Re: [asterisk-users] Too many open files
You can also Try: ulimit -c unlimited , then restart asterisk Juan. Linux User #441131 On Thu, Jan 26, 2012 at 5:53 PM, Chad Wallace cwall...@lodgingcompany.comwrote: On Thu, 26 Jan 2012 10:35:14 -0700 Mike Diehl mdi...@diehlnet.com wrote: While trying to track down a T.38 issue, I came across a series of log entries like this: [Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open files [Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create socket What causes it and how do I fix/mitigate it? In the script that runs asterisk, execute this command before running asterisk: ulimit -n 8192 Then restart asterisk. Or, if you have the /etc/default/asterisk file on your system (like I do), uncomment the MAX_FILES line (or increase it), and the default init script should take care of it when you next restart asterisk. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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 -- _ -- 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] ulimit
Hi, I use: ulimit -c unlimited My asterisk box handles about 250 concurrent Channels Regards Juan. Linux User #441131 On Wed, Aug 10, 2011 at 2:23 AM, Pezhman Lali l...@lopl.net wrote: Dear for having an stable system which limit option is good for ulimit comand ? 2-is any option for making asterisk crash-free? Best -- Pezhman Lali -- _ -- 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 -- _ -- 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] A lA orden los libros!!!
http://www.101ftb.com/K00W10P513 Juan. Linux User #441131 -- _ -- 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] how to know status of asterisk from php
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) priline=Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) I think, you should start asterisk before executing asterisk commands regards Juan. Linux User #441131 On Thu, Apr 28, 2011 at 1:19 AM, virendra bhati virbh...@gmail.com wrote: Hi, As per you suggestion I write small php scripts but didn't get result. Below is the php script and output of programs too. *PHP Script:-* ?php $priline = system('/usr/sbin/asterisk -rnx pri show spans',$pri); $asterisk = system(/etc/init.d/asterisk status, $asterisks); $mysql = system(/etc/init.d/mysql status,$mysqls); echo priline=.$priline; echo br; echo pri=.$pri; echo br; echo asterisk=.$asterisk; echo br; echo asterisks=.$asterisks; echo br; echo mysql=.$mysql; echo br; echo mysqls=.$mysqls; echo br; ? *Output:-* Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) priline=Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) pri=1 asterisk= asterisks=127 mysql= mysqls=127 On Wed, Apr 27, 2011 at 8:43 PM, Juan David Diaz juanch...@gmail.comwrote: Hi: http://php.net/manual/en/function.system.php Then, the commands you shoul run: /usr/sbin/asterisk -rnxpri show spans /etc/init.d/asterisk status /etc/init.d/mysql status . . . . and so on!! good luck! Regards. Juan. Linux User #441131 On Wed, Apr 27, 2011 at 6:22 AM, virendra bhati virbh...@gmail.comwrote: Hi How to know status of Asterisk,Mysql. PRI lines and other services from PHP scripts ? Thanks and regards Virendra Bhati +91-9172341457 -- _ -- 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 -- _ -- 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 -- - Thanks and regards Virendra Bhati +91-9172341457 -- _ -- 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 -- _ -- 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] how to know status of asterisk from php
Hi: http://php.net/manual/en/function.system.php Then, the commands you shoul run: /usr/sbin/asterisk -rnxpri show spans /etc/init.d/asterisk status /etc/init.d/mysql status . . . . and so on!! good luck! Regards. Juan. Linux User #441131 On Wed, Apr 27, 2011 at 6:22 AM, virendra bhati virbh...@gmail.com wrote: Hi How to know status of Asterisk,Mysql. PRI lines and other services from PHP scripts ? Thanks and regards Virendra Bhati +91-9172341457 -- _ -- 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 -- _ -- 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 pri card replecement
Only by replacing it.should not be a problem. Juan. Linux User #441131 On Wed, Mar 9, 2011 at 8:13 AM, Satish Patel satish...@hotmail.com wrote: Hey guys, Currently we have non HWEC sangoma pri card but now we are planing to replace card with HWEC support card for echo cancellation. So in this case do I need to re-install everything? Like zaptel, asterisk etc.. Or just replace the card? -- Sent from my iPhone -- _ -- 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 -- _ -- 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] T1 PRI shows yellow/red alarm
