no alarms, Sangoma wanpipemon utility showed connected. I tried calling in on a DID number, and in the 'full' log, with debug and verbose set to 100: [Dec 5 00:51:37] VERBOSE[5283] sig_analog.c: -- Starting simple switch on 'DAHDI/9-1' [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=1.17E+04, Et=1.45E+06, s/n= 0.01 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=2.76E+03, Et=1.10E+06, s/n= 0.00 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=2.06E+04, Et=1.39E+06, s/n= 0.01 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=4.68E+03, Et=1.40E+06, s/n= 0.00 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00, s/n= -nan [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00, s/n= -nan [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00, s/n= -nan [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00, s/n= -nan [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00, s/n= -nan [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=1.30E+10, Et=2.11E+12, s/n= 0.01 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: analog_exception 9 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: Exception on 15, channel 9 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: __analog_handle_event 9 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: Got event UNKNOWN/OTHER(131127) on channel 9 (index 0) [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: DTMF Down '7' [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: Begin DTMF digit: 0x37 '7' on DAHDI/9-1 [Dec 5 00:51:38] DEBUG[5283] chan_dahdi.c: Begin DTMF digit: 0x37 '7' on DAHDI/9-1 [Dec 5 00:51:38] DTMF[5283] channel.c: DTMF begin '7' received on DAHDI/9-1 [Dec 5 00:51:38] DTMF[5283] channel.c: DTMF begin ignored '7' on DAHDI/9-1 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=2.02E+10, Et=4.01E+12, s/n= 0.01 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=1.88E+10, Et=3.89E+12, s/n= 0.00 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=4.78E+10, Et=1.17E+12, s/n= 0.04 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=5.10E+03, Et=6.26E+06, s/n= 0.00 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: analog_exception 9 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: Exception on 15, channel 9 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: __analog_handle_event 9 [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: Got event UNKNOWN/OTHER(262199) on channel 9 (index 0) [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: Detected digit '7' [Dec 5 00:51:38] DEBUG[5283] sig_analog.c: End DTMF digit: 0x37 '7' on DAHDI/9-1 [Dec 5 00:51:38] DEBUG[5283] chan_dahdi.c: End DTMF digit: 0x37 '7' on DAHDI/9-1 [Dec 5 00:51:38] DTMF[5283] channel.c: DTMF end '7' received on DAHDI/9-1, duration 0 ms [Dec 5 00:51:38] DTMF[5283] channel.c: DTMF end accepted without begin '7' on DAHDI/9-1 [Dec 5 00:51:38] DTMF[5283] channel.c: DTMF end passthrough '7' on DAHDI/9-1 [Dec 5 00:51:38] DEBUG[5283] chan_dahdi.c: Enabled echo cancellation on channel 9 [Dec 5 00:51:38] VERBOSE[5283] sig_analog.c: -- Unknown extension '7' in context 'from-pstn' requested ... At this point I hear 'invalid extension' and get hung up on, but if you grep out all the DTMF events from this call, you get: root@astsouth:/var/log/asterisk# grep 'DTMF end' /tmp/foo | grep received [Dec 5 00:51:38] DTMF[5283] channel.c: DTMF end '7' received on DAHDI/9-1, duration 0 ms [Dec 5 00:51:41] DTMF[5283] channel.c: DTMF end '1' received on DAHDI/9-1, duration 80 ms [Dec 5 00:51:41] DTMF[5283] channel.c: DTMF end '5' received on DAHDI/9-1, duration 80 ms [Dec 5 00:51:41] DTMF[5283] channel.c: DTMF end '7' received on DAHDI/9-1, duration 80 ms [Dec 5
Re: [asterisk-users] Asterisk on Windows
On Tue, 10 Dec 2013 23:02:45 +0200 Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Wed, Dec 04, 2013 at 02:12:41PM -0500, Ruddy Gbaguidi wrote: I never tought this is become a Linux vs Windows fight. We have been using asterisk on linux from a long time now and happy with it. But some of our customers who has windows in their environment want to use our call center software we developed on top of asterisk. So, the question was : Did anybody ever tried to isolate the asterisk SIP server/module and make it run under Windows ? Since, asterisk 12 is using pjsip (which is cross platform already), I tought it may be possible and wanted advices. I would love that every single customer switch to Linux and Ubuntu tomorrow morning but at the moment, that's not the case. There was an old half-working port of Asterisk to Cygwin which does run on Windows. It has not worked since at least 1.6.0 . That's just a unix-like interface which won't address the issues the OP has/had with running/configuring asterisk. IMHO it would probably be even more challenging. And IIRC the OP was looking for a non emulated solution anyway. Feel free to try to fix it. I suspect it won't be easy. Patches would be welcomed, I guess (look at what odd fixes that were accepted to make Asterisk build and work on OS/X). That advice was already given by multiple posters. OS X is unix-like as well so I fail to see what help that could be in an endeavour to port asterisk. And for others: the name is [MS-]Windows. Not 'wind-blows or whatever name you find for it. Please respect this list. If you don't have anything useful to add to the thread, please refrain from replying. I have to agree with the name calling part but the OP did imply that Windows was superior and that a Windows port would be profitable. You can't really expect to get away with that on a list devoted to an open source application without making a complete fool out of yourself. If it was a post regarding one of the many proprietary closed source applications/games without a native port to Linux/BSD/OS X then it would be a valid complaint. Having access to the source as well as liberal licensing terms which allow porting isn't a valid complaint and never will be. B -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Queue advise
On 13-12-09 06:47 PM, Bryan Anderson wrote: I have a call queue that rings about 15 users and they are wanting to set it up so that the last person to answer a call doesn't ring on the next incoming call. What would be the best way to handle this? I have been looking at the strategies and none of those seem to be right for this. My current thoughts are probably a macro that places a penalty on the user tell the next call is answered. Any advice for this would be greatly appreciated. You have agents that log into a queue that don't want to get calls? Is that what you are saying? Options 1 - log the agent out, they don't get the next call. Option 2 - Set up weights for your agents, as answer a new call, increment then up so they don't get the next. Either way, I see issues with the setup. Best ways is to rethink your queue strategy and stop using ring all. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue with linear strategy does not work
I have a queue with linear strategy. When I add dynamic members it does NOT ring the members in the order they are added. I use the command AddQueueMember to add members but it seems to be random how it rings the members. Hope somebody can help. This is the description of linear strategy: *linear: Rings interfaces in the order they are listed in the configuration file. Dynamic members will be rung in the order in which they were added. * Regards Thorben Jensen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
