Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
I understand, I'll contact Digium technical support. Thanks a lot! On Fri, Sep 21, 2012 at 2:15 PM, Shaun Ruffell sruff...@digium.com wrote: On Fri, Sep 21, 2012 at 01:59:28PM -0300, equis software wrote: Shaun, here is the last test that I made. Time 0: IVR1 (with digiumA) have erros (Master changed to TE2/0/1) IVR2 (with digiumB) OK Time 1: (just swap spams) IVR1 (with digiumB) OK IVR2 (with digiumA) have erros (Master changed to TE2/0/1) I'm seriously thinking that the problem is in the digiumA What you think? Honestly, it's hard to say without really looking at your complete configuration. At this point it might be quickest if you contact Digium technical support since they are in a position to collect complete information about your setup. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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 -- _ -- 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] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
Shaun, here is the last test that I made. Time 0: IVR1 (with digiumA) have erros (Master changed to TE2/0/1) IVR2 (with digiumB) OK Time 1: (just swap spams) IVR1 (with digiumB) OK IVR2 (with digiumA) have erros (Master changed to TE2/0/1) I'm seriously thinking that the problem is in the digiumA What you think? On Mon, Sep 17, 2012 at 12:21 PM, equis software equissoftw...@gmail.comwrote: and I was happy without this message!!! Thanks Shaun On Mon, Sep 17, 2012 at 11:22 AM, Shaun Ruffell sruff...@digium.comwrote: On Mon, Sep 17, 2012 at 09:33:12AM -0300, equis software wrote: Took 3 days without errors like dahdi: Master changed to TE2/0/2 having installed the dahdi 2.6.1 A heads up: With DAHDI-Linux 2.5.0 and up you will only see the Master changed messages when you set the debug module parameter [1] [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=9299 You can enable this flag without reloading like: $ echo 1 /sys/module/dahdi/parameters/debug Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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 -- _ -- 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] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
Took 3 days without errors like dahdi: Master changed to TE2/0/2 having installed the dahdi 2.6.1 But I have warnings that I copy below [Sep 17 09:25:27] WARNING[24233] mtp.c: MTP2 timer T3 timeout (failed to receive 'N', or 'E' after sending 'O'), initial alignment failed on link 'i1'. [Sep 17 09:25:28] NOTICE[24233] mtp.c: Got event on link 'i1': 5 (0/500). [Sep 17 09:25:28] NOTICE[2872] l4isup.c: Unhandled zaptel event 0x5 on CIC=30. [Sep 17 09:25:28] NOTICE[2872] l4isup.c: Unhandled zaptel event 0x4 on CIC=30. [Sep 17 09:25:28] NOTICE[24233] mtp.c: Got event on link 'i1': 4 (0/11). [Sep 17 09:25:29] NOTICE[2872] l4isup.c: Unhandled zaptel event 0x5 on CIC=30. [Sep 17 09:25:29] NOTICE[24233] mtp.c: Got event on link 'i1': 5 (0/500). [Sep 17 09:25:29] NOTICE[24233] l4isup.c: T1 timeout (waiting for RLC) CIC=20. [Sep 17 09:25:29] WARNING[24233] mtp.c: No signalling links inservice and no cluster receivers alive, dropping packet! [Sep 17 09:25:29] WARNING[24233] chan_ss7.c: MTP is now UP on link 'i1'. [Sep 17 09:25:29] NOTICE[24233] mtp.c: Sending TRA to peer on link 'i1' [Sep 17 09:25:29] WARNING[24233] mtp.c: Got SLTM with unexpected sls=1, OPC=13152 DPC=8458 on 'i1/16' sls=0, state=5. [Sep 17 09:25:30] NOTICE[24233] l4isup.c: Got GROUP RESET message, opc=0x3360, dpc=0x210a, sls=0x1, cic=17, range=14. [Sep 17 09:25:30] NOTICE[24233] l4isup.c: Got GROUP RESET message, opc=0x3360, dpc=0x210a, sls=0x1, cic=1, range=14. [Sep 17 09:25:58] NOTICE[24233] l4isup.c: Process CGU, cic=1, range=14 [Sep 17 09:25:58] NOTICE[24233] l4isup.c: Process CGU, cic=33, range=30 [Sep 17 09:26:00] NOTICE[24233] l4isup.c: Got GROUP RESET message, opc=0x3360, dpc=0x210a, sls=0x1, cic=33, range=30. [Sep 17 09:26:00] NOTICE[24233] l4isup.c: Process CGU, cic=17, range=14 [Sep 17 09:26:13] NOTICE[2925] l4isup.c: Short read on linkset siuc CIC=24 (read only 0 of 160) errno=11 (Resource temporarily unavailable) (supressed 0). [Sep 17 09:26:58] NOTICE[24233] l4isup.c: Process CGU, cic=1, range=14 [Sep 17 09:27:00] NOTICE[24233] l4isup.c: Process CGU, cic=17, range=14 [Sep 17 09:27:17] NOTICE[24233] mtp.c: Got status indication 'OS' while INSERVICE on link 'i1'. [Sep 17 09:27:17] WARNING[24233] chan_ss7.c: MTP is now DOWN on link 'i1'. [Sep 17 09:27:17] NOTICE[24233] mtp.c: MTP changeover last_ack=34, last_sent=34, from schannel 16, no INSERVICE schannel found [Sep 17 09:27:17] NOTICE[24233] mtp.c: Failover not possible, no other signalling link and no other host available. [Sep 17 09:27:17] WARNING[24233] chan_ss7.c: MTP is now DOWN on link 'i1'. [Sep 17 09:27:17] NOTICE[2947] l4isup.c: Unhandled zaptel event 0x4 on CIC=2. [Sep 17 09:27:17] NOTICE[24233] mtp.c: Got event on link 'i1': 4 (0/500). [Sep 17 09:27:19] NOTICE[24233] mtp.c: Got event on link 'i1': 5 (0/500). [Sep 17 09:27:19] NOTICE[2947] l4isup.c: Unhandled zaptel event 0x5 on CIC=2. [Sep 17 09:27:19] NOTICE[2947] l4isup.c: Unhandled zaptel event 0x4 on CIC=2. [Sep 17 09:27:19] NOTICE[24233] mtp.c: Got event on link 'i1': 4 (0/11). [Sep 17 09:27:19] WARNING[24233] mtp.c: MTP2 timer T3 timeout (failed to receive 'N', or 'E' after sending 'O'), initial alignment failed on link 'i1'. [Sep 17 09:27:21] NOTICE[24233] mtp.c: Got event on link 'i1': 5 (0/500). [Sep 17 09:27:21] NOTICE[2947] l4isup.c: Unhandled zaptel event 0x5 on CIC=2. [Sep 17 09:27:21] WARNING[24233] chan_ss7.c: MTP is now UP on link 'i1'. [Sep 17 09:27:21] NOTICE[24233] mtp.c: Sending TRA to peer on link 'i1' [Sep 17 09:27:21] WARNING[24233] mtp.c: Got SLTM with unexpected sls=1, OPC=13152 DPC=8458 on 'i1/16' sls=0, state=5. [Sep 17 09:27:22] NOTICE[24233] l4isup.c: Got GROUP RESET message, opc=0x3360, dpc=0x210a, sls=0x1, cic=17, range=14. [Sep 17 09:27:22] NOTICE[24233] l4isup.c: Got GROUP RESET message, opc=0x3360, dpc=0x210a, sls=0x1, cic=1, range=14. On Fri, Sep 14, 2012 at 1:09 PM, Shaun Ruffell sruff...@digium.com wrote: [friendly request that you inline or bottom post and trim your replies. See http://brooksreview.net/2011/01/interleaved-email/ which makes most of the points I would make on the subject] On Fri, Sep 14, 2012 at 09:00:39AM -0300, equis software wrote: My test were... SIEMENS - LTG1 --- cable1 - SS7 --- IVR1 (have errors) |- LTG2 --- cable2 - SS7 --- IVR2 (OK) minutes later... SIEMENS - LTG1 --- cable1 - SS7 --- IVR2 (OK) |- LTG2 --- cable2 - SS7 --- IVR1 (have same errors) And your problem follows the server not the card or the cable? Have you tried other slots? Are you screwing the cards down? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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
Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
and I was happy without this message!!! Thanks Shaun On Mon, Sep 17, 2012 at 11:22 AM, Shaun Ruffell sruff...@digium.com wrote: On Mon, Sep 17, 2012 at 09:33:12AM -0300, equis software wrote: Took 3 days without errors like dahdi: Master changed to TE2/0/2 having installed the dahdi 2.6.1 A heads up: With DAHDI-Linux 2.5.0 and up you will only see the Master changed messages when you set the debug module parameter [1] [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=9299 You can enable this flag without reloading like: $ echo 1 /sys/module/dahdi/parameters/debug Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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 -- _ -- 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] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
My test were... SIEMENS - LTG1 --- cable1 - SS7 --- IVR1 (have errors) |- LTG2 --- cable2 - SS7 --- IVR2 (OK) minutes later... SIEMENS - LTG1 --- cable1 - SS7 --- IVR2 (OK) |- LTG2 --- cable2 - SS7 --- IVR1 (have same errors) Errors: Sep 12 11:49:42 call3 kernel: [1018444.069418] dahdi: Master changed to TE2/0/1 Sep 12 11:49:48 call3 kernel: [1018449.724411] dahdi: Master changed to TE2/0/2 Sep 12 11:49:48 call3 kernel: [1018449.823093] dahdi: Master changed to TE2/0/1 Sep 12 11:49:52 call3 kernel: [1018454.175277] dahdi: Master changed to TE2/0/2 Sep 12 11:49:52 call3 kernel: [1018454.198138] dahdi: Master changed to TE2/0/1 Sep 12 11:49:53 call3 kernel: [1018455.493002] dahdi: Master changed to TE2/0/2 Sep 12 11:49:53 call3 kernel: [1018455.593648] dahdi: Master changed to TE2/0/1 On Fri, Sep 14, 2012 at 5:31 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 09/14/2012 10:25 AM, A J Stiles wrote: [snip] It could be nothing more than a dry solder joint on one of the RJ45s. For the sake of five minutes' work with a soldering iron, that's got to be worth eliminating. Wouldn't that void your warranty? I would take it up with Digium support and let them sort it out. Regards, Patrick -- __**__**_ -- 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-**usershttp://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] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
Shaun, could it be related to http://lists.digium.com/pipermail/asterisk-users/2012-March/271304.html ?? I'm using DAHDI Version: 2.4.1.2 On Fri, Sep 14, 2012 at 9:00 AM, equis software equissoftw...@gmail.comwrote: My test were... SIEMENS - LTG1 --- cable1 - SS7 --- IVR1 (have errors) |- LTG2 --- cable2 - SS7 --- IVR2 (OK) minutes later... SIEMENS - LTG1 --- cable1 - SS7 --- IVR2 (OK) |- LTG2 --- cable2 - SS7 --- IVR1 (have same errors) Errors: Sep 12 11:49:42 call3 kernel: [1018444.069418] dahdi: Master changed to TE2/0/1 Sep 12 11:49:48 call3 kernel: [1018449.724411] dahdi: Master changed to TE2/0/2 Sep 12 11:49:48 call3 kernel: [1018449.823093] dahdi: Master changed to TE2/0/1 Sep 12 11:49:52 call3 kernel: [1018454.175277] dahdi: Master changed to TE2/0/2 Sep 12 11:49:52 call3 kernel: [1018454.198138] dahdi: Master changed to TE2/0/1 Sep 12 11:49:53 call3 kernel: [1018455.493002] dahdi: Master changed to TE2/0/2 Sep 12 11:49:53 call3 kernel: [1018455.593648] dahdi: Master changed to TE2/0/1 On Fri, Sep 14, 2012 at 5:31 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 09/14/2012 10:25 AM, A J Stiles wrote: [snip] It could be nothing more than a dry solder joint on one of the RJ45s. For the sake of five minutes' work with a soldering iron, that's got to be worth eliminating. Wouldn't that void your warranty? I would take it up with Digium support and let them sort it out. Regards, Patrick -- __**__**_ -- 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-**usershttp://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] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
After exchanging the cable with other equipment was running smoothly in my computer the problem persisted while the other team the cable that could be bad worked. With this test done, now suspect the problem I have it on the Digium card. I perform the test specified in: http://kb.digium.com/articles/Troubleshooting/How-do-I-run-a-pattern-loopback-test-on-a-Digium-E1-T1-card?retURL=%2Fpopup=false With this results call3 tools # ./patlooptest /dev/dahdi/1 -t 300 -v Using Timeout of 300 Seconds Going for it... Timeout achieved Ending Program Test ran 36499 loops of 2039 bytes/loop with 0 errors call3 tools # ./patlooptest /dev/dahdi/32 -t 300 -v Using Timeout of 300 Seconds Going for it... Timeout achieved Ending Program Test ran 36550 loops of 2039 bytes/loop with 0 errors This test would indicate that the board has no problem ... but where is the fault? On Wed, Sep 12, 2012 at 12:56 PM, equis software equissoftw...@gmail.comwrote: Thanks, I'll try changing cables. On Wed, Sep 12, 2012 at 12:21 PM, Shaun Ruffell sruff...@digium.comwrote: On Wed, Sep 12, 2012 at 12:19:41PM -0300, equis software wrote: I have a server with an asterisk ss7 link connected to a Siemens working well for over a year. A few days ago I started having problems with signaling. I found the following logs in / var / log / messages [1018427.030959] dahdi: Master changed to TE2/0/2 [1018427.120740] dahdi: Master changed to TE2/0/1 [1018427.789173] dahdi: Master changed to TE2/0/2 [1018427.884828] dahdi: Master changed to TE2/0/1 [1018431.209621] dahdi: Master changed to TE2/0/2 [1018431.300289] dahdi: Master changed to TE2/0/1 [1018434.763742] dahdi: Master changed to TE2/0/2 If I stop the asterisk and dahdi driver just let the messages continue to appear. Any ideas?? My guess is you have a cabling problem and that span TE2/0/1 is going into and out of alarm. Each time it goes into alarm the core of DAHDI will look for a new span to use as a timing source for asterisk. If TE2/0/1 is the preferred timing source, when it comes out of alarm, DAHDI will switch the timing back to it. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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 -- _ -- 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] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