Dean, what's your zaptel Zapata config_ regards Juan. Linux User #441131 On Mon, Feb 21, 2011 at 1:44 PM, Dean Hoover kb7...@gmail.com wrote: We are running Asterisk version 1.4.23-1, libpri-1.4.9 and zaptel-1.4.12.1 and two Digium TE220Ps. Debugs are set to 10. We have a T1 PRI connected to the telco. Over the last 4-5 days, we have getting Yellow/Red alarms coming from the T1 PRI. The other two ports in use are connected to internal test switches (Avaya Legend/Avaya Definity), and are not showing any errors. /var/log/asterisk/messages reports: [Feb 21 12:21:56] NOTICE[4795] chan_dahdi.c: PRI got event: Alarm (4) on Primary D-channel of span 2 [Feb 21 12:21:56] DEBUG[4795] chan_dahdi.c: Got event Alarm (4) on D-channel for span 2 /var/log/syslog reports: Feb 21 12:21:56 asterisk kernel: [509981.796536] wct2xxp: Setting yellow alarm on span 2 Feb 21 12:21:56 asterisk kernel: [509981.796562] timing source auto card 0! Feb 21 12:21:56 asterisk kernel: [509981.813535] timing source auto card 0! Feb 21 12:22:01 asterisk kernel: [509986.813869] wct2xxp: Clearing yellow alarm on span 2 Intensive PRI debugging does not show any errors prior to the alarm. The other part to this is for a while it was pretty intermittent. One day we would get it 2 times, another 8-12 times. Today, however, it seems to be happening around every 11-13 minutes. Before this started, there were no errors for the 6 days prior. The first response from the telco 4 days ago said that it was an issue on their T3, then came back saying we were sending something to reset the circuit, but I interpret PRI got event as meaning we received something from them. They put a COM tracer in our building, on that circuit, since Friday afternoon. They took it with them to examine the results this morning, and are supposed to call me when they know something. While they are doing that, I want to make sure that I have all the information I need in order to diagnose it. I haven't found a way to trace the actual B8ZS/ESF frames, and was wondering if there was a way for me to log those events. It's not that I don't trust them, but by the same token I haven't changed anything on my end, the other port on the Digium card isn't reporting an issue, and a complete shutdown of the Asterisk server didn't change the results. Any advice would be greatly appreciated. Dean Hoover Milwaukee, Wisconsin -- _ -- 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 -- _ -- 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] T1 PRI shows yellow/red alarm
I don't see any problem.. but, i don't see the 2nd SPAN @ zaptel: *yellow alarm on span 2* regards. Juan. Linux User #441131 On Mon, Feb 21, 2011 at 2:11 PM, Dean Hoover kb7...@gmail.com wrote: Here you go: /etc/zaptel.conf: loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 #Added 2nd 2xT1 card span=3,0,0,d4,ami em=49-72 span=4,0,0,d4,ami fxoks=73-96 --- /etc/asterisk/zapata.conf: [channels] group=1 context=default signalling=pri_cpe switchtype=qsig channel=1-23 group=2 context=twtelecom-in signalling=pri_cpe switchtype=5ess echocancel=yes channel=25-47 group=3 context=definity-in signalling=em_w channel=49-72 group=10 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=73 group=11 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=74 group=12 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=75 group=13 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=76 group=14 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=77 group=15 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=78 group=16 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=79 group=17 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=80 group=18 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=81 group=19 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=82 group=20 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=83 group=21 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=84 group=22 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=85 group=23 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=86 group=24 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=87 group=25 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=88 group=26 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=89 group=27 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=90 group=28 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=91 group=29 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=92 group=30 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=93 group=31 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=94 group=32 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=95 group=33 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=96 --- On Mon, Feb 21, 2011 at 12:58 PM, Juan David Diaz juanch...@gmail.com wrote: Dean, what's your zaptel Zapata config_ regards Juan. Linux User #441131 On Mon, Feb 21, 2011 at 1:44 PM, Dean Hoover kb7...@gmail.com wrote: We are running Asterisk version 1.4.23-1, libpri-1.4.9 and zaptel-1.4.12.1 and two Digium TE220Ps. Debugs are set to 10. We have a T1 PRI connected to the telco. Over the last 4-5 days, we have getting Yellow/Red alarms coming from the T1 PRI. The other two ports in use are connected to internal test switches (Avaya Legend/Avaya Definity), and are not showing any errors. /var/log/asterisk/messages reports: [Feb 21 12:21:56] NOTICE[4795] chan_dahdi.c: PRI got event: Alarm (4) on Primary D-channel of span 2 [Feb 21 12:21:56] DEBUG[4795] chan_dahdi.c: Got event Alarm (4) on D-channel for span 2 /var/log/syslog reports: Feb 21 12:21:56 asterisk kernel: [509981.796536] wct2xxp: Setting yellow alarm on span 2 Feb 21 12:21:56 asterisk kernel: [509981.796562] timing source auto card 0! Feb 21 12:21:56 asterisk kernel: [509981.813535] timing source auto card 0! Feb 21 12:22:01 asterisk kernel: [509986.813869] wct2xxp: Clearing yellow alarm on span 2 Intensive PRI debugging does not show any errors prior to the alarm. The other part to this is for a while it was pretty intermittent. One day we would get it 2 times, another 8-12 times. Today, however, it seems to be happening around every 11-13 minutes. Before this started, there were no errors for the 6 days prior. The first response from the telco 4 days ago said that it was an issue on their T3, then came back saying we were sending something to reset the circuit, but I interpret PRI got event as meaning we received something from them. They put a COM tracer in our building
