Re: [asterisk-users] Critical Asterisk Outage - Installing g729 crashed asterisk.
Buggy module. I mis-read. and downloaded the one for pre 1.8.4 On Thu, Oct 18, 2012 at 12:53 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Thu, Oct 18, 2012 at 1:49 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Thu, Oct 18, 2012 at 1:35 AM, Jared Baxley jared.bax...@gmail.comwrote: I was following Digium's instructions to the letter to install g729. but upon telling asterisk to load the module, the system hung. after a few minutes later a CTRL-C and attempted to run the command again. Same result. any g729 show command returns nothing... no error no results. Reboot the server and asterisk will not process calls. Freepbx shows the following. [2012-Oct-18 01:21:04] [INFO] (bin/retrieve_conf:109) - found language dir fr for directory, not installed on system, skipping [2012-Oct-18 01:21:07] [FATAL] (libraries/utility.functions.php:429) - retreive_conf failed to get engine information and cannot configure up a softwitch with out it. Error: ERROR-UNABLE-TO-PARSE [2012-Oct-18 01:21:07] [CRITICAL] (admin/functions.inc.php:366) - [NOTIFICATION]-[freepbx]-[RCONFFAIL] - retrieve_conf failed, config not applied This seems to indicate that the g729 module is working [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: G.729A transcoding module version 1.8.0_3.1.5, Copyright (C) 1999-2009 Digium, Inc. [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This module is supplied under a commercial license granted by Digium, Inc. [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Please see the full license text supplied by the accompanying [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: register utility, or ask for a copy from Digium. [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This product includes software developed by the OpenSSL Project [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Copyright (C) 1998-2006 The OpenSSL Project [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered action G729LicenseStatus [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered action G729LicenseList [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Host-ID: 60:b6:d3:a0:c9:be:6e:46:74:41:07:b3:8e:76:59:b1:ba:5c:59:05 [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found license 'G729-XXX' providing 40 channels [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found total of 40 G.729 licenses How do i roll this back? Just delete codec_g729a.so ? You can do a noload in modules.conf. This doesn't appear to be the problem though. It may be. Did you try saving a change in FreePBX and applying it? It seems more like a FreePBX config error that should be overwritten by FreePBX database to flat files. Thanks, Steve Totaro See here http://www.freepbx.org/forum/freepbx/users/apply-configuration-changes-errors-with-failed-to-get-engine-info-retreive-conf Very similar problem with FreePBX, your G729 looks fine. Check permissions and ownership of any files you changed. 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] Unable to load users.conf
On Thu, 18 Oct 2012, Rizha Yuherdianto wrote: 3) im root Glad to meet you. :D If you meant the user running Asterisk is root, this is a less than optimal situation that can lead to really big problems. Why? Steve please explain. Well, if an attacker manages to inject some code and Asterisk is running as root, poof goes your system or you get an astronomical bill from your trunk provider. Likewise with file permissions. Suppose you're trying to get something working and you suspect it's a permissions issue so you chmod a bunch of stuff to 777. Then suppose a local user with a grudge does something like this: echo '#exec rm --farce --recursive /*'\ /etc/asterisk/extensions-local.conf (or whatever your package names one of it's 'include' files.) The next time Asterisk reloads the dialplan, poof. -- 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
Re: [asterisk-users] Critical Asterisk Outage - Installing g729 crashed asterisk.