My ultimate intention is to blame the board, but we have changed teh cable and the LTG in the central and the problem continues, while the same LTG with another server works fine. Is too strange!! On Thu, Sep 13, 2012 at 4:14 PM, Danny Nicholas da...@debsinc.com wrote: I would still consider the cable. They are funny things and they make nice meters to check them without putting your communications at risk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Sharp Sent: Thursday, September 13, 2012 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message On 9/13/2012 2:31 PM, equis software wrote: After exchanging the cable with other equipment was running smoothly in my computer the problem persisted while the other team the cable that could be bad worked. With this test done, now suspect the problem I have it on the Digium card. I perform the test specified in: http://kb.digium.com/articles/Troubleshooting/How-do-I-run-a-pattern-l oopback-test-on-a-Digium-E1-T1-card?retURL=%2Fpopup=false With this results Have you done loopback testing with your telco to make sure your line is clean? I'd point fingers there before blaming the Digium card or a cable. -- _ -- 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 -- _ -- 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] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
I have a server with an asterisk ss7 link connected to a Siemens working well for over a year. A few days ago I started having problems with signaling. I found the following logs in / var / log / messages Sep 12 11:49:25 call3 kernel: [1018427.030959] dahdi: Master changed to TE2/0/2 Sep 12 11:49:25 call3 kernel: [1018427.120740] dahdi: Master changed to TE2/0/1 Sep 12 11:49:26 call3 kernel: [1018427.789173] dahdi: Master changed to TE2/0/2 Sep 12 11:49:26 call3 kernel: [1018427.884828] dahdi: Master changed to TE2/0/1 Sep 12 11:49:29 call3 kernel: [1018431.209621] dahdi: Master changed to TE2/0/2 Sep 12 11:49:29 call3 kernel: [1018431.300289] dahdi: Master changed to TE2/0/1 Sep 12 11:49:33 call3 kernel: [1018434.763742] dahdi: Master changed to TE2/0/2 If I stop the asterisk and dahdi driver just let the messages continue to appear. Any ideas?? -- _ -- 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] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
Sorry, in the last line I try to say ... With asterisk stopped, and only driver dahdi up, messages still appear... On Wed, Sep 12, 2012 at 12:19 PM, equis software equissoftw...@gmail.comwrote: I have a server with an asterisk ss7 link connected to a Siemens working well for over a year. A few days ago I started having problems with signaling. I found the following logs in / var / log / messages Sep 12 11:49:25 call3 kernel: [1018427.030959] dahdi: Master changed to TE2/0/2 Sep 12 11:49:25 call3 kernel: [1018427.120740] dahdi: Master changed to TE2/0/1 Sep 12 11:49:26 call3 kernel: [1018427.789173] dahdi: Master changed to TE2/0/2 Sep 12 11:49:26 call3 kernel: [1018427.884828] dahdi: Master changed to TE2/0/1 Sep 12 11:49:29 call3 kernel: [1018431.209621] dahdi: Master changed to TE2/0/2 Sep 12 11:49:29 call3 kernel: [1018431.300289] dahdi: Master changed to TE2/0/1 Sep 12 11:49:33 call3 kernel: [1018434.763742] dahdi: Master changed to TE2/0/2 If I stop the asterisk and dahdi driver just let the messages continue to appear. Any ideas?? -- _ -- 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] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
Thanks, I'll try changing cables. On Wed, Sep 12, 2012 at 12:21 PM, Shaun Ruffell sruff...@digium.com wrote: On Wed, Sep 12, 2012 at 12:19:41PM -0300, equis software wrote: I have a server with an asterisk ss7 link connected to a Siemens working well for over a year. A few days ago I started having problems with signaling. I found the following logs in / var / log / messages [1018427.030959] dahdi: Master changed to TE2/0/2 [1018427.120740] dahdi: Master changed to TE2/0/1 [1018427.789173] dahdi: Master changed to TE2/0/2 [1018427.884828] dahdi: Master changed to TE2/0/1 [1018431.209621] dahdi: Master changed to TE2/0/2 [1018431.300289] dahdi: Master changed to TE2/0/1 [1018434.763742] dahdi: Master changed to TE2/0/2 If I stop the asterisk and dahdi driver just let the messages continue to appear. Any ideas?? My guess is you have a cabling problem and that span TE2/0/1 is going into and out of alarm. Each time it goes into alarm the core of DAHDI will look for a new span to use as a timing source for asterisk. If TE2/0/1 is the preferred timing source, when it comes out of alarm, DAHDI will switch the timing back to it. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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 -- _ -- 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] chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway!
Hi all. I have this problem with my Digium 2E1 card and PRI, for hours It works well, with some meesages... [Aug 10 09:20:31] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 But PRI continue uphours later... PRI go down. I thought the problem was in the telco, but the strange thing is that I have a loop cable in the second E1 and when I scan both E1 are with alarms=LMFA/OK and i have only the first E1 connected to the telco!! messages [Aug 10 09:20:31] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Aug 10 09:20:32] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Aug 10 09:20:32] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Aug 10 09:20:32] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Aug 10 09:20:32] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Aug 10 09:20:36] WARNING[32270] chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway! gentoo1 ~ # dahdi_scan [1] active=yes alarms=LMFA/OK description=T2XXP (PCI) Card 0 Span 1 name=TE2/0/1 manufacturer=Digium devicetype=Wildcard TE220 (5th Gen) location=Board ID Switch 0 basechan=1 totchans=31 irq=16 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI,HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS/CRC4 [2] active=yes alarms=LMFA/OK description=T2XXP (PCI) Card 0 Span 2 name=TE2/0/2 manufacturer=Digium devicetype=Wildcard TE220 (5th Gen) location=Board ID Switch 0 basechan=32 totchans=31 irq=16 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI,HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS/CRC4 -- _ -- 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] DAHDI problems
Is a normal functionality? when I do #dahdi_cfg -vv In my Asterisk console shows this [Jul 24 13:39:08] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 If I do this a lot of times...then [Jul 24 13:39:20] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24 13:39:21] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24 13:39:21] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24 13:39:21] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24 13:39:21] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24 13:39:21] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24 13:39:22] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24 13:39:22] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24 13:39:22] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 [Jul 24 13:39:22] WARNING[30263]: chan_dahdi.c:2840 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! [Jul 24 13:39:22] WARNING[30264]: chan_dahdi.c:4332 handle_alarms: Detected alarm on channel 1: Yellow Alarm . . I'm using Libpri 1.4.11 Dahdi 2.6.0 Asterisk 1.4.40 -- _ -- 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 problems
I´m using crc4 conf span = 1,0,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 The alarm clrears few seconds later... On Tue, Jul 24, 2012 at 2:14 PM, Mitul Limbani mi...@enterux.in wrote: Try enabling crc4 on ur config. Mitul On Jul 24, 2012 10:38 PM, Russ Meyerriecks rmeyerrie...@digium.com wrote: On Tue, Jul 24, 2012 at 01:45:01PM -0300, equis software wrote: Is a normal functionality? when I do #dahdi_cfg -vv If I do this a lot of times...then In rapid succession, or over spread out over a long timeframe? [Jul 24 13:39:22] WARNING[30264]: chan_dahdi.c:4332 handle_alarms: Detected alarm on channel 1: Yellow Alarm . . Is the channel hanging in yellow indefinitly? -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.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 -- _ -- 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] DAHDI problems
The alarm appears inmediately... On Tue, Jul 24, 2012 at 2:42 PM, Mitul Limbani mi...@enterux.in wrote: If u r then try removing it. Mitul On Jul 24, 2012 11:09 PM, equis software equissoftw...@gmail.com wrote: I´m using crc4 conf span = 1,0,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 The alarm clrears few seconds later... On Tue, Jul 24, 2012 at 2:14 PM, Mitul Limbani mi...@enterux.in wrote: Try enabling crc4 on ur config. Mitul On Jul 24, 2012 10:38 PM, Russ Meyerriecks rmeyerrie...@digium.com wrote: On Tue, Jul 24, 2012 at 01:45:01PM -0300, equis software wrote: Is a normal functionality? when I do #dahdi_cfg -vv If I do this a lot of times...then In rapid succession, or over spread out over a long timeframe? [Jul 24 13:39:22] WARNING[30264]: chan_dahdi.c:4332 handle_alarms: Detected alarm on channel 1: Yellow Alarm . . Is the channel hanging in yellow indefinitly? -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.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 -- _ -- 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 -- _ -- 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] DAHDI problems
I was really worried because I have a rare problem fall of link (once per month) and I don't know why On Tue, Jul 24, 2012 at 3:02 PM, equis software equissoftw...@gmail.comwrote: May be, I suspect of it because in another server doesn't happend. On Tue, Jul 24, 2012 at 2:52 PM, Russ Meyerriecks rmeyerrie...@digium.com wrote: On Tue, Jul 24, 2012 at 02:36:04PM -0300, equis software wrote: The alarm clrears few seconds later... Then I don't see anything wrong here? You're just re-configuring the span and the pri stack is dealing with the temporary outage. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.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 -- _ -- 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 callers with Callback option without lose their place
Is there any option in Asterisk distribution of this? Thanks. -- _ -- 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] Queue callers with Callback option without lose their place
was what I was afraid ... Thanks 2012/5/31 Miguel Molina mfmolina-lis...@millenium.com.co Known as Virtual Hold, you'll have to program inside asterisk to achieve that. El 31/05/12 10:48, equis software escribió: Is there any option in Asterisk distribution of this? Thanks. -- _ -- 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 -- _ -- 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] Anybody knows a good SBC to download?
Anybody knows a good SBC to download? Thanks -- _ -- 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] Finish ChanSpy() when channel spied hangs up
Yes, Asterisk 1.8.10.0, chanspy have the option 'E', which terminate the
function when the call hangs.
Thanks to all.
On Thu, Mar 8, 2012 at 12:05 PM, Jim Dickenson dicken...@cfmc.com wrote:
I had submitted a patch some time ago to add option s to chanspy. This
would cause chanspy to exit once the specified change was not longer there.
I do not know if it ever got into a released version as I use ABE. It was
not in 1.6 but might be in 1.8.
--
Jim Dickenson
mailto:dicken...@cfmc.com dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 8, 2012, at 4:20 AM, equis software wrote:
I need call to C every time that A call to B, but when A-B hangs up i need
to hang up Asterisk-C call too.
Anyboby know another solution?
On Wed, Mar 7, 2012 at 2:51 PM, equis software equissoftw...@gmail.comwrote:
Here's my dialplan...
[default]
exten = _X.,1,System(echo -e Channel: SIP/519912@SOFTSWITCH\\nContext:
spy\\nExtension: 23\\nSet:SPYCHANNEL=${CHANNEL} /tmp/${UNIQUEID}.call)
exten = _X.,n,System(mv /tmp/${UNIQUEID}.call
/var/spool/asterisk/outgoing/)
exten = _X.,n,Dial(SIP/${EXTEN}@SOFTSWITCH)
[spy]
exten = s,1,Answer
exten = s,2,Chanspy(${SPYCHANNEL}|q)
exten = s,3,Hangup
A call to B
and C (519912) is called by Asterisk to spy the call.