Re: [asterisk-users] T1 PRI shows yellow/red alarm
Your message was: Here you go: /etc/zaptel.conf: loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 #Added 2nd 2xT1 card span=3,0,0,d4,ami em=49-72 span=4,0,0,d4,ami fxoks=73-96 - m, I would change the timing sources of the spans: span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs Have you try to plug the PRI into another Span that has been working properly?? Regards. Juan. Linux User #441131 On Mon, Feb 21, 2011 at 2:23 PM, Dean Hoover kb7...@gmail.com wrote: This doesn't represent the 2nd span? span=2,1,0,esf,b8zs bchan=25-47 dchan=48 Dean On Mon, Feb 21, 2011 at 1:18 PM, Juan David Diaz juanch...@gmail.com wrote: I don't see any problem.. but, i don't see the 2nd SPAN @ zaptel: yellow alarm on span 2 regards. Juan. Linux User #441131 On Mon, Feb 21, 2011 at 2:11 PM, Dean Hoover kb7...@gmail.com wrote: Here you go: /etc/zaptel.conf: loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 #Added 2nd 2xT1 card span=3,0,0,d4,ami em=49-72 span=4,0,0,d4,ami fxoks=73-96 --- /etc/asterisk/zapata.conf: [channels] group=1 context=default signalling=pri_cpe switchtype=qsig channel=1-23 group=2 context=twtelecom-in signalling=pri_cpe switchtype=5ess echocancel=yes channel=25-47 group=3 context=definity-in signalling=em_w channel=49-72 group=10 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=73 group=11 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=74 group=12 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=75 group=13 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=76 group=14 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=77 group=15 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=78 group=16 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=79 group=17 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=80 group=18 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=81 group=19 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=82 group=20 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=83 group=21 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=84 group=22 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=85 group=23 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=86 group=24 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=87 group=25 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=88 group=26 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=89 group=27 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=90 group=28 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=91 group=29 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=92 group=30 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=93 group=31 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=94 group=32 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=95 group=33 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=96 --- On Mon, Feb 21, 2011 at 12:58 PM, Juan David Diaz juanch...@gmail.com wrote: Dean, what's your zaptel Zapata config_ regards Juan. Linux User #441131 On Mon, Feb 21, 2011 at 1:44 PM, Dean Hoover kb7...@gmail.com wrote: We are running Asterisk version 1.4.23-1, libpri-1.4.9 and zaptel-1.4.12.1 and two Digium TE220Ps. Debugs are set to 10. We have a T1 PRI connected to the telco. Over the last 4-5 days, we have getting Yellow/Red alarms coming from the T1 PRI. The other two ports in use are connected to internal test switches (Avaya Legend/Avaya Definity), and are not showing any errors. /var/log/asterisk/messages reports: [Feb 21 12:21:56] NOTICE[4795] chan_dahdi.c: PRI got event: Alarm (4) on Primary D-channel of span 2 [Feb 21 12:21:56] DEBUG[4795] chan_dahdi.c: Got
Re: [asterisk-users] T1 PRI shows yellow/red alarm