Resolved... Thanks to Digium for pointing out I installed a buggy module... My tired eyes though 1.8.14 was 1.8.1.4 ... Good night all. On Thu, Oct 18, 2012 at 12:35 AM, Jared Baxley jared.bax...@gmail.comwrote: I was following Digium's instructions to the letter to install g729. but upon telling asterisk to load the module, the system hung. after a few minutes later a CTRL-C and attempted to run the command again. Same result. any g729 show command returns nothing... no error no results. Reboot the server and asterisk will not process calls. Freepbx shows the following. [2012-Oct-18 01:21:04] [INFO] (bin/retrieve_conf:109) - found language dir fr for directory, not installed on system, skipping [2012-Oct-18 01:21:07] [FATAL] (libraries/utility.functions.php:429) - retreive_conf failed to get engine information and cannot configure up a softwitch with out it. Error: ERROR-UNABLE-TO-PARSE [2012-Oct-18 01:21:07] [CRITICAL] (admin/functions.inc.php:366) - [NOTIFICATION]-[freepbx]-[RCONFFAIL] - retrieve_conf failed, config not applied This seems to indicate that the g729 module is working [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: G.729A transcoding module version 1.8.0_3.1.5, Copyright (C) 1999-2009 Digium, Inc. [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This module is supplied under a commercial license granted by Digium, Inc. [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Please see the full license text supplied by the accompanying [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: register utility, or ask for a copy from Digium. [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This product includes software developed by the OpenSSL Project [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Copyright (C) 1998-2006 The OpenSSL Project [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered action G729LicenseStatus [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered action G729LicenseList [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Host-ID: 60:b6:d3:a0:c9:be:6e:46:74:41:07:b3:8e:76:59:b1:ba:5c:59:05 [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found license 'G729-XXX' providing 40 channels [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found total of 40 G.729 licenses How do i roll this back? Just delete codec_g729a.so ? -- _ -- 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] Unable to load users.conf
If you meant the user running Asterisk is root, this is a less than optimal situation that can lead to really big problems. Why? Steve please explain. Well, if an attacker manages to inject some code and Asterisk is running as root, poof goes your system or you get an astronomical bill from your trunk provider. Likewise with file permissions. Suppose you're trying to get something working and you suspect it's a permissions issue so you chmod a bunch of stuff to 777. Then suppose a local user with a grudge does something like this: echo '#exec rm --farce --recursive /*'\ /etc/asterisk/extensions-**local.conf (or whatever your package names one of it's 'include' files.) The next time Asterisk reloads the dialplan, poof. any link for me so i can learn more about security practices with asterisk? i'm using a public ip. -- _ -- 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] Automatic jump from line to line for incoming calls and the problem in DAHDI
On Wednesday 17 October 2012, bilal ghayyad wrote:
Actually I am not talking on how to handle it in the extensions.conf
because I am doing same as you wrote. But even so, I am facing a problem
that some calls are captured and some calls are not captured.
Currently, I set the callwaiting=no in the chan_dahdi.conf, it seems it is
working fine. But I am not sure if this is really the required
configuration to fix it or there is something else.
Any advise.
We need to determine the exact circumstances under which Asterisk is missing
incoming calls, which is going to require some low-level hacking.
Wire up two LEDs back to back (so whichever way the current is flowing, one of
them will always light up).
*-|-*
*-|-*
Put in series with this pair a 470 nF capacitor and a 100kΩ resistor:
*-| |-/\/\/\/-[LEDs]-*
When this contraption is connected across an analogue phone line, the LEDs
will light up. (One lights on the crest of the waveform and one on the
trough, but this should be happening too fast for the eye to see, and it will
just look like both are lit.) The capacitor blocks DC on the line, so the
LEDs will be off unless AC is present. Make up 4 of these devices, so you can
monitor the ringing status of each of the analogue lines going into your
Asterisk box.
In your extensions.conf, make sure that you have NoOp(${EXTEN}) somewhere in
the [from-pstn] context, so you can see what number the upstream exchange was
sending. Be sure to include something which will keep a caller on the line
for awhile. Connect up a laptop with an SSH client to the network, so you
have an Asterisk console *and* can see the ringing status of all 4 lines.
Lastly, you will need 4 mobile phones; and possibly volunteers to operate
them, while you watch the Asterisk console and the LEDs.
Now you can investigate properly what is happening when you dial your analogue
lines:
* Call a line which is not busy, by its own number. Does Asterisk respond to
the call?
* Call a line which *is* busy, by its own number. Does the call automatically
appear on another line? Does Asterisk respond to it?
* Call a non-busy line while another line is busy. Does Asterisk respond to
the call?
You should be able to work out eventually just what is causing Asterisk to
miss calls.
--
AJS
Answers come *after* questions.