Whe the A-B conversation over, C continue connected to Asterisk, I need
Asterisk hangs up this call.
In my case C is another machine that records the call and can´t hang up
when A-B has finished because it doesn't know.
I don't know if i'm clear
On Wed, Mar 7, 2012 at 1:12 PM, Jonas Kellens
jonas.kell...@telenet.bewrote:
**
Doesn't this automatically finish ?
Jonas.
On 03/07/2012 05:03 PM, equis software wrote:
Is there any way to do this?
Thanks
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Re: [asterisk-users] Finish ChanSpy() when channel spied hangs up
I need call to C every time that A call to B, but when A-B hangs up i need
to hang up Asterisk-C call too.
Anyboby know another solution?
On Wed, Mar 7, 2012 at 2:51 PM, equis software equissoftw...@gmail.comwrote:
Here's my dialplan...
[default]
exten = _X.,1,System(echo -e Channel: SIP/519912@SOFTSWITCH\\nContext:
spy\\nExtension: 23\\nSet:SPYCHANNEL=${CHANNEL} /tmp/${UNIQUEID}.call)
exten = _X.,n,System(mv /tmp/${UNIQUEID}.call
/var/spool/asterisk/outgoing/)
exten = _X.,n,Dial(SIP/${EXTEN}@SOFTSWITCH)
[spy]
exten = s,1,Answer
exten = s,2,Chanspy(${SPYCHANNEL}|q)
exten = s,3,Hangup
A call to B
and C (519912) is called by Asterisk to spy the call.
Whe the A-B conversation over, C continue connected to Asterisk, I need
Asterisk hangs up this call.
In my case C is another machine that records the call and can´t hang up
when A-B has finished because it doesn't know.
I don't know if i'm clear
On Wed, Mar 7, 2012 at 1:12 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
**
Doesn't this automatically finish ?
Jonas.
On 03/07/2012 05:03 PM, equis software wrote:
Is there any way to do this?
Thanks
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Re: [asterisk-users] Ongoing attack from 188.138.100.16
fail2ban works perfect!! On Wed, Mar 7, 2012 at 12:47 PM, Jamie A. Stapleton jstaple...@computer-business.com wrote: Block them. They are one of the Internet's top bad IP addresses. http://www.threatstop.com/checkip -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Tuesday, March 06, 2012 7:29 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Ongoing attack from 188.138.100.16 I've been logging sip registrations from this IP address for 2 days now. I've emailed the domain's admin, but nothing seems to come of it. I've routed him into oblivion, but still, I think 50 requests a second for 2 days is a bit much. Any ideas? -- Take care and have fun, Mike Diehl. -- _ -- 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 -- _ -- 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] Finish ChanSpy() when channel spied hangs up
Is there any way to do this? Thanks -- _ -- 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] Finish ChanSpy() when channel spied hangs up
Here's my dialplan...
[default]
exten = _X.,1,System(echo -e Channel: SIP/519912@SOFTSWITCH\\nContext:
spy\\nExtension: 23\\nSet:SPYCHANNEL=${CHANNEL} /tmp/${UNIQUEID}.call)
exten = _X.,n,System(mv /tmp/${UNIQUEID}.call
/var/spool/asterisk/outgoing/)
exten = _X.,n,Dial(SIP/${EXTEN}@SOFTSWITCH)
[spy]
exten = s,1,Answer
exten = s,2,Chanspy(${SPYCHANNEL}|q)
exten = s,3,Hangup
A call to B
and C (519912) is called by Asterisk to spy the call.
Whe the A-B conversation over, C continue connected to Asterisk, I need
Asterisk hangs up this call.
In my case C is another machine that records the call and can´t hang up
when A-B has finished because it doesn't know.
I don't know if i'm clear
On Wed, Mar 7, 2012 at 1:12 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
**
Doesn't this automatically finish ?
Jonas.
On 03/07/2012 05:03 PM, equis software wrote:
Is there any way to do this?
Thanks
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[asterisk-users] .call files in /var/spool/asterisk/outgoing
Hi !
I have a python script that create and move .call files to
/var/spool/asterisk/outgoing
Sometimes...(in this case after 500 successfull calls) Asterisk don´t make
the calls and the .call files are in the outgoing forever...
Any Ideas?
I'm using Asterisk 1.4.22 (in 1.4.36 was the same behavior)
In my python script I move .call files using ...
import shutil
shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
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Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
Yes, same server, same filesystem...
On Fri, Aug 12, 2011 at 12:26 PM, Roger Burton West ro...@firedrake.orgwrote:
On Fri, Aug 12, 2011 at 12:23:22PM -0300, equis software wrote:
shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
Are both /var/tmp and /var/spool/asterisk/outgoing on the same
filesystem?
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Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
I made 500 calls but not simultaneously. My script checks that there are no more than 3 .call files in the outgoing. I change in my python script, now move file with os.system... import os os.system (mv+ + tmpFile + + callFile) see what happens... On Fri, Aug 12, 2011 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote: Also, keep in mind that the spooling mechanism has mechanical limits based on processor speed, line capacity, etc. If I were doing 500 calls, I would use sleep to space the starting of the calls (maybe 5 or 15 second intervals). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: Friday, August 12, 2011 10:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote: Yes, same server, same filesystem... I don't do Python, but a web search for shutil.move suggests that it doesn't reliably use the rename syscall. Might be worth shelling out to your system's mv command. R -- _ -- 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 -- _ -- 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] Outgoing agent´s calls
I found this solution... In every line that Agent want to make an outgoing call, this call is routed by my softswitch to Asterisk, in dialplan using func_odbc.conf I could know if there any agent logged in this line because I have this information in my DB. Then I set accoutncode field from CDR with the agent id. If there aren´t any agent logged in this line I reject the call. Thanks! On Thu, Feb 3, 2011 at 12:02 PM, Danny Nicholas da...@debsinc.com wrote: Then DISA (I had it as DASI in OP because I’m working from not so good memory) is probably your best bet. It is a simple built-in feature that let’s you get an access code in the dialplan before performing an action such as dialing. Check this link http://nerdvittles.com/index.php?p=73 -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Thursday, February 03, 2011 6:14 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Outgoing agent´s calls Yes, my agents dial “willy-nilly”... I can´t use the ex-girlfriend because, the line numbers that uses the agents are diferent. May be agent 1 today use line number 553455 and tomorrow 553461... On Wed, Feb 2, 2011 at 3:51 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Wednesday, February 02, 2011 12:26 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Outgoing agent´s calls Hi, is there any way to manage outgoing calls from agents? Mi agents are answering in pstn lines. I can send agents outgoing calls to my Asterisk but I don't know wich agent is making the call...because, may be he is unregister... Is there any solution? Thanks You could start with DASI and ex-girlfriend logic in your dialplan. I’m assuming now that your agents dial “willy-nilly” (with no restrictions and you find out what they did when you read the CDR). -- _ -- 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 -- _ -- 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] Outgoing agent´s calls
Yes, my agents dial “willy-nilly”... I can´t use the ex-girlfriend because, the line numbers that uses the agents are diferent. May be agent 1 today use line number 553455 and tomorrow 553461... On Wed, Feb 2, 2011 at 3:51 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Wednesday, February 02, 2011 12:26 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Outgoing agent´s calls Hi, is there any way to manage outgoing calls from agents? Mi agents are answering in pstn lines. I can send agents outgoing calls to my Asterisk but I don't know wich agent is making the call...because, may be he is unregister... Is there any solution? Thanks You could start with DASI and ex-girlfriend logic in your dialplan. I’m assuming now that your agents dial “willy-nilly” (with no restrictions and you find out what they did when you read the CDR). -- _ -- 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] Outgoing agent´s calls
Hi, is there any way to manage outgoing calls from agents? Mi agents are answering in pstn lines. I can send agents outgoing calls to my Asterisk but I don't know wich agent is making the call...because, may be he is unregister... Is there any solution? Thanks -- _ -- 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] Error messages with chan_dahdi
On Sat, Dec 4, 2010 at 2:11 PM, Shaun Ruffell sruff...@digium.com wrote: On 12/4/10 9:15 AM, equis software wrote: HI, I'm using asterisk-1.4.24, dahdi-linux-complete-2.4.0+2.4.0 and libpri-1.4.11.4 When dial, when 492131 answer, in console appear some error messages -- AGI Script Executing Application: (DIAL) Options: (DAHDI/g1/492131|60) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/492131 [Dec 4 11:15:59] WARNING[7669]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable echo cancellation on channel 2 (No such device) -- DAHDI/2-1 is ringing [Dec 4 11:16:02] WARNING[7669]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable echo cancellation on channel 2 (No such device) These No such device errors when trying to enable the echocan are most likely the result of not having configured an echocan for the channel in /etc/dahdi/system.conf. See line 309 in http://svn.asterisk.org/view/dahdi/tools/trunk/system.conf.sample?view=markup Yes, I wasn't cofigure the echocan, thanks! -- DAHDI/2-1 answered DAHDI/1-1 -- Native bridging DAHDI/1-1 and DAHDI/2-1 [Dec 4 11:16:02] ERROR[7669]: chan_dahdi.c:8735 dahdi_pri_error: ROSE REJECT: [Dec 4 11:16:02] ERROR[7669]: chan_dahdi.c:8735 dahdi_pri_error: INVOKE ID: 3 [Dec 4 11:16:02] ERROR[7669]: chan_dahdi.c:8735 dahdi_pri_error: PROBLEM: Invoke: Unrecognized Operation These errors I don't know about off the top of my head (and they are probably the more critical ones for you I'm guessing). Yes again, this is my real problem, despite the error messages,I can make the call without problems, but I'm worried about it... -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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 -- _ -- 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] Version compatibility question...