Ooops, my bad I Did not read the zaptel config file correctly, my apologize. span=1,0,0,esf,b8zs bchan=1-23 dchan=24 *span=2,1,0,esf,b8zs bchan=25-47 dchan=48* Juan. Linux User #441131 On Mon, Feb 21, 2011 at 2:34 PM, Juan David Diaz juanch...@gmail.comwrote: Your message was: Here you go: /etc/zaptel.conf: loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 #Added 2nd 2xT1 card span=3,0,0,d4,ami em=49-72 span=4,0,0,d4,ami fxoks=73-96 - m, I would change the timing sources of the spans: span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs Have you try to plug the PRI into another Span that has been working properly?? Regards. Juan. Linux User #441131 On Mon, Feb 21, 2011 at 2:23 PM, Dean Hoover kb7...@gmail.com wrote: This doesn't represent the 2nd span? span=2,1,0,esf,b8zs bchan=25-47 dchan=48 Dean On Mon, Feb 21, 2011 at 1:18 PM, Juan David Diaz juanch...@gmail.com wrote: I don't see any problem.. but, i don't see the 2nd SPAN @ zaptel: yellow alarm on span 2 regards. Juan. Linux User #441131 On Mon, Feb 21, 2011 at 2:11 PM, Dean Hoover kb7...@gmail.com wrote: Here you go: /etc/zaptel.conf: loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 #Added 2nd 2xT1 card span=3,0,0,d4,ami em=49-72 span=4,0,0,d4,ami fxoks=73-96 --- /etc/asterisk/zapata.conf: [channels] group=1 context=default signalling=pri_cpe switchtype=qsig channel=1-23 group=2 context=twtelecom-in signalling=pri_cpe switchtype=5ess echocancel=yes channel=25-47 group=3 context=definity-in signalling=em_w channel=49-72 group=10 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=73 group=11 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=74 group=12 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=75 group=13 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=76 group=14 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=77 group=15 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=78 group=16 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=79 group=17 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=80 group=18 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=81 group=19 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=82 group=20 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=83 group=21 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=84 group=22 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=85 group=23 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=86 group=24 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=87 group=25 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=88 group=26 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=89 group=27 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=90 group=28 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=91 group=29 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=92 group=30 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=93 group=31 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=94 group=32 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=95 group=33 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=96 --- On Mon, Feb 21, 2011 at 12:58 PM, Juan David Diaz juanch...@gmail.com wrote: Dean, what's your zaptel Zapata config_ regards Juan. Linux User #441131 On Mon, Feb 21, 2011 at 1:44 PM, Dean Hoover kb7...@gmail.com wrote: We are running Asterisk version 1.4.23-1, libpri-1.4.9 and zaptel-1.4.12.1 and two Digium TE220Ps. Debugs are set to 10. We have a T1 PRI connected to the telco. Over the last 4-5 days, we have getting Yellow/Red alarms coming from the T1 PRI. The other two ports in use
Re: [asterisk-users] T1 PRI shows yellow/red alarm
have you check the PRI crossover cable? Juan. Linux User #441131 On Mon, Feb 21, 2011 at 2:35 PM, Juan David Diaz juanch...@gmail.comwrote: Ooops, my bad I Did not read the zaptel config file correctly, my apologize. span=1,0,0,esf,b8zs bchan=1-23 dchan=24 *span=2,1,0,esf,b8zs bchan=25-47 dchan=48* Juan. Linux User #441131 On Mon, Feb 21, 2011 at 2:34 PM, Juan David Diaz juanch...@gmail.comwrote: Your message was: Here you go: /etc/zaptel.conf: loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 #Added 2nd 2xT1 card span=3,0,0,d4,ami em=49-72 span=4,0,0,d4,ami fxoks=73-96 - m, I would change the timing sources of the spans: span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs Have you try to plug the PRI into another Span that has been working properly?? Regards. Juan. Linux User #441131 On Mon, Feb 21, 2011 at 2:23 PM, Dean Hoover kb7...@gmail.com wrote: This doesn't represent the 2nd span? span=2,1,0,esf,b8zs bchan=25-47 dchan=48 Dean On Mon, Feb 21, 2011 at 1:18 PM, Juan David Diaz juanch...@gmail.com wrote: I don't see any problem.. but, i don't see the 2nd SPAN @ zaptel: yellow alarm on span 2 regards. Juan. Linux User #441131 On Mon, Feb 21, 2011 at 2:11 PM, Dean Hoover kb7...@gmail.com wrote: Here you go: /etc/zaptel.conf: loadzone = us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 #Added 2nd 2xT1 card span=3,0,0,d4,ami em=49-72 span=4,0,0,d4,ami fxoks=73-96 --- /etc/asterisk/zapata.conf: [channels] group=1 context=default signalling=pri_cpe switchtype=qsig channel=1-23 group=2 context=twtelecom-in signalling=pri_cpe switchtype=5ess echocancel=yes channel=25-47 group=3 context=definity-in signalling=em_w channel=49-72 group=10 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=73 group=11 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=74 group=12 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=75 group=13 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=76 group=14 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=77 group=15 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=78 group=16 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=79 group=17 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=80 group=18 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=81 group=19 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=82 