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[asterisk-users] Problems with AGI and existing channel
Hi, Asterisk 1.8.10.0-1digium1~squeeze built by pbuilder @ nighthawk on a x86_64 running Linux on 2012-03-08 23:05:09 UTC We have some problem when running a AGI script (build with PHP), existing channels (all of them) gets a hickup and then continues. We are using AGI to lookup incoming calls in directory. It is kinda annoying, and I don't understand how it can be related. Best regards Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with AGI and existing channel
On Thursday 18 October 2012, Magnus Löfqvist wrote: We have some problem when running a AGI script (build with PHP), existing channels (all of them) gets a hickup and then continues. We are using AGI to lookup incoming calls in directory. It is kinda annoying, and I don't understand how it can be related. I think you forgot to attach the actual script. Without that, we can't help you. -- AJS Answers come *after* questions. -- _ -- 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] [Spam] Re: Problems with AGI and existing channel
Hi, The script is not essentials in this. I have tried with a empty script... It stills generate problems with existing channels. But if a try with a empty perl script, it dosent... seems to be something that php is doing when starting up. Best regards Magnus -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För A J Stiles Skickat: den 18 oktober 2012 13:57 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: [Spam] Re: [asterisk-users] Problems with AGI and existing channel On Thursday 18 October 2012, Magnus Löfqvist wrote: We have some problem when running a AGI script (build with PHP), existing channels (all of them) gets a hickup and then continues. We are using AGI to lookup incoming calls in directory. It is kinda annoying, and I don't understand how it can be related. I think you forgot to attach the actual script. Without that, we can't help you. -- AJS Answers come *after* questions. -- _ -- 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] Problems with AGI and existing channel
On Thu, Oct 18, 2012 at 6:33 AM, Magnus Löfqvist m...@vmi.se wrote: Hi, ** ** “Asterisk 1.8.10.0-1digium1~squeeze built by pbuilder @ nighthawk on a x86_64 running Linux on 2012-03-08 23:05:09 UTC” ** ** We have some problem when running a AGI script (build with PHP), existing channels (all of them) gets a “hickup” and then continues. You say all of your connected channels experience an audio glitch? Sounds like PHP is briefly consuming all of your CPU or RAM and causing Asterisk to fail to meet timing demands. Whatever your PHP CLI interface is (mine is just `php` but yours may be different), run time php /dev/null The returned time should be very, very small. Mine was real0m0.018s. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Unable to load users.conf
On Thu, 18 Oct 2012, Rizha Yuherdianto wrote: any link for me so i can learn more about security practices with asterisk? i'm using a public ip. I think John Todd did a doc that is included with Asterisk. http://nerdvittles.com/ has had a couple of recent articles on security. There was a post yesterday to asterisk-biz with the obtuse subject 'Strange request' that had a couple of links. I haven't read them but they look like good background material to some of the challenges unique to VOIP. Read, read, read. It's a constantly changing battle. -- 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
Re: [asterisk-users] Problems with AGI and existing channel
On Thu, 18 Oct 2012, Magnus Löfqvist wrote: I have tried with a empty script... It stills generate problems with existing channels. But if a try with a empty perl script, it dosent... seems to be something that php is doing when starting up. Can you explain more precisely what you mean by 'hickup?' If you truely mean an 'empty' file, then that's not a valid AGI. Which AGI library are you using. AGI is a 'per channel' kind of thing. If your AGI is affecting all active channels then that is kind of odd. Any chance of a console log with verbose and debug cranked up while this is happening? If you enable AGI debugging that may also give a clue. -- 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
[asterisk-users] Setting CDR fields in connected macro of Queue command
Trying to set some CDR fields in the connected macro of a queue
command. None of the custom fields I set are stored in the database,
but I can set userfield and it does get set. I think that the macro
runs on the agent's channel, not the caller's, and this might contribute
to the problem.
From the sample below userfield (and its alias operatorid) are saved
in the CDR record, but salesqueue_answered is not.
What am I missing?