Hi, Could I install Asterisk 1.4.19, Dahdi 2.4.0 and libpri 1.4.3 ?? Thanks -- _ -- 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 + openBTS
Do you know if OpenBTS support handoff? Thanks On Fri, Aug 20, 2010 at 12:32 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Fri, Aug 20, 2010 at 10:41 AM, Tim Panton t...@westhawk.co.uk wrote: On 19 Aug 2010, at 20:59, Randy R wrote: On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News) alansli...@gmail.com wrote: On 19/08/10 18:20, equis software wrote: I want to know about asterisk and openBTS This island runs it's GSM network on OpenBTS: http://www.niueisland.com/ This was the place he presented about. Read the blog here: http://openbts.sourceforge.net/NiuePilot/ and more about the installation here: http://vuc.me/2010/island-telephony-adventure/ I was part of the team that went to Niue to install OpenBTS, I'm happy to answer questions if you have them, although I'm not the radio guy - asterisk is more my thing :-) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk In all reality, Asterisk could be substituted with any other platform. All the magic happens in the USRP, OpenBTS, and the cellular phones. Asterisk is merely handling the routing and voice, same as it ever was. It is just the top of the stack. I have two USRPs and a handful of daughter boards, and yes I have two flex 800s that have been physically altered so they can also be flex 1800s with a simple command line. These are the boards you want for GSM (Cellular). There is also a project to be able to listen into phone calls (thanks to the French making encryption so weak) besides a ton of other applications that can be dreamed up. You can do passive radar, track people that have cell phones powered on, RFID (Free tolls anyone?), WiFi, heck, you can even kill people with certain types of pacemakers. While OpenBTS is cool and is on topic with Asterisk, read up on GNURadio and all the projects and applications you can come up with. It is really cool technology. Start here http://gnuradio.org/redmine/projects/show/gnuradio but you can easily find things like this http://tech.mit.edu/V128/N30/subway/Defcon_Presentation.pdf or come up with your own with a bit of imagination and skillz. Thanks, Steve Totaro -- _ -- 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 + openBTS
Hi Tim, I'm not a radio guy too! I saw your name on the test in Niue. I have a softswitch. Can I replace Asterisk by my softswitch? Thanks On Fri, Aug 20, 2010 at 12:32 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Fri, Aug 20, 2010 at 10:41 AM, Tim Panton t...@westhawk.co.uk wrote: On 19 Aug 2010, at 20:59, Randy R wrote: On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News) alansli...@gmail.com wrote: On 19/08/10 18:20, equis software wrote: I want to know about asterisk and openBTS This island runs it's GSM network on OpenBTS: http://www.niueisland.com/ This was the place he presented about. Read the blog here: http://openbts.sourceforge.net/NiuePilot/ and more about the installation here: http://vuc.me/2010/island-telephony-adventure/ I was part of the team that went to Niue to install OpenBTS, I'm happy to answer questions if you have them, although I'm not the radio guy - asterisk is more my thing :-) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk In all reality, Asterisk could be substituted with any other platform. All the magic happens in the USRP, OpenBTS, and the cellular phones. Asterisk is merely handling the routing and voice, same as it ever was. It is just the top of the stack. I have two USRPs and a handful of daughter boards, and yes I have two flex 800s that have been physically altered so they can also be flex 1800s with a simple command line. These are the boards you want for GSM (Cellular). There is also a project to be able to listen into phone calls (thanks to the French making encryption so weak) besides a ton of other applications that can be dreamed up. You can do passive radar, track people that have cell phones powered on, RFID (Free tolls anyone?), WiFi, heck, you can even kill people with certain types of pacemakers. While OpenBTS is cool and is on topic with Asterisk, read up on GNURadio and all the projects and applications you can come up with. It is really cool technology. Start here http://gnuradio.org/redmine/projects/show/gnuradio but you can easily find things like this http://tech.mit.edu/V128/N30/subway/Defcon_Presentation.pdf or come up with your own with a bit of imagination and skillz. Thanks, Steve Totaro -- _ -- 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 + openBTS
I want to know about asterisk and openBTS If anybody made any test and experience... Thanks -- _ -- 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 + openBTS
May be he was David Burguess, another founder is Harvind Samra ... Do you know about any Equipment working? On Thu, Aug 19, 2010 at 2:27 PM, Alan Lord (News) alansli...@gmail.comwrote: On 19/08/10 18:20, equis software wrote: I want to know about asterisk and openBTS If anybody made any test and experience... I saw a presentation a few months ago where one of the openBTS project founders talked about one early system they set up on a very small and remote Pacific island along with Asterisk. If I can remember/find anything more I'll post here. Cheers Alan -- The Open Learning Centre http://www.theopenlearningcentre.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 -- _ -- 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] WARNING message when play
This are the console messages with AGI debugging AGI Rx STREAM FILE msgBienvenida112 1234567890*# -- Playing 'msgBienvenida112' (escape_digits=1234567890*#) (sample_offset 0) [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write() failed: Broken pipe On Fri, Jun 11, 2010 at 4:23 PM, Steve Edwards asterisk@sedwards.comwrote: On Fri, 11 Jun 2010, equis software wrote: When I use an eagi script when play a message appear a lot of warning messages, but it play very well I´m using Asterisk 1.4.32 dahdi-linux-2.3.0.1 chan_ss7-1.4.1 Any ideas?? -- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0) [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe I don't use EAGI, and coming from file.c may indicate otherwise, but broken pipe messages and AGI may mean you are violating the AGI protocol. Enabling AGI debugging and watching the console output may give you a clue. EAGI implies you are doing something with the incoming audio on FD 3. Any chance you are closing FD3? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] WARNING message when play
In Asterisk 1.4.22 it doesn't happend, in version 1.4.23.1 and above appear this messages On Mon, Jun 14, 2010 at 9:08 AM, equis software equissoftw...@gmail.comwrote: This are the console messages with AGI debugging AGI Rx STREAM FILE msgBienvenida112 1234567890*# -- Playing 'msgBienvenida112' (escape_digits=1234567890*#) (sample_offset 0) [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write() failed: Broken pipe On Fri, Jun 11, 2010 at 4:23 PM, Steve Edwards asterisk@sedwards.comwrote: On Fri, 11 Jun 2010, equis software wrote: When I use an eagi script when play a message appear a lot of warning messages, but it play very well I´m using Asterisk 1.4.32 dahdi-linux-2.3.0.1 chan_ss7-1.4.1 Any ideas?? -- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0) [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe I don't use EAGI, and coming from file.c may indicate otherwise, but broken pipe messages and AGI may mean you are violating the AGI protocol. Enabling AGI debugging and watching the console output may give you a clue. EAGI implies you are doing something with the incoming audio on FD 3. Any chance you are closing FD3? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] WARNING message when play
If I use CONTROL STREAM FILE the messages disapearIt´s this ok? On Mon, Jun 14, 2010 at 12:22 PM, Warren Selby wcse...@selbytech.comwrote: On Mon, Jun 14, 2010 at 8:27 AM, equis software equissoftw...@gmail.comwrote: In Asterisk 1.4.22 it doesn't happend, in version 1.4.23.1 and aboveappear this messages This message was added around 1.4.23 to let you know that you're violating the AGI protocol. Read up on the AGI protocol then check through your AGI file to see what you're missing. Usually it's not reading something that asterisk is sending back to your script, or not sending a response that asterisk is expecting. Everything else typically will still do what you want, but technically you're violating the protocol. -- Thanks, --Warren Selby http://www.selbytech.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 -- _ -- 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] WARNING message when play
When I use an eagi script when play a message appear a lot of warning messages, but it play very well I´m using Asterisk 1.4.32 dahdi-linux-2.3.0.1 chan_ss7-1.4.1 Any ideas?? -- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0) [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe -- _ -- 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] Dialplan behaviour
I understand the “ex-girlfriend” situation, in fact I want to do that, the problem is when I don´t put the last line and call from 92 or 91 this don´t work. I put the ex-girlfriend exception because without this, calls from 91 and 92 don´t match their extensions. On Mon, Mar 8, 2010 at 4:52 PM, Danny Nicholas da...@debsinc.com wrote: You made the troncal-sip context into a “ex-girlfriend” situation where only calls from extensions 91 and 92 would process. When you added the last line, that made it an open context with two egf exceptions. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Monday, March 08, 2010 1:46 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Dialplan behaviour I have this [TRONCAL-SIP] exten=225/91,1,Answer exten=225/91,2,Echo exten=225/91,3,Hangup exten=225/92,1,Answer exten=225/92,2,Playback(conf-invalid) exten=225/92,3,Hangup When I make a call CLI-- Recv IAM CIC=8ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=225,1,Answer Works ok -- _ -- 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] Dialplan behaviour
Yes, this work, thanks! On Wed, Mar 10, 2010 at 11:33 AM, Danny Nicholas da...@debsinc.com wrote: This may just be my opinion, but EG logic works best in an established call, like this [TRONCAL-SIP] exten = 225,1,answer exten=225/91,2,Answer exten=225/91,3,Echo exten=225/92,2,Answer exten=225/92,3,Playback(conf-invalid) exten=225,hangup This way, 225 is answered and hungup regardless of caller, and 91/92 get their specific handling. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Wednesday, March 10, 2010 8:14 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Dialplan behaviour I understand the “ex-girlfriend” situation, in fact I want to do that, the problem is when I don´t put the last line and call from 92 or 91 this don´t work. I put the ex-girlfriend exception because without this, calls from 91 and 92 don´t match their extensions. On Mon, Mar 8, 2010 at 4:52 PM, Danny Nicholas da...@debsinc.com wrote: You made the troncal-sip context into a “ex-girlfriend” situation where only calls from extensions 91 and 92 would process. When you added the last line, that made it an open context with two egf exceptions. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Monday, March 08, 2010 1:46 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Dialplan behaviour I have this [TRONCAL-SIP] exten=225/91,1,Answer exten=225/91,2,Echo exten=225/91,3,Hangup exten=225/92,1,Answer exten=225/92,2,Playback(conf-invalid) exten=225/92,3,Hangup When I make a call CLI-- Recv IAM CIC=8ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=225,1,Answer Works ok -- _ -- 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 -- _ -- 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] Dialplan behaviour
I have this [TRONCAL-SIP] exten=225/91,1,Answer exten=225/91,2,Echo exten=225/91,3,Hangup exten=225/92,1,Answer exten=225/92,2,Playback(conf-invalid) exten=225/92,3,Hangup When I make a call CLI-- Recv IAM CIC=8ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=225,1,Answer Works ok -- _ -- 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] Swift from eagi, problems with prosody rate
Hi, I'm trying to use Swift tts from eagi, my problem is when I send EXEC SWIFT *prosody rate*=\'.8\' Hello World\, this is a test\,/*prosody* |0|1 Would I use a scape character? Thanks -- _ -- 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] Swift from eagi, problems with prosody rate
I solve the problem, was a string formated problem. Thanks On Mon, Mar 1, 2010 at 10:05 AM, equis software equissoftw...@gmail.comwrote: Hi, I'm trying to use Swift tts from eagi, my problem is when I send EXEC SWIFT *prosody rate*=\'.8\' Hello World\, this is a test\,/* prosody*|0|1 Would I use a scape character? Thanks -- _ -- 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] chan_ss7 or libss7, which is more stable?
Hi, I´m trying to use SS/ in Asterisk. I'm thinking in chan_ss7 and libss7, and I want to know some other experience with this. Thanks! -- _ -- 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] Disable/enable CDR in dialplan
Hi, I need to eneble or disable cdr registration dynamicaly in my dialplan. Is there any way to do this? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Disable/enable CDR in dialplan
Thanks! On Mon, Sep 28, 2009 at 2:28 PM, Paul Dugas p...@dugasenterprises.comwrote: NoCDR On Mon, Sep 28, 2009 at 1:10 PM, equis software equissoftw...@gmail.com wrote: Hi, I need to eneble or disable cdr registration dynamicaly in my dialplan. Is there any way to do this? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Paul Dugas -- Computer Engineer -- Dugas Enterprises, LLC 522 Black Canyon Park, Canton GA 30114 USA p...@dugasenterprises.com -- +1.404.932.1355 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] Call no reject when receive 'PROGRESS with cause code 27 received' in zap channel
Hi, I have an asterisk connected with PRI (Zap channels). If I try to call a number, and recieve cause code 27 because the line 553192 is out of service, but the call continue...is it ok? Here the console messages -- Executing [...@troncal-pri-76:5] Dial(Zap/1-1, Zap/g1/553192) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/553192 -- PROGRESS with cause code 27 received -- Zap/2-1 is making progress passing it to Zap/1-1 -- Zap/2-1 is proceeding passing it to Zap/1-1 Why asterisk don´t stop the call? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Extensions patterns algorithm
Hi, has anybody some python code algorithm to parse an extension pattern? I have a number and need to know if match with some pattern. Thansk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Best free text to speech..
I try festival and espeak, festival is better then espeak but I need more natural sounds On Mon, Jun 8, 2009 at 4:57 PM, equis software equissoftw...@gmail.comwrote: I need to imlplement an IVR service where customers call and put a telephone number, then I reproduce the name and address. On Mon, Jun 8, 2009 at 3:57 PM, Michelle Dupuis supp...@ocg.ca wrote: Just out of curiosity, how are you planning to use it? (Reading email, etc?) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Monday, June 08, 2009 7:58 AM *To:* Asterisk Users List *Subject:* [asterisk-users] Best free text to speech.. Hi, i need to use a text to speech in my service. What do think is the best free project? Thanks ___ -- 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 ___ -- 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] Best free text to speech..
Hi, i need to use a text to speech in my service. What do think is the best free project? Thanks ___ -- 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] Best free text to speech..
Witch festival version are you talking about? I need spanish(argentinian) voice... On Mon, Jun 8, 2009 at 10:29 AM, David Backeberg dbackeb...@gmail.comwrote: On Mon, Jun 8, 2009 at 9:02 AM, Danny Nicholasda...@debsinc.com wrote: Cepstral and Festival are both “Free”. In Cepstral, you pay a license fee for the voice you use. In Festival, you tune the mechanical voice the way you want. So if you want “Truly free”, choose Festival. If you want a Human, “Professional” voice, Cepstral offers a reasonably priced product. I've never seen prices for Cepstral, but another commercial product is ATT Natural Voices. http://www.naturalvoices.att.com/ I will say that Festival sounds better now than it did a few years ago. I don't know the exact date, but at some point between when we chose to go with Natural Voices and now, Festival released a new algorithm that sounds dramatically better than it used to. I also really love the Scottish voices that come with Festival. They're improper for my usage, but if you have a business catering to customers where those accents would be appropriate it's a nice product. ___ -- 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 ___ -- 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] Best free text to speech..
I need to imlplement an IVR service where customers call and put a telephone number, then I reproduce the name and address. On Mon, Jun 8, 2009 at 3:57 PM, Michelle Dupuis supp...@ocg.ca wrote: Just out of curiosity, how are you planning to use it? (Reading email, etc?) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Monday, June 08, 2009 7:58 AM *To:* Asterisk Users List *Subject:* [asterisk-users] Best free text to speech.. Hi, i need to use a text to speech in my service. What do think is the best free project? Thanks ___ -- 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 ___ -- 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] Calling party category
Hi, in MFC-R2 signaling there is a value Calling party category signal (e.g., normal subscriber, high-priority subscriber, operator, coin-operated telephone) How can I get that information in my Asterisk?? Is there any similar value in SIP? Thanks ___ -- 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] Getting SIP field P-Asserted-Identity from EAGI
Hi, using EAGI variables like agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode I get a lot of data about a call, but I need to obtain P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get that? Or have you any solution?? Thanks!!! ___ -- 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] Getting SIP field P-Asserted-Identity from EAGI
Thanks to all, I'll try it!