group=20 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=83 group=21 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=84 group=22 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=85 group=23 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=86 group=24 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=87 group=25 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=88 group=26 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=89 group=27 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=90 group=28 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=91 group=29 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=92 group=30 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=93 group=31 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=94 group=32 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=95 group=33 context=testivr-in signalling=fxo_ks threewaycalling=yes transfer=yes channel=96 --- On Mon, Feb 21, 2011 at 12:58 PM, Juan David Diaz juanch...@gmail.com wrote: Dean, what's your zaptel Zapata config_ regards Juan. Linux User #441131 On Mon, Feb 21, 2011 at 1:44 PM, Dean Hoover kb7...@gmail.com wrote: We are running Asterisk version 1.4.23-1, libpri-1.4.9 and zaptel-1.4.12.1 and two Digium TE220Ps. Debugs are set to 10. We have
Re: [asterisk-users] Setting two E1 cards
system.conf: span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 span=2,2,0,ccs,hdb3,crc4 # termtype: te bchan=32-46,48-62 dchan=47 echocanceller=mg2,32-46,48-62 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 span=3,3,0,ccs,hdb3,crc4 # termtype: te bchan=63-77,79-93 dchan=78 echocanceller=mg2,63-77,79-93 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 span=4,4,0,ccs,hdb3,crc4 # termtype: te bchan=94-108,110-124 dchan=109 echocanceller=mg2,94-108,110-124 # Span 5: TE4/1/1 T4XXP (PCI) Card 1 Span 1 span=5,5,0,ccs,hdb3,crc4 # termtype: te bchan=125-139,141-155 dchan=140 echocanceller=mg2,125-139,141-155 # Span 6: TE4/1/2 T4XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3,crc4 # termtype: te bchan=156-170,172-186 dchan=171 echocanceller=mg2,156-170,172-186 # Span 7: TE4/1/3 T4XXP (PCI) Card 1 Span 3 span=7,7,0,ccs,hdb3,crc4 # termtype: te bchan=187-201,203-217 dchan=202 echocanceller=mg2,187-201,203-217 # Span 8: TE4/1/4 T4XXP (PCI) Card 1 Span 4 span=8,8,0,ccs,hdb3,crc4 # termtype: te bchan=218-232,234-248 dchan=233 echocanceller=mg2,218-232,234-248 # Global data loadzone= us defaultzone = us dahdi_channels (if included at chan_dahdi.conf): Span 1: TE4/0/2 T4XXP (PCI) Card 0 Span 1 group=1 context=pri_aux_port_1 switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31 ; Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 group=2 context=pri_aux_port_2 switchtype = euroisdn signalling = pri_cpe channel = 32-46,48-62 ; Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 group=3 context=pri_aux_port_3 switchtype = euroisdn signalling = pri_cpe channel = 63-77,79-93 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=pri_aux_port_4 switchtype = euroisdn signalling = pri_cpe channel = 94-108,110-124 ; Span 5: TE4/1/1 T4XXP (PCI) Card 1 Span 1 group=5 context=pri_aux_port_5 switchtype = euroisdn signalling = pri_cpe channel = 125-139,141-155 ; Span 6: TE4/1/2 T4XXP (PCI) Card 1 Span 2 group=6 context=pri_aux_port_6 switchtype = euroisdn signalling = pri_cpe channel = 156-170,172-186 ; Span 7: TE4/1/3 T4XXP (PCI) Card 1 Span 3 group=7 context=pri_aux_port_7 switchtype = euroisdn signalling = pri_cpe channel = 187-201,203-217 ; Span 8: TE4/1/4 T4XXP (PCI) Card 1 Span 4 group=8 context=pri_aux_port_8 switchtype = euroisdn signalling = pri_cpe channel = 218-232,234-248 Regards!!! Juan. Linux User #441131 On Thu, Feb 17, 2011 at 11:46 AM, Alejandro Cabrera Obed aco1...@gmail.comwrote: Dear, I always had one E1 card with one span, so I've never had any problem in set it up through /etc/dahdi/sustem.conf and /etc/asterisk/chan_dahdi.conf because I put span=1. But now I have a PBX with two E1 cards with 4 span (8 span in total). How do I have to define both card in system.conf and chan_dahdi.conf, and how do I have to refer each span to the corresponding card ??? Thanks a lot, Alejandro -- _ -- 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 -- _ -- 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] Asterisk Performance
Hi Asterisk Users, I would like to handle about 250 simultaneous (calls agents only) calls with PRI or a SIP trunk with the following configuration Dell R710 Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz Memory 12GB, 1333MHz RAID 1 - 1 Tb X 2 Is that possible?? Kind Regards Juan. Linux User #441131 -- _ -- 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] HA - asterisk service is not starting