On Thu, Feb 26, 2009 at 12:25 PM, Danny Nicholas da...@debsinc.com wrote:
Based on this page -
http://blog.herbertm.ca/2007/09/03/extracting-dids-from-the-sip-header
You could put this in your dialplan
[ext-did-custom]
exten = s,1,Set(ASSERT=${SIP_HEADER(Call-ID)})
exten = s,2,AGI(xxx.pl,${ASSERT})
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software
*Sent:* Thursday, February 26, 2009 7:37 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Getting SIP field P-Asserted-Identity from
EAGI
Hi, using EAGI variables like
agi_request
agi_channel
agi_language
agi_type
agi_uniqueid
agi_callerid
agi_dnid
agi_rdnis
agi_context
agi_extension
agi_priority
agi_enhanced
agi_accountcode
I get a lot of data about a call, but I need to obtain
P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get
that? Or have you any solution??
Thanks!!!
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[asterisk-users] Join empty queue property
I want the callers don't join in a queue when the agents are busy. I suposse it is easy but i can't get the solution for this. Can you suggest me something? Thanks. ___ -- 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] Join empty queue property
Because I want this behabior, if all agents are busy I redirect the call to an IVR to register the question. In fact there is a property in queue.conf ; This setting controls whether callers can join a queue with no members. There ; are three choices: ; ; yes- callers can join a queue with no members or only unavailable members ; no - callers cannot join a queue with no members ; strict - callers cannot join a queue with no members or only unavailable ; members ; ; joinempty = yes I try to use joinempty = strict But didn´t work... On Mon, Dec 29, 2008 at 2:31 PM, Carlos Chavez cur...@telecomabmex.comwrote: Then why use a queue? The purpose of a queue is exactly to keep people waiting while agents are all busy. The only way I can see something like what you want is to put a very low timeout (maybe 10 seconds) so if all your agents are busy then the caller will get dropped from the queue and continue with the dialplan where you can redirect them. On Mon, 2008-12-29 at 10:26 -0200, equis software wrote: I want the callers don't join in a queue when the agents are busy. I suposse it is easy but i can't get the solution for this. Can you suggest me something? Thanks. ___ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- 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 ___ -- 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] Priority between calls from different queues
Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Thanks ___ -- 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] Priority between calls from different queues
I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are ready to go to agent in different queues. Also, you can give priority to different callers within queue by setting QUEUE_PRIO variable before sending call to queue. You could try to describe why you need two queues and what should be rules to distribute calls - so we can help you with overall architecture. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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 ___ -- 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] Priority between calls from different queues
I understand, but I don´t have announcements. Regards On Fri, Nov 28, 2008 at 12:16 PM, Darrin Henshaw [EMAIL PROTECTED]wrote: One thing you also will run into is listed here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf. Here is the interesting part: Note that calls are not offered to queue members whilst the announcement is playing and it is possible for callers to slip ahead in the queue as a result. For example, call 1 arrives and is queued. Call 2 arrives ten seconds later and is queued. After twenty seconds, call 1 is played the periodic announce message. Exactly one second after call 1 starts hearing the message an agent becomes free. Since call 1 is tied up with announcements, call 2 is successfully offered to the agent. Call 1 remains on hold and yet a call which arrived later has been serviced. Basically you can see that if you have announcements played, that could cause your order of answered calls to be not what you expect. Cheers, [image: logo] Darrin Henshaw |* *IT Administrator | MCTS: Exchange 2007 | MCSE 2003 |LPIC Ignition Support Center* *|* *www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *equis software *Sent:* Friday, November 28, 2008 10:06 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Priority between calls from different queues I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are ready to go to agent in different queues. Also, you can give priority to different callers within queue by setting QUEUE_PRIO variable before sending call to queue. You could try to describe why you need two queues and what should be rules to distribute calls - so we can help you with overall architecture. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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 -- This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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 image001.jpg___ -- 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] Priority between calls from different queues
In both queues have the same wrapuptime, there´s not a problem... With weight property I can´t resolve my problem...I want to answer calls of both queues sorted by time, like a big FIFO or like if I had only one queue regards On Fri, Nov 28, 2008 at 12:31 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 4:16 PM, Darrin Henshaw [EMAIL PROTECTED] wrote: One thing you also will run into is listed here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf. Here is the interesting part: Note that calls are not offered to queue members whilst the announcement is playing and it is possible for callers to slip ahead in the queue as a result. For example, call 1 arrives and is queued. Call 2 arrives ten seconds later and is queued. After twenty seconds, call 1 is played the periodic announce message. Exactly one second after call 1 starts hearing the message an agent becomes free. Since call 1 is tied up with announcements, call 2 is successfully offered to the agent. Call 1 remains on hold and yet a call which arrived later has been serviced. Basically you can see that if you have announcements played, that could cause your order of answered calls to be not what you expect. With queues there are much more such situation than just this one ;) Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Friday, November 28, 2008 10:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Priority between calls from different queues I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. For in-between queues you can use weight. So, if queue1 has more weight than queue2, and agent1 is available (and is in both queues), he will receive call from queue1 (no matter how long other caller waits in queue2). Also, there's wrapuptime. It means - how many seconds agent should not receive call after completing previous queue call. So, if agent receives call from queue1 and it has wrapuptime 10 seconds, then he ends call, he might immediately receive call from queue2 - no matter that queue2 has lower weight or whatever settings. To overcome this, you have to enable shared_lastcall (available since 1.6.0). Regards, Atis Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are ready to go to agent in different queues. Also, you can give priority to different callers within queue by setting QUEUE_PRIO variable before sending call to queue. You could try to describe why you need two queues and what should be rules to distribute calls - so we can help you with overall architecture. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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 This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error
Re: [asterisk-users] Priority between calls from different queues
Thanks a lot! Your explanation was very clear. Thanks again. On Fri, Nov 28, 2008 at 2:14 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 4:51 PM, equis software [EMAIL PROTECTED] wrote: In both queues have the same wrapuptime, there´s not a problem... With weight property I can´t resolve my problem...I want to answer calls of both queues sorted by time, like a big FIFO or like if I had only one queue I'm afraid that it's not possible. There will be too much cases when one queue can choose to call agent ignoring another queue. What i meant with wrapuptime - even if it's the same (and you don't use shared_lastcall), second queue won't know that agent has just ended conversation - so it will send call to agent. I guess that there would be some more such race conditions for having free agent. If you really need FIFO, you would have much better luck with having one queue and then thinking how to customize it for different callers. Single instance of Queue is built like FIFO for calls (with bucket of agents). For example - wait time you can specify as argument to Queue(). As for different caller amount, you can assign them to groups and use GROUP_COUNT to determine how many they are in each group. If you need some more differentiation, just ask, and we'll try to give ideas. Oh, btw - you could also try to create one fake agent in queue1 and queue2 (with ringinuse=yes) and use Local channel to send those calls to queue-real where your agents reside. However, i'm not sure that this will work, as queue-real might answer channel, even if you set r option.. not sure is this a problem, but it could be complex :) Regards, Atis regards On Fri, Nov 28, 2008 at 12:31 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 4:16 PM, Darrin Henshaw [EMAIL PROTECTED] wrote: One thing you also will run into is listed here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf. Here is the interesting part: Note that calls are not offered to queue members whilst the announcement is playing and it is possible for callers to slip ahead in the queue as a result. For example, call 1 arrives and is queued. Call 2 arrives ten seconds later and is queued. After twenty seconds, call 1 is played the periodic announce message. Exactly one second after call 1 starts hearing the message an agent becomes free. Since call 1 is tied up with announcements, call 2 is successfully offered to the agent. Call 1 remains on hold and yet a call which arrived later has been serviced. Basically you can see that if you have announcements played, that could cause your order of answered calls to be not what you expect. With queues there are much more such situation than just this one ;) Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Friday, November 28, 2008 10:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Priority between calls from different queues I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. For in-between queues you can use weight. So, if queue1 has more weight than queue2, and agent1 is available (and is in both queues), he will receive call from queue1 (no matter how long other caller waits in queue2). Also, there's wrapuptime. It means - how many seconds agent should not receive call after completing previous queue call. So, if agent receives call from queue1 and it has wrapuptime 10 seconds, then he ends call, he might immediately receive call from queue2 - no matter that queue2 has lower weight or whatever settings. To overcome this, you have to enable shared_lastcall (available since 1.6.0). Regards, Atis Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot
Re: [asterisk-users] Queue App - Set monitoring dynamically
Thanks a lot Matt! On Sun, Nov 16, 2008 at 9:25 PM, Matt Riddell [EMAIL PROTECTED] wrote: On 15/11/2008 3:58 a.m., equis software wrote: I found this property in queue.conf ; Calls may be recorded using Asterisk's monitor resource ; This can be enabled from within the Queue application, starting recording ; when the call is actually picked up; thus, only successful calls are ; recorded, and you are not recording while people are listening to MOH. ; To enable monitoring, simply specify monitor-format; it will be disabled ; otherwise. ; ; monitor-format = gsm|wav|wav49 but I need to monitor queue calls depending on the time, not everytime. I have configured queue.conf in my postgres, but I don´t like to put an automatic taskin Linux to insert row, reload in Asterisk and then delete the row and reload Asterisk when time is over. Dynamically I can set in queue app to caller or called ca monitor their calls using - w — allow the called user to write the conversation to disk via Monitor. - W — allow the calling user to write the conversation to disk via Monitor. But I need to monitor all the calls. I do some operation in dialplan depending the time using GotoIfTime(21:00-09:00|sat|*|*?context1|96|1) Have Queue App any option to resolve this that I´m forgetting? You could create two queues with the same members. I.E. myqueue and myqueue-record. Then based on time, send the calls to one queue or the other. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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 ___ -- 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] Queue App - Set monitoring dynamically
I found this property in queue.conf ; Calls may be recorded using Asterisk's monitor resource ; This can be enabled from within the Queue application, starting recording ; when the call is actually picked up; thus, only successful calls are ; recorded, and you are not recording while people are listening to MOH. ; To enable monitoring, simply specify monitor-format; it will be disabled ; otherwise. ; ; monitor-format = gsm|wav|wav49 but I need to monitor queue calls depending on the time, not everytime. I have configured queue.conf in my postgres, but I don´t like to put an automatic taskin Linux to insert row, reload in Asterisk and then delete the row and reload Asterisk when time is over. Dynamically I can set in queue app to caller or called ca monitor their calls using - w — allow the called user to write the conversation to disk via Monitor. - W — allow the calling user to write the conversation to disk via Monitor. But I need to monitor all the calls. I do some operation in dialplan depending the time using GotoIfTime(21:00-09:00|sat|*|*?context1|96|1) Have Queue App any option to resolve this that I´m forgetting? Any Ideas? Thanks ___ -- 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] Dial issue
DTMF is working properly because in other part of my eagi script I use this DTMF. Thanks Doverli On Sat, Sep 27, 2008 at 3:16 AM, Max Alex [EMAIL PROTECTED] wrote: Hi, can you please confirm that DTMF is working properly or not? Thanks, Max Alex Voip Developer On Sat, Sep 27, 2008 at 12:24 AM, equis software [EMAIL PROTECTED]wrote: Hi, when I make a call I need that the caller can** hang up by dialing ** * (H option in Dial command), the call but it don´t work. Command EXEC DIAL Zap/g1/433391|20|H In CLI... -- AGI Script Executing Application: (DIAL) Options: (Zap/g1/433391|20|H) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/433391 -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/510093-082160f0 (--- At this moment I press * several times, but nothing happens Then I hung up the phone--) -- Hungup 'Zap/1-1' Any Ideas? ___ -- 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 ___ -- 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] Dial issue
Hi, when I make a call I need that the caller can** hang up by dialing ***(H option in Dial command), the call but it don´t work. Command EXEC DIAL Zap/g1/433391|20|H In CLI... -- AGI Script Executing Application: (DIAL) Options: (Zap/g1/433391|20|H) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/433391 -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/510093-082160f0 (--- At this moment I press * several times, but nothing happens Then I hung up the phone--) -- Hungup 'Zap/1-1' Any Ideas? ___ -- 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] I can´t hear the warning sound in Dial command
Hi! I´m using cmd Dial from an EAGI script My problem is that I cant hear the warning sound EAGI script: SET VARIABLE LIMIT_WARNING_FILE beep SET VARIABLE LIMIT_PLAYAUDIO_CALLEE yes SET VARIABLE LIMIT_PLAYAUDIO_CALLER yes EXEC DIAL Zap/g1/676354|20|HL(132000:3000:3000) CLI: -- AGI Script Executing Application: (DIAL) Options: (Zap/g1/676354|20|HL(132000:3000:3000)) -- Limit Data for this call: timelimit = 132000 play_warning = 3000 play_to_caller = yes play_to_callee = yes warning_freq = 3000 start_sound= (null) warning_sound = beep end_sound = (null) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/676354 -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/510093-08238830 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/1-1' -- AGI Script Prepago completed, returning 0 What I´m doing wrong?? Thanks!! Asterisk 1.4.19 built by root @ a1 on a i686 running Linux ___ -- 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] Click-to-talk (Java application)
Hi, I want to use any java open source solution to implement click-to talk in my web page connected to my Asterisk. I don´t need a callback solution. Regards. ___ -- 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] Call only for registered sip users...
What I need to configure in my * to permit make calls only registered sip users?? Thanks! ___ -- 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] Best Linux distribution to use in Asterisk server
Hi, I allways use Gentoo y my Asterisk servers and work well, but what do you think about to use Ubuntu or another distibution?? Thanks ___ -- 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] Click-to-talk (Java application)
Hi! I need to implement click-to-talk web application.(not click-to-call or callback) I try to use njiax, and iaxclient but I can´t made it work. Has anybody other solution?? ___ -- 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] Click-to-talk (Java application)
Thanks, I´m interested in non comercial solutions. On Mon, Apr 21, 2008 at 11:00 AM, Tim Panton [EMAIL PROTECTED] wrote: On 21 Apr 2008, at 14:31, equis software wrote: Hi! I need to implement click-to-talk web application.(not click-to-call or callback) I try to use njiax, and iaxclient but I can´t made it work. Has anybody other solution?? Yep. We can help on a commercial basis. Contact me off-list if you are interested. Tim. ___ -- 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 ___ -- 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] Best Click-to-call client
I think videoreps.net It´s not free. But, I discover that I really need is click-to-talk, excuse me. On Wed, Apr 16, 2008 at 5:05 PM, Bob G [EMAIL PROTECTED] wrote: Introducing Click-to-Call http://1ezphone.com/ Posted: 16 Apr 2008 9:55 AM PDT The 1EZphone browser softphone has created so much buzz in the media that a lot of individual users and companies who have a web-presence; Websites, Online Advertising, Blogs, Customer support etc have asked for a Click-to-Call service. The 1Ezphone web-based Click-to-Call service is based on our browser VoIP lite technology that allows users to make and receive phone calls from any browser without the need to download software. The Click-to Call API can be embedded on any Website, E-mail, and Online Advertisement when a user clicks your object they immediately call your salesperson or customer service representative telephone number and speak to your agent over their PC. Building a reliable Click-to-Call requires substantial amount of knowledge in VoIP, and a good backend infrastructure. The good news is that now it is easy add Click-to Call to any online service in just a few minutes with just a few lines of code using 1ezphone's. Since the release of our APIs, we got several requests from companies and developers who were interested in knowing in building their own Click-to-Call service directly to their SIP servers. You can have the button/widget running through the 1Ezphones servers without getting into the complex world of VoIP or any expensive setup or build a service to your own backend infrastructure. If you are interested in adding Click-to-Call for your customers or building your own Click-to Call system please contact 1ezphone at [EMAIL PROTECTED] - Original Message - From: BJ Weschke To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best Click-to-call client Date: Wed, 16 Apr 2008 11:37:40 -0400 equis software wrote: Hi, I need to make Click-to-Call web application to connect with an asterisk server. I´m using Java What solution recommend me? I did a spiel on this at Astricon last year. The slide deck is somewhere around for those interested, but now we also have some code to show for it. :-) Take a look at this developer branch at http://www.asterisk.org/node/48440 and then we've put some pieces together for the Java side of things using Ignite's Realtime API for messaging. http://svn.btwtech.com/svnview/coolvocals/trunk/cti-server/click2call/ http://svn.btwtech.com/svnview/coolvocals/trunk/cti-client/click2call/ Basically the idea here is that there's a servlet that honors requests into it (think AJAX Remote calls from the browser) and then turns around and puts that request into a jabber message that goes to a centralized Servlet that can proxy requests across multiple servers (scalability/LCR/etc) and that in turn launches an Originate call in to the AMI of the machine that was decided would receive the request. Once that hand off is done, the proxy machine that received and directed the original request is now out of the middle of things and jabber messages are sent directly back to the client to signal call progress of the click to call. Is it a shrink wrapped and ready to go package that's completely documented and involves no technical knowledge whatsoever for implementation? U.. no, but that might happen in the relatively near future. :-) What it IS though is solid working code (yes, it has been fully unit tested out and is functional) contributed back to the community so we can all start to make something with it if we so choose. If there's enough interest, I'd certainly entertain opening up a blog site and open up the branch of the Java code for community contributions as well in addition to doing a more detailed tutorial on usage of the code at the upcoming Astricon this year. BJ -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- 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 -- Want an e-mail address like mine? Get a *free e-mail *account today at www.mail.comhttp://www.mail.com/Product.aspx ! ___ -- 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 ___ -- 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] Best Click-to-call client
Hi, I need to make Click-to-Call web application to connect with an asterisk server. I´m using Java What solution recommend me? Thanks ___ -- 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] queue logging
You can´t see the ring event in the queue.log. You can get this event using the Manager connection. On Wed, Apr 9, 2008 at 5:01 AM, Arjan Kroon | Mobillion [EMAIL PROTECTED] wrote: Hi, I' using with asterisk a queue with tree members and round robin. When a caller enters this queue and it is connecting to one of the members, is there a possibility to see which member the caller is connected to? And is there a way to see in de application to see if the connection from the caller to one of the members was successful of not successful? I know you can see it in de queue. log. But I want to know if I can see it also in the hangup (h) in de application? Kind Regards ___ -- 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 image001.gif___ -- 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] Catch end of Eagi script when caller hung up...HELP ME PLEASE!!
Hi, I need to catch then end of an eagi script (python) when caller hungup because I want to generate my own CDR. I try this def run() signal.signal(signal.SIGHUP, self.logsignal) def logsignal(self,signum, frame): self.putCDR() but didn't work. Then try with several signals like: signal.signal(signal.SIGTERM, self.logsignal) signal.signal(signal.SIGTSTP, self.logsignal) signal.signal(signal.SIGPIPE, self.logsignal) ___ -- 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] Catch end of Eagi script when caller hung up...HELP ME PLEASE!!
Excuse me, but I thik this function is ok because I did this... def run() signal.signal(signal.SIGALRM, self.logsignal) signal.alarm(10) def logsignal(self,signum, frame): self.putCDR() And work very well, offcourse I need to putCDR() only with SIGHUP not with the SIGALRM. On Wed, Apr 9, 2008 at 10:37 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 09 April 2008 07:41:17 equis software wrote: Hi, I need to catch then end of an eagi script (python) when caller hungup because I want to generate my own CDR. I try this def run() signal.signal(signal.SIGHUP, self.logsignal) def logsignal(self,signum, frame): self.putCDR() but didn't work. Then try with several signals like: signal.signal(signal.SIGTERM, self.logsignal) signal.signal(signal.SIGTSTP, self.logsignal) signal.signal(signal.SIGPIPE, self.logsignal) If you read the Python documentation, you'll see that your signal handler must be a routine that takes 2 arguments, not the 3 that you're providing here. -- Tilghman ___ -- 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 ___ -- 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] Eagi
Hi! If the caller hungs up while an eagi script is running, I can´t regiter the cdr manually at the end of the script. I tryied to trap SIGHUP but it didn´t work. I want to register my own cdr into the script because I have a lot of data that I need to put in the cdr. The 'h' option or DeadAgi aren´t a solution for me. Thanks ___ -- 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] AGI-python script
Any new about this? Thanks On Thu, Mar 27, 2008 at 11:29 AM, equis software [EMAIL PROTECTED] wrote: I was trying to trap SIGHUP, but could be another signal because it didn't work. I'm doing this class MyScript(): def logsignal(self,signum, frame): self.putCDR() def run(self): signal.signal(signal.SIGHUP, self.logsignal) def putCDR(): put my cdr in my db. I was tryin trap other signals to test this and work well def run(self): signal.signal(signal.SIGALRM, self.logsignal) signal.alarm(3) Thanks a lot! On Wed, Mar 26, 2008 at 4:54 PM, Steve Edwards [EMAIL PROTECTED] wrote: On Wed, 26 Mar 2008, equis software wrote: Hi! I have some IVRs made in python. If the caller hangup before the end of the script I can´t register in my database the cdr. From your description, I'm not sure exactly what you are asking, but 1 of these should solve your problem. 1) Trap SIGHUP. 2) Use the h extension. 3) Use deadagi(). Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 ___ -- 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] AGI-python script
I was trying to trap SIGHUP, but could be another signal because it didn't work. I'm doing this class MyScript(): def logsignal(self,signum, frame): self.putCDR() def run(self): signal.signal(signal.SIGHUP, self.logsignal) def putCDR(): put my cdr in my db. I was tryin trap other signals to test this and work well def run(self): signal.signal(signal.SIGALRM, self.logsignal) signal.alarm(3) Thanks a lot! On Wed, Mar 26, 2008 at 4:54 PM, Steve Edwards [EMAIL PROTECTED] wrote: On Wed, 26 Mar 2008, equis software wrote: Hi! I have some IVRs made in python. If the caller hangup before the end of the script I can´t register in my database the cdr. From your description, I'm not sure exactly what you are asking, but 1 of these should solve your problem. 1) Trap SIGHUP. 2) Use the h extension. 3) Use deadagi(). Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 ___ -- 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] AGI-python script
Hi! I have some IVRs made in python. If the caller hangup before the end of the script I can´t register in my database the cdr. Any idea to do this? Thanks ___ -- 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] Best Java library to interact with Asterisk
Hi, I need to interact with my Asterisk and need a good Java class library. What do you think is the best? Thanks ___ -- 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] AgentLogin by console
Thanks Tzafir, but this functionality needs sombody answer the call. I need to do this automatically. On Jan 22, 2008 4:10 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Jan 22, 2008 at 04:04:29PM -0200, equis software wrote: Hi! Is there any way to login an agent by console command? I want to login an agent doing this system call. asterisk -rx 'AgentCallbackLogin 304 [EMAIL PROTECTED]' Any ideas, thanks. Something of the sort of: originate SIP/peer-to-login application AgentCallbackLogin -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AgentLogin by console
Hi! Is there any way to login an agent by console command? I want to login an agent doing this system call. asterisk -rx 'AgentCallbackLogin 304 [EMAIL PROTECTED]' Any ideas, thanks. ___ -- 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] Enable/Disable Sip without registration
I try to configure that only registered sips can make calls. How can I do that? I was looking in sip.conf but I didn´t found wath opition configure this functionality. ___ --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] Enable/Disable Sip without registration
Sorry I don´t understand. Could you explain me with more detailed? Thanks! On Dec 12, 2007 10:35 AM, ram [EMAIL PROTECTED] wrote: On Dec 12, 2007 6:31 PM, equis software [EMAIL PROTECTED] wrote: I try to configure that only registered sips can make calls. How can I do that? I was looking in sip.conf but I didn´t found wath opition configure this functionality. Create a users in sip.conf with context so that user will register with asterisk to make calls ram ___ --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 ___ --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] cdr_pgsql error in 1.4.15
But with older versions of Asterisk It didn´t happend. Which libraries? Postgres Libraries or Asterisk librarries? On Dec 1, 2007 10:10 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Saturday 01 December 2007 09:43:41 equis software wrote: In Asterisk 1.4.15 if I try to configure cdr_pgsql.conf , asterisk crash with this message asterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_pgsql.so: undefined symbol: PQescapeStringConn Is this a knowed error? This sounds like version skew -- like you have headers from a later version of Postgres, but an earlier version of the libraries. -- Tilghman ___ --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 ___ --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 1.4.15 crash without generating core file
Is not a configuration problem because if I run an older version of Asterisk (ie 1.4.12) it generates this core in /tmp With 1.4.15 I did # updatedb and then #locate core* and I didn´t found. I found why Asterisk 1.4.15 crash, is because of this error: asterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_pgsql.so: undefined symbol: PQescapeStringConn I comment cdr_pgsql.conf and it doesn´t crash any more.(Off course I haven´t cdr in postgres) On Dec 3, 2007 7:45 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Dec 02, 2007 at 01:06:50PM +1300, Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 equis software wrote: Hi, I'm testing Asterisk 1.4.15 with the -g option. When it crash didn´t generate core file in the /tmp folder. What is happening?? Check the directory you were in when you ran Asterisk. Or alternatively: echo '/tmp/core.%e.%t' /proc/sys/kernel/core_pattern This will mean that all core files (for anything that generates core file, not only asterisk) will be created under /tmp with the file name 'core.name_of_executable.time_of_crash' (time: seconds since epoch). To make this change permanent (if you actually think it is wise), use /etc/sysctl.conf . -- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue App - crash (1.4.15)
This is the core trace
(gdb) bt
#0 0xb7e5a231 in strcasecmp () from /lib/libc.so.6
#1 0xb7ce0a3f in local_ast_moh_start (chan=0x82496a8, mclass=0xb720f828
default, interpclass=0x0)
at res_musiconhold.c:646
#2 0x08083695 in ast_moh_start (chan=0x64, mclass=0x64 Address 0x64 out of
bounds,
interpclass=0x88 Address 0x88 out of bounds) at channel.c:4614
#3 0xb741818b in queue_exec (chan=0x82496a8, data=0xb720f828) at
app_queue.c:3601
#4 0x080c638d in pbx_extension_helper (c=0x82496a8, con=0x64,
context=0x8249828 my-queue, exten=0x8249878 80,
priority=7, label=0x0, callerid=0x821e5b0 562390, action=136482640) at
pbx.c:532
#5 0x080c7041 in __ast_pbx_run (c=0x82496a8) at pbx.c:2314
#6 0x080c7fd1 in pbx_thread (data=0x64) at pbx.c:2631
#7 0x080f7e99 in dummy_start (data=0x64) at utils.c:843
#8 0xb7f7f13d in pthread_start_thread () from /lib/libpthread.so.0
#9 0xb7ea81ba in clone () from /lib/libc.so.6
(gdb) bt full
#0 0xb7e5a231 in strcasecmp () from /lib/libc.so.6
No symbol table info available.