Hi Alexis, Thanks for for answer. This is the master node debug, asterisk zaptel starts correctly when I start it manual. The HA config is the same for both nodes. There is an strange event: When I start both nodes, Master node does not start the services, but if I turn off master, Slave works as expected, it checks master is down and turn both services on. If I turn on Master again and turn off Slave, The master takes control and works pretty fine bringing up both services again. This malfunction or configuration mistake, I think, it´s only happening while master is turned on by first time...but I simply don´t know what could it be. Thanks for your help!! Kind Regards. Juan. Linux User #441131 On Wed, Nov 17, 2010 at 3:59 AM, Alexis de BRUYN ale...@de-bruyn.fr wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Juan David, While starting heartbeat service (with *clusterMaster 192.168.1.147 zaptel asterisk* on haresources), zaptel and asterisk does not start as I´m expecting, this is the debug result: ResourceManager[3260]: 2010/11/16_12:27:12 info: Releasing resource group: clustermaster 192.168.1.147 zaptel asterisk ResourceManager[3260]: 2010/11/16_12:27:12 info: Running /etc/init.d/asterisk stop *ResourceManager[3260]: 2010/11/16_12:27:12 debug: Starting /etc/init.d/asterisk stop -- This Should be start, but I don´t know why it´s stop* [...] Is your Asterisk starting while you invoke directly /etc/init.d/asterisk (and hearbeat is off)? Try to figure if this is an Asterisk/Zaptel or HA misconfiguration/issue first. Do you have the same debug result on both nodes? Which logs do you provide here: Master or Slave ? Best regards, - -- Alexis de BRUYN email : ale...@de-bruyn.fr -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.10 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkzjmW4ACgkQPdN4bPTxnXdjFQCfZS9xy2/d3zTwi+OINCGJFFfv vzkAoKqjPTOdW7dcSknwJt6UJVkINoWn =9NCH -END PGP SIGNATURE- -- _ -- 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 -- _ -- 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] HA - asterisk service is not starting
Hi Alexis, Thanks for for answer. This is the master node debug, asterisk zaptel starts correctly when I start it manual. The HA config is the same for both nodes. There is an strange event: When I start both nodes, Master node does not start the services, but if I turn off master, Slave works as expected, it checks master is down and turn both services on. If I turn on Master again and turn off Slave, The master takes control and works pretty fine bringing up both services again. This malfunction or configuration mistake, I think, it´s only happening while master is turned on by first time...but I simply don´t know what could it be. Thanks for your help!! Kind Regards. Juan. Linux User #441131 On Wed, Nov 17, 2010 at 3:59 AM, Alexis de BRUYN ale...@de-bruyn.frwrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Juan David, While starting heartbeat service (with *clusterMaster 192.168.1.147 zaptel asterisk* on haresources), zaptel and asterisk does not start as I´m expecting, this is the debug result: ResourceManager[3260]: 2010/11/16_12:27:12 info: Releasing resource group: clustermaster 192.168.1.147 zaptel asterisk ResourceManager[3260]: 2010/11/16_12:27:12 info: Running /etc/init.d/asterisk stop *ResourceManager[3260]: 2010/11/16_12:27:12 debug: Starting /etc/init.d/asterisk stop -- This Should be start, but I don´t know why it´s stop* [...] Is your Asterisk starting while you invoke directly /etc/init.d/asterisk (and hearbeat is off)? Try to figure if this is an Asterisk/Zaptel or HA misconfiguration/issue first. Do you have the same debug result on both nodes? Which logs do you provide here: Master or Slave ? Best regards, - -- Alexis de BRUYN email : ale...@de-bruyn.fr -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.10 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkzjmW4ACgkQPdN4bPTxnXdjFQCfZS9xy2/d3zTwi+OINCGJFFfv vzkAoKqjPTOdW7dcSknwJt6UJVkINoWn =9NCH -END PGP SIGNATURE- -- _ -- 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 -- _ -- 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] HA - asterisk service is not starting
Hi Asterisk Users, I´m in trouble setting a HA cluster for Asterisk service. While starting heartbeat service (with *clusterMaster 192.168.1.147 zaptel asterisk* on haresources), zaptel and asterisk does not start as I´m expecting, this is the debug result: ResourceManager[3260]: 2010/11/16_12:27:12 info: Releasing resource group: clustermaster 192.168.1.147 zaptel asterisk ResourceManager[3260]: 2010/11/16_12:27:12 info: Running /etc/init.d/asterisk stop *ResourceManager[3260]: 2010/11/16_12:27:12 debug: Starting /etc/init.d/asterisk stop -- This Should be start, but I don´t know why it´s stop* Shutting down asterisk: [ OK ] ResourceManager[3260]: 2010/11/16_12:27:13 debug: /etc/init.d/asterisk stop done. RC=0 ResourceManager[3260]: 2010/11/16_12:27:13 info: Running /etc/init.d/zaptel stop *ResourceManager[3260]: 2010/11/16_12:27:13 debug: Starting /etc/init.d/zaptel stop **-- This Should be start, but I don´t know why it´s stop* Unloading zaptel hardware drivers:. ResourceManager[3260]: 2010/11/16_12:27:13 debug: /etc/init.d/zaptel stop done. RC=0 */etc/ha.d/haresources* * * clusterMaster 192.168.1.147 zaptel asterisk */etc/ha.d/ha.cf* * * debugfile /var/log/ha-debug logfile /var/log/ha-log logfacility local0 keepalive 2 deadtime 32 warntime 16 initdead 64 bcast eth0 udpport 694 auto_failback on node clusterMaster clusterSlave Can you help me, please, to find what am I doing wrong Thanks in advance Kind Regards. Juan. Linux User #441131 -- _ -- 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] Fwd: HA - asterisk service is not starting