#1 0xb7ce0a3f in local_ast_moh_start (chan=0x82496a8, mclass=0xb720f828
default, interpclass=0x0)
at res_musiconhold.c:646
mohclass = (struct mohclass *) 0x88
#2 0x08083695 in ast_moh_start (chan=0x64, mclass=0x64 Address 0x64 out of
bounds,
interpclass=0x88 Address 0x88 out of bounds) at channel.c:4614
No locals.
#3 0xb741818b in queue_exec (chan=0x82496a8, data=0xb720f828) at
app_queue.c:3601
makeannouncement = 0
res = 136360144
ringing = 0
lu = (struct ast_module_user *) 0x824b590
user_priority = 0x820b0d0 1196740700.14
max_penalty_str = 0x820b0d0 1196740700.14
prio = 0
max_penalty = 0
reason = QUEUE_UNKNOWN
tries = 0
noption = 0
args = {argc = 5, argv = 0xb720f9a8, queuename = 0xb720f7b0
my-queue, options = 0xb720f7bb t,
url = 0xb720f7bd , announceoverride = 0xb720f7be , queuetimeoutstr =
0xb720f7bf 300, agi = 0x0}
qe = {parent = 0x8229de0, moh = default, '\0' repeats 72 times,
announce = '\0' repeats 79 times,
context = '\0' repeats 79 times, digits = '\0' repeats 79 times,
valid_digits = 0, pos = 1, prio = 0,
last_pos_said = 0, last_periodic_announce_time = 1196740706,
last_periodic_announce_sound = 0, last_pos = 0, opos = 1,
handled = 0, max_penalty = 0, start = 1196740706, expire = 1196741006,
chan = 0x82496a8, next = 0x0}
#4 0x080c638d in pbx_extension_helper (c=0x82496a8, con=0x64,
context=0x8249828 my-queue, exten=0x8249878 80,
priority=7, label=0x0, callerid=0x821e5b0 562390, action=136482640) at
pbx.c:532
e = (struct ast_exten *) 0x82207a0
res = 0
q = {incstack = {0x0 repeats 128 times}, stacklen = 0, status = 5,
swo = 0x0, data = 0x0,
foundcontext = 0x8249828 my-queue}
passdata = my-queue|t|||300, '\0' repeats 8173 times
matching_action = -1222567212
#5 0x080c7041 in __ast_pbx_run (c=0x82496a8) at pbx.c:2314
---Type return to continue, or q return to quit---
dst_exten = '\0' repeats 124 times,
1Vå·\000\000\000\000\000\000\000\000\2167ø·, '\0' repeats 12 times,
jð· jð·D!·ô\177ø· jð·\f\000\000\000D!·t2ø·0jð·ôOð·|!·¥Rå·
jð·ô\177ø·À\022\\bH\201\024\b|!·\215\001ø·\020\000\000\000\f\000\000\000ðh\bÈç\031\b\230\232$\b\000\000\000\000!4ø·Ñ
\006\b
pos = 0
digit = 0
found = 1
res = 0
error = 0
#6 0x080c7fd1 in pbx_thread (data=0x64) at pbx.c:2631
No locals.
#7 0x080f7e99 in dummy_start (data=0x64) at utils.c:843
_buffer = {__routine = 0x8067f40 ast_unregister_thread, __arg =
0xa0019, __canceltype = -1222557972,
__prev = 0x0}
ret = (void *) 0x8249a98
a = {start_routine = 0x80c7fc0 pbx_thread, data = 0x82496a8,
name = 0x8249a98 pbx_thread, ' ' repeats 11 times, started at [ 2655]
pbx.c ast_pbx_start()}
#8 0xb7f7f13d in pthread_start_thread () from /lib/libpthread.so.0
No symbol table info available.
#9 0xb7ea81ba in clone () from /lib/libc.so.6
No symbol table info available.
(gdb) thread apply all bt
Thread 29 (process 7281):
#0 0xb7f8523b in read () from /lib/libpthread.so.0
#1 0x in ?? ()
Thread 28 (process 7469):
#0 0xb7e9f90a in poll () from /lib/libc.so.6
#1 0xb7f7ec60 in __pthread_manager () from /lib/libpthread.so.0
#2 0xb7ea81ba in clone () from /lib/libc.so.6
Thread 27 (process 7471):
#0 0xb7e9f90a in poll () from /lib/libc.so.6
#1 0x0806f26f in listener (unused=0x0) at asterisk.c:980
#2 0x080f7e99 in dummy_start (data=0x) at utils.c:843
#3 0xb7f7f13d in pthread_start_thread () from /lib/libpthread.so.0
#4 0xb7ea81ba in clone () from /lib/libc.so.6
Thread 26 (process 7542):
#0 0xb7e9f90a in poll () from /lib/libc.so.6
#1 0x080b600f in accept_thread (ignore=0x0) at manager.c:2340
#2 0x080f7e99 in dummy_start (data=0x1388) at utils.c:843
#3 0xb7f7f13d in pthread_start_thread () from /lib/libpthread.so.0
#4 0xb7ea81ba in clone () from /lib/libc.so.6
Thread 25 (process 7557):
#0 0xb7f81fa4
[asterisk-users] Queue App - (1.4.15) free agents with callers waiting
Hi! In 1.4.15 I have 3 agents, while 4 calls are waiting, 2 agents are ringing and the third agent don´t ring. I´m using autofill=true ___ --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] cdr_pgsql error in 1.4.15
In Asterisk 1.4.15 if I try to configure cdr_pgsql.conf , asterisk crash with this message asterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_pgsql.so: undefined symbol: PQescapeStringConn Is this a knowed error? ___ --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] Problems with Asterisk 1.4.14 and Queue app
It´s very strange, when Asterisk 1.4.15 crash don´t make a core file...
I´m sure it´s running with -g option!!
On Nov 30, 2007 11:02 AM, equis software [EMAIL PROTECTED] wrote:
Hi, Jared. I'm going to test in 1.4.15 and then I'll tell you what
happend.
Thanks
On Nov 29, 2007 3:35 PM, equis software [EMAIL PROTECTED] wrote:
You are right!
Here there is the backtrace
(gdb) bt
#0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
#1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
default, interpclass=0x0) at res_musiconhold.c:646
#2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 Address 0x64
out of bounds, interpclass=0x2e0 Address 0x2e0 out of bounds) at
channel.c:4609
#3 0xb73eb17b in queue_exec (chan=0x821cec0, data=0xb71a3788) at
app_queue.c:3600
#4 0x080c5d6b in pbx_extension_helper (c=0x821cec0, con=0x64,
context=0x821d040 my-queue, exten=0x821d090 80, priority=8, label=0x0,
callerid=0x819dc90 490814, action=E_MATCHMORE) at pbx.c:532
#5 0x080c6a31 in __ast_pbx_run (c=0x821cec0) at pbx.c:2293
#6 0x080c78f1 in pbx_thread (data=0x64) at pbx.c:2608
#7 0x080f7759 in dummy_start (data=0x64) at utils.c:843
#8 0xb7f1513d in pthread_start_thread () from /lib/libpthread.so.0
#9 0xb7e3e1ba in clone () from /lib/libc.so.6
(gdb) bt full
#0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
No symbol table info available.
#1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
default, interpclass=0x0) at res_musiconhold.c:646
mohclass = (struct mohclass *) 0x2e0
#2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 Address 0x64
out of bounds, interpclass=0x2e0 Address 0x2e0 out of bounds) at
channel.c:4609
No locals.
#3 0xb73eb17b in queue_exec (chan=0x821cec0, data=0xb71a3788) at
app_queue.c:3600
makeannouncement = 0
res = 136429000
ringing = 0
lu = (struct ast_module_user *) 0x8250230
user_priority = 0x821bdc8 1196248345.116
max_penalty_str = 0x821bdc8 1196248345.116
prio = 0
max_penalty = 0
reason = QUEUE_UNKNOWN
tries = 0
noption = 0
args = {argc = 5, argv = 0xb71a3908, queuename = 0xb71a3710
my-queue, options = 0xb71a371b t, url = 0xb71a371d ,
announceoverride = 0xb71a371e , queuetimeoutstr = 0xb71a371f 300,
agi = 0x0}
qe = {parent = 0x8227198, moh = default, '\0' repeats 72
times, announce = '\0' repeats 79 times, context = '\0' repeats 79
times,
digits = '\0' repeats 79 times, valid_digits = 0, pos = 1, prio = 0,
last_pos_said = 0, last_periodic_announce_time = 1196248351,
last_periodic_announce_sound = 0, last_pos = 0, opos = 1, handled = 0,
max_penalty = 0, start = 1196248351, expire = 1196248651, chan = 0x821cec0,
next = 0x0}
#4 0x080c5d6b in pbx_extension_helper (c=0x821cec0, con=0x64,
context=0x821d040 my-queue, exten=0x821d090 80, priority=8, label=0x0,
callerid=0x819dc90 490814, action=E_MATCHMORE) at pbx.c:532
e = (struct ast_exten *) 0x8255fc0
res = 0
q = {incstack = {0x0 repeats 128 times}, stacklen = 0, status
= 5, swo = 0x0, data = 0x0, foundcontext = 0x821d040 my-queue}
passdata = my-queue|t|||300, '\0' repeats 8173 times
matching_action = 136488696
#5 0x080c6a31 in __ast_pbx_run (c=0x821cec0) at pbx.c:2293
dst_exten = '\0' repeats 124 times,
1¶Þ·\000\000\000\000\000\000\000\000\216\227ñ·, '\0' repeats 12 times,
Êé· Êé·D\236\032·ôßñ·
Êé·\f\000\000\000D\236\032·t\222ñ·0Êé·ô¯é·|\236\032·¥²Þ·
Êé·ôßñ·\200\026%\b¨x\024\b|\236\032·\215añ·\020\000\000\000\f\000\000\000pÍ%\b°Ò!\bÐL\\b\000\000\000\000!\224ñ·\201
\006\b
pos = 0
digit = 0
found = 1
res = 0
error = 0
#6 0x080c78f1 in pbx_thread (data=0x64) at pbx.c:2608
No locals.
#7 0x080f7759 in dummy_start (data=0x64) at utils.c:843
_buffer = {__routine = 0x8067ef0 ast_unregister_thread, __arg
= 0x1b8019, __canceltype = -1222992148, __prev = 0x0}
ret = (void *) 0x8224cd0
---Type return to continue, or q return to quit---
a = {start_routine = 0x80c78e0 pbx_thread, data = 0x821cec0,
name = 0x8224cd0 pbx_thread, ' ' repeats 11 times, started at [
2632] pbx.c ast_pbx_start()}
#8 0xb7f1513d in pthread_start_thread () from /lib/libpthread.so.0
No symbol table info available.
#9 0xb7e3e1ba in clone () from /lib/libc.so.6
No symbol table info available.
Thanks
On Nov 29, 2007 3:04 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Thu, 2007-11-29 at 14:28 -0300, equis software wrote:
I have problems with 1.4.14, it crash every few minutes.
The same configuration and machine in Asterisk 1.4.6 it doesn´t
happend.
Are you able to get a good backtrace from the core file generated by
the
crash? Without more details, it's going
Re: [asterisk-users] Problems with Asterisk 1.4.14 and Queue app
Hi, Jared. I'm going to test in 1.4.15 and then I'll tell you what happend.
Thanks
On Nov 29, 2007 3:35 PM, equis software [EMAIL PROTECTED] wrote:
You are right!
Here there is the backtrace
(gdb) bt
#0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
#1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
default, interpclass=0x0) at res_musiconhold.c:646
#2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 Address 0x64 out
of bounds, interpclass=0x2e0 Address 0x2e0 out of bounds) at channel.c
:4609
#3 0xb73eb17b in queue_exec (chan=0x821cec0, data=0xb71a3788) at
app_queue.c:3600
#4 0x080c5d6b in pbx_extension_helper (c=0x821cec0, con=0x64,
context=0x821d040 my-queue, exten=0x821d090 80, priority=8, label=0x0,
callerid=0x819dc90 490814, action=E_MATCHMORE) at pbx.c:532
#5 0x080c6a31 in __ast_pbx_run (c=0x821cec0) at pbx.c:2293
#6 0x080c78f1 in pbx_thread (data=0x64) at pbx.c:2608
#7 0x080f7759 in dummy_start (data=0x64) at utils.c:843
#8 0xb7f1513d in pthread_start_thread () from /lib/libpthread.so.0
#9 0xb7e3e1ba in clone () from /lib/libc.so.6
(gdb) bt full
#0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
No symbol table info available.
#1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
default, interpclass=0x0) at res_musiconhold.c:646
mohclass = (struct mohclass *) 0x2e0
#2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 Address 0x64 out
of bounds, interpclass=0x2e0 Address 0x2e0 out of bounds) at channel.c
:4609
No locals.
#3 0xb73eb17b in queue_exec (chan=0x821cec0, data=0xb71a3788) at
app_queue.c:3600
makeannouncement = 0
res = 136429000
ringing = 0
lu = (struct ast_module_user *) 0x8250230
user_priority = 0x821bdc8 1196248345.116
max_penalty_str = 0x821bdc8 1196248345.116
prio = 0
max_penalty = 0
reason = QUEUE_UNKNOWN
tries = 0
noption = 0
args = {argc = 5, argv = 0xb71a3908, queuename = 0xb71a3710
my-queue, options = 0xb71a371b t, url = 0xb71a371d ,
announceoverride = 0xb71a371e , queuetimeoutstr = 0xb71a371f 300,
agi = 0x0}
qe = {parent = 0x8227198, moh = default, '\0' repeats 72
times, announce = '\0' repeats 79 times, context = '\0' repeats 79
times,
digits = '\0' repeats 79 times, valid_digits = 0, pos = 1, prio = 0,
last_pos_said = 0, last_periodic_announce_time = 1196248351,
last_periodic_announce_sound = 0, last_pos = 0, opos = 1, handled = 0,
max_penalty = 0, start = 1196248351, expire = 1196248651, chan = 0x821cec0,
next = 0x0}
#4 0x080c5d6b in pbx_extension_helper (c=0x821cec0, con=0x64,
context=0x821d040 my-queue, exten=0x821d090 80, priority=8, label=0x0,
callerid=0x819dc90 490814, action=E_MATCHMORE) at pbx.c:532
e = (struct ast_exten *) 0x8255fc0
res = 0
q = {incstack = {0x0 repeats 128 times}, stacklen = 0, status =
5, swo = 0x0, data = 0x0, foundcontext = 0x821d040 my-queue}
passdata = my-queue|t|||300, '\0' repeats 8173 times
matching_action = 136488696
#5 0x080c6a31 in __ast_pbx_run (c=0x821cec0) at pbx.c:2293
dst_exten = '\0' repeats 124 times,
1¶Þ·\000\000\000\000\000\000\000\000\216\227ñ·, '\0' repeats 12 times,
Êé· Êé·D\236\032·ôßñ·
Êé·\f\000\000\000D\236\032·t\222ñ·0Êé·ô¯é·|\236\032·¥²Þ·
Êé·ôßñ·\200\026%\b¨x\024\b|\236\032·\215añ·\020\000\000\000\f\000\000\000pÍ%\b°Ò!\bÐL\\b\000\000\000\000!\224ñ·\201
\006\b
pos = 0
digit = 0
found = 1
res = 0
error = 0
#6 0x080c78f1 in pbx_thread (data=0x64) at pbx.c:2608
No locals.
#7 0x080f7759 in dummy_start (data=0x64) at utils.c:843
_buffer = {__routine = 0x8067ef0 ast_unregister_thread, __arg =
0x1b8019, __canceltype = -1222992148, __prev = 0x0}
ret = (void *) 0x8224cd0
---Type return to continue, or q return to quit---
a = {start_routine = 0x80c78e0 pbx_thread, data = 0x821cec0,
name = 0x8224cd0 pbx_thread, ' ' repeats 11 times, started at [
2632] pbx.c ast_pbx_start()}
#8 0xb7f1513d in pthread_start_thread () from /lib/libpthread.so.0
No symbol table info available.
#9 0xb7e3e1ba in clone () from /lib/libc.so.6
No symbol table info available.
Thanks
On Nov 29, 2007 3:04 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Thu, 2007-11-29 at 14:28 -0300, equis software wrote:
I have problems with 1.4.14, it crash every few minutes.
The same configuration and machine in Asterisk 1.4.6 it doesn´t
happend.
Are you able to get a good backtrace from the core file generated by the
crash? Without more details, it's going to be close to impossible for
the Asterisk developers to guess at why it's crashing for you.
There's some good information at
http://www.asterisk.org/doxygen/1.4/AstDebug.html on how to generate the
backtrace and attach it to a bug in the bug tracker.
--
Jared Smith
Community Relations
[asterisk-users] Asterisk 1.4.15 crash without generating core file
Hi, I'm testing Asterisk 1.4.15 with the -g option. When it crash didn´t generate core file in the /tmp folder. What is happening?? ___ --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] Problems with Asterisk 1.4.14 and Queue app
I have problems with 1.4.14, it crash every few minutes. The same configuration and machine in Asterisk 1.4.6 it doesn´t happend. Is there anybody with similiar problems? Thanks ___ --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] Problems with Asterisk 1.4.14 and Queue app
You are right!
Here there is the backtrace
(gdb) bt
#0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
#1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
default, interpclass=0x0) at res_musiconhold.c:646
#2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 Address 0x64 out of
bounds, interpclass=0x2e0 Address 0x2e0 out of bounds) at channel.c:4609
#3 0xb73eb17b in queue_exec (chan=0x821cec0, data=0xb71a3788) at
app_queue.c:3600
#4 0x080c5d6b in pbx_extension_helper (c=0x821cec0, con=0x64,
context=0x821d040 my-queue, exten=0x821d090 80, priority=8, label=0x0,
callerid=0x819dc90 490814, action=E_MATCHMORE) at pbx.c:532
#5 0x080c6a31 in __ast_pbx_run (c=0x821cec0) at pbx.c:2293
#6 0x080c78f1 in pbx_thread (data=0x64) at pbx.c:2608
#7 0x080f7759 in dummy_start (data=0x64) at utils.c:843
#8 0xb7f1513d in pthread_start_thread () from /lib/libpthread.so.0
#9 0xb7e3e1ba in clone () from /lib/libc.so.6
(gdb) bt full
#0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
No symbol table info available.
#1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
default, interpclass=0x0) at res_musiconhold.c:646
mohclass = (struct mohclass *) 0x2e0
#2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 Address 0x64 out of
bounds, interpclass=0x2e0 Address 0x2e0 out of bounds) at channel.c:4609
No locals.
#3 0xb73eb17b in queue_exec (chan=0x821cec0, data=0xb71a3788) at
app_queue.c:3600
makeannouncement = 0
res = 136429000
ringing = 0
lu = (struct ast_module_user *) 0x8250230
user_priority = 0x821bdc8 1196248345.116
max_penalty_str = 0x821bdc8 1196248345.116
prio = 0
max_penalty = 0
reason = QUEUE_UNKNOWN
tries = 0
noption = 0
args = {argc = 5, argv = 0xb71a3908, queuename = 0xb71a3710
my-queue, options = 0xb71a371b t, url = 0xb71a371d ,
announceoverride = 0xb71a371e , queuetimeoutstr = 0xb71a371f 300, agi
= 0x0}
qe = {parent = 0x8227198, moh = default, '\0' repeats 72 times,
announce = '\0' repeats 79 times, context = '\0' repeats 79 times,
digits = '\0' repeats 79 times, valid_digits = 0, pos = 1, prio = 0,
last_pos_said = 0, last_periodic_announce_time = 1196248351,
last_periodic_announce_sound = 0, last_pos = 0, opos = 1, handled = 0,
max_penalty = 0, start = 1196248351, expire = 1196248651, chan = 0x821cec0,
next = 0x0}
#4 0x080c5d6b in pbx_extension_helper (c=0x821cec0, con=0x64,
context=0x821d040 my-queue, exten=0x821d090 80, priority=8, label=0x0,
callerid=0x819dc90 490814, action=E_MATCHMORE) at pbx.c:532
e = (struct ast_exten *) 0x8255fc0
res = 0
q = {incstack = {0x0 repeats 128 times}, stacklen = 0, status = 5,
swo = 0x0, data = 0x0, foundcontext = 0x821d040 my-queue}
passdata = my-queue|t|||300, '\0' repeats 8173 times
matching_action = 136488696
#5 0x080c6a31 in __ast_pbx_run (c=0x821cec0) at pbx.c:2293
dst_exten = '\0' repeats 124 times,
1¶Þ·\000\000\000\000\000\000\000\000\216\227ñ·, '\0' repeats 12 times,
Êé· Êé·D\236\032·ôßñ·
Êé·\f\000\000\000D\236\032·t\222ñ·0Êé·ô¯é·|\236\032·¥²Þ·
Êé·ôßñ·\200\026%\b¨x\024\b|\236\032·\215añ·\020\000\000\000\f\000\000\000pÍ%\b°Ò!\bÐL\\b\000\000\000\000!\224ñ·\201
\006\b
pos = 0
digit = 0
found = 1
res = 0
error = 0
#6 0x080c78f1 in pbx_thread (data=0x64) at pbx.c:2608
No locals.
#7 0x080f7759 in dummy_start (data=0x64) at utils.c:843
_buffer = {__routine = 0x8067ef0 ast_unregister_thread, __arg =
0x1b8019, __canceltype = -1222992148, __prev = 0x0}
ret = (void *) 0x8224cd0
---Type return to continue, or q return to quit---
a = {start_routine = 0x80c78e0 pbx_thread, data = 0x821cec0,
name = 0x8224cd0 pbx_thread, ' ' repeats 11 times, started at [ 2632]
pbx.c ast_pbx_start()}
#8 0xb7f1513d in pthread_start_thread () from /lib/libpthread.so.0
No symbol table info available.
#9 0xb7e3e1ba in clone () from /lib/libc.so.6
No symbol table info available.
Thanks
On Nov 29, 2007 3:04 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Thu, 2007-11-29 at 14:28 -0300, equis software wrote:
I have problems with 1.4.14, it crash every few minutes.
The same configuration and machine in Asterisk 1.4.6 it doesn´t
happend.
Are you able to get a good backtrace from the core file generated by the
crash? Without more details, it's going to be close to impossible for
the Asterisk developers to guess at why it's crashing for you.
There's some good information at
http://www.asterisk.org/doxygen/1.4/AstDebug.html on how to generate the
backtrace and attach it to a bug in the bug tracker.
--
Jared Smith
Community Relations Manager
Digium, Inc.
___
--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] Best text-to-speech
Hi! I need to use text to speech, what is the best application? Thanks! ___ --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] Zaptel 1.4.4 compiling problems
Hi! I have this error compiling Zaptel 1.4.4 make: *** No rule to make target `xpp/xpp_usb.ko', needed by `install-modules'. Stop. The Zaptel 1.2.5 compile ok. Any ideas?? ___ --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