Juan. Linux User #441131 -- Forwarded message -- From: Juan David Diaz juanch...@gmail.com Date: Tue, Nov 16, 2010 at 1:38 PM Subject: HA - asterisk service is not starting To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi Asterisk Users, I´m in trouble setting a HA cluster for Asterisk service. While starting heartbeat service (with *clusterMaster 192.168.1.147 zaptel asterisk* on haresources), zaptel and asterisk does not start as I´m expecting, this is the debug result: ResourceManager[3260]: 2010/11/16_12:27:12 info: Releasing resource group: clustermaster 192.168.1.147 zaptel asterisk ResourceManager[3260]: 2010/11/16_12:27:12 info: Running /etc/init.d/asterisk stop *ResourceManager[3260]: 2010/11/16_12:27:12 debug: Starting /etc/init.d/asterisk stop -- This Should be start, but I don´t know why it´s stop* Shutting down asterisk: [ OK ] ResourceManager[3260]: 2010/11/16_12:27:13 debug: /etc/init.d/asterisk stop done. RC=0 ResourceManager[3260]: 2010/11/16_12:27:13 info: Running /etc/init.d/zaptel stop *ResourceManager[3260]: 2010/11/16_12:27:13 debug: Starting /etc/init.d/zaptel stop **-- This Should be start, but I don´t know why it´s stop* Unloading zaptel hardware drivers:. ResourceManager[3260]: 2010/11/16_12:27:13 debug: /etc/init.d/zaptel stop done. RC=0 */etc/ha.d/haresources* * * clusterMaster 192.168.1.147 zaptel asterisk */etc/ha.d/ha.cf* * * debugfile /var/log/ha-debug logfile /var/log/ha-log logfacility local0 keepalive 2 deadtime 32 warntime 16 initdead 64 bcast eth0 udpport 694 auto_failback on node clusterMaster clusterSlave Can you help me, please, to find what am I doing wrong Thanks in advance Kind Regards. Juan. Linux User #441131 -- _ -- 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] Vicibox vs VicidialNow
The only big difference I know, is: VicidialNow - *based on CentOS* - Vicidial 2.0.5.1rc1 ViciBox - *Based on OpenSuse* - Vicidial 2.0.5 The core of the call center for both of them is Vicidial. Regards. 2010/7/25 Alejandro Cabrera Obed aco1...@gmail.com Dear all, I need a call center asterisk's based solution and I see there are two important solution for 120+ agents: VicidialNow and ViciBox Can you tell me the difference between these open source call center solution please ??? Special thanks Alejandro -- _ -- 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 -- Juan. Linux User #441131 -- _ -- 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 + E1 card
Your installation should work, you must configure the card channels and load the card module on your OS. Regards. 2010/6/16 Alejandro Cabrera Obed aco1...@gmail.com Dear all, I have to install an E1 card in my Asterisk 1.4.23 server and here is my short question: Is it necessary to install or update any Asterisk/Zaptel/Any extra module or the default installation is good enough to just plug and run the E1 card Thanks a lot Alejandro -- _ -- 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 -- Juan. Linux User #441131 -- _ -- 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 Query
2010/4/29 garge rama garge.r...@gmail.com Hi, I am new to asterisk and trying to make calls with TDM400P asterisk digium card. I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and libpri-1.4.10.2 packages which are downloaded from asterisk website ( www.asterisk.org) and able to compile successfully. TDM400P Digium card (having only one FXS connected to J4) has installed successfully in PC. I would like to make calls across SIP [x-lite] to analog phone connected to TDM400P Digium card (fxs-j4). For this the following four conf files are modified as shown below. * chan_dahdi.conf* *==* [channels] context=test usecallerid=yes hidecallerid=no immediate=no signaling=fxo_ks echocancel=yes group=1 channel=1 *extensions.conf*** *=* [my-phones] ---*EXTEN does not exists for your sip peer context* exten = 2000,1,Dial(SIP/2000) ; Should look like: *exten = ,1,Asterisk_Application(Action)* ;Dial(Zap/1/${Phone_Number_you want}) [test] exten = ,1,Dial(Zap/1) exten = ,2,HangUp() *sip.conf*** *===* [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend *context=**my-phones * secret=1234 host=dynamic *system.conf* *==* fxoks=1 loadzone = be defaultzone = be With those changes x-lite getting registered with asterisk and analog device/phone is getting ring tone with off-hook and also getting debug prints on cli, but not able to make calls. Test Setup: X-lite [configured as 2000, password… other info] running on asterisk PC à registered with asterisk. Analog phone connected to TDM400P Digium card - FXS-J4 running on same asterisk PC à getting ring tone Test Result: = Tried by calling from x-lite à getting message on CLI “call from ‘2000’ to ‘’ rejected because extension not found” Tried by calling 2000 from analog phone [Digium-FXS-J4] - getting some engage/disconnected tone while pressing digts [2000] on phone itself. Welcome for your valuable suggestions and comments. Thank You in advance. Regards, Garge. -- _ -- 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 -- Juan. Linux User #441131 -- _ -- 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] How can I record the conversations in a conference call?
I use the option 'r' on 1.4, to record the meetme application. Asterisk leaves these records at /var/lib/asterisk/sounds/meetmeXX. take a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe if you need more information. Regards. 2010/4/14 Renato bianchini renato...@yahoo.com.br Hello, I wanna record the conversations in a conference call, anyone know how can I do it? I've already configurated a room on meetme.conf but I don't know as I can record the conversations. I'm using SUSE 11 and Asterisk 1.6.2. Thank you so much for help me. Bye -- Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10http://br.rd.yahoo.com/mail/taglines/mail/*http://br.maisbuscados.yahoo.com/- Celebridadeshttp://br.rd.yahoo.com/mail/taglines/mail/*http://br.maisbuscados.yahoo.com/celebridades/- Músicahttp://br.rd.yahoo.com/mail/taglines/mail/*http://br.maisbuscados.yahoo.com/m%C3%BAsica/- Esporteshttp://br.rd.yahoo.com/mail/taglines/mail/*http://br.maisbuscados.yahoo.com/esportes/ -- _ -- 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 -- Juan. -- _ -- 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] How can I record the conversations in a conference call?
Please note: A Zaptel timer must be present for conferencing to work!, but if the user does not have ZAP/DAHDI hardware, he can use ZAP/DAHDI DUMMY 2010/4/15 Luki lugos...@gmail.com I use the option 'r' on 1.4, to record the meetme application. Asterisk leaves these records at /var/lib/asterisk/sounds/meetmeXX. That option only works for conferences using ZAP/DAHDI hardware. You can, however, start to Monitor() the channel prior to entering the conference, but you should only do it for the first caller. MeetMeCount() will help. The caveat is that if the first caller disconnects, the remainder of the conference will not be recorded. If anyone has a better solution, please tell us :). Luki -- _ -- 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 -- Juan. -- _ -- 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] Problems in Asterisk Real Time (Urgent help )
Have you check if MySql is already running? Have you check HD space? regards. 2010/2/24 ahmed magdy amagdy.ibra...@gmail.com Hello, Asterisk Real time database worked on astersik 1.6.2.0 but now i am working on Asterisk to latest version which is 1.6.2.2 ,there is a a warning [Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Feb 24 16:26:14] NOTICE[4053]: chan_sip.c:21500 handle_request_register: Registration from '555sip:5...@192.168.50.109 sip%3a...@192.168.50.109' failed for '192.168.50.105' - No matching peer found is there a problem in version compatability? if anyone knows anything ,help me please. -- Ahmed Magdy Mahmoud -- _ -- 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 -- Juan. -- _ -- 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
Re: [asterisk-users] Open source or low-budget recommendation for call-center software
I think Vicidial, works great. Regards. 2010/2/22 Apa Minerala apaminer...@yahoo.com Hello, We used to recommend a commercial software but client is a small callcenter who cannot afford something big. Would you recommend something open-source which could work for a 40-seater? Thank you, Tudor www.sunabasarabia.com Moldova 11c/min Romania 2c/min $1 de test de la bun inceput -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.14 - Play an audio or signal
Thanks a lot Alec, I´ll check 2009/12/22 Alec Davis siva...@paradise.net.nz straight from our 1.6.1 dialplan, don't know about 1.2.14. exten = s,n,Set(LIMIT_WARNING_FILE=beep) exten = s,n,Set(LIMIT_TIMEOUT_FILE=call-terminated) ;terminate after 1 hour, start beep warnings at 10 minutes, every 5 minutes exten = s,n,Dial(${AVAILCHAN_NOSESSION}/${ARG2}#,,rL(360:300:30)) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Juan David Diaz *Sent:* Tuesday, 22 December 2009 11:40 a.m. *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Asterisk 1.2.14 - Play an audio or signal Good Day List Users, Is there any way to play an audiofile or at least a beep into an established call, I want to do this event each 3 minutes in the call, for now I have a shell to get the call time and evaluate the 3 minutes.do you know any way to play that sound? I tried app_inject, it works really nice in asterisk 1.4.X releases; but my PBX runs 1.2.14 and It can´t be upgraded (policy reasons). Regards and Thanks every one. -- Juan. ___ -- 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 -- Juan. ___ -- 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
[asterisk-users] Asterisk 1.2.14 - Play an audio or signal
Good Day List Users, Is there any way to play an audiofile or at least a beep into an established call, I want to do this event each 3 minutes in the call, for now I have a shell to get the call time and evaluate the 3 minutes.do you know any way to play that sound? I tried app_inject, it works really nice in asterisk 1.4.X releases; but my PBX runs 1.2.14 and It can´t be upgraded (policy reasons). Regards and Thanks every one. -- Juan. ___ -- 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