Asterisk 1.8.10.1~dfsg-1ubuntu1
same =n,Queue(sales,tc,QueueConnected)
[macro-QueueConnected]
; this runs on the agent/member's channel
exten =s,1,NoOp()
same =n,Set(CDR(salesqueue_answered)=1)
same
=n,Set(OPERATORID=${ODBC_OPERATORID_FROM_ADDRESS(${MEMBERINTERFACE})})
; userfield is mapped to operatorid in cdr_adaptive_odbc because
setting operatorid directly doesn't work here
same =n,Set(CDR(userfield)=${OPERATORID})
--
Mitch
--
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[asterisk-users] Counting calls in progress from AMI
I want to track the number of calls up at any given time, through the AMI. I found the Link and Unlink commands as the most likely candidates - is that the right way? Also, a comment on the wiki suggests that Link may be called several times for a single bridge if transcoding is required. That blows up accuracy of my count of course... 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] Counting calls in progress from AMI
The simplest way to accurately do this would be to issue command core show channels verbose From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 18, 2012 9:58 AM To: Asterisk Users List Subject: [asterisk-users] Counting calls in progress from AMI I want to track the number of calls up at any given time, through the AMI. I found the Link and Unlink commands as the most likely candidates - is that the right way? Also, a comment on the wiki suggests that Link may be called several times for a single bridge if transcoding is required. That blows up accuracy of my count of course... 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] Counting calls in progress from AMI
I need to do this from the AMI (not the CLI)...I don't *think* a comparable command exists from the AMI. As well, I don't want to poll the system for calls so I'm hoping to trap a call bridged,unbridged type event. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Thursday, October 18, 2012 10:59 AM To: Asterisk Users List Subject: Re: [asterisk-users] Counting calls in progress from AMI The simplest way to accurately do this would be to issue command “core show channels verbose” From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 18, 2012 9:58 AM To: Asterisk Users List Subject: [asterisk-users] Counting calls in progress from AMI I want to track the number of calls up at any given time, through the AMI. I found the Link and Unlink commands as the most likely candidates - is that the right way? Also, a comment on the wiki suggests that Link may be called several times for a single bridge if transcoding is required. That blows up accuracy of my count of course... 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] Counting calls in progress from AMI
The AMI Command function issues CLI commands, but carry on. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 18, 2012 10:20 AM To: Asterisk Users List Subject: Re: [asterisk-users] Counting calls in progress from AMI I need to do this from the AMI (not the CLI)...I don't *think* a comparable command exists from the AMI. As well, I don't want to poll the system for calls so I'm hoping to trap a call bridged,unbridged type event. _ From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Thursday, October 18, 2012 10:59 AM To: Asterisk Users List Subject: Re: [asterisk-users] Counting calls in progress from AMI The simplest way to accurately do this would be to issue command core show channels verbose From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 18, 2012 9:58 AM To: Asterisk Users List Subject: [asterisk-users] Counting calls in progress from AMI I want to track the number of calls up at any given time, through the AMI. I found the Link and Unlink commands as the most likely candidates - is that the right way? Also, a comment on the wiki suggests that Link may be called several times for a single bridge if transcoding is required. That blows up accuracy of my count of course... 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] Counting calls in progress from AMI
I guess you are looking for event handler, which can be polled programatically n not via manual command entry? Mitul On Oct 18, 2012 8:53 PM, Danny Nicholas da...@debsinc.com wrote: The AMI Command function issues CLI commands, but carry on. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Michelle Dupuis *Sent:* Thursday, October 18, 2012 10:20 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Counting calls in progress from AMI ** ** I need to do this from the AMI (not the CLI)...I don't *think* a comparable command exists from the AMI. As well, I don't want to poll the system for calls so I'm hoping to trap a call bridged,unbridged type event. -- *From:* asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [ da...@debsinc.com] *Sent:* Thursday, October 18, 2012 10:59 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Counting calls in progress from AMI The simplest way to accurately do this would be to issue command “core show channels verbose” *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Michelle Dupuis *Sent:* Thursday, October 18, 2012 9:58 AM *To:* Asterisk Users List *Subject:* [asterisk-users] Counting calls in progress from AMI I want to track the number of calls up at any given time, through the AMI. I found the Link and Unlink commands as the most likely candidates - is that the right way? Also, a comment on the wiki suggests that Link may be called several times for a single bridge if transcoding is required. That blows up accuracy of my count of course... 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 -- _ -- 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] Counting calls in progress from AMI
core show calls works for me on asterisk-11 I know it's a poll, but it's a very simple call ;) On 18 October 2012 16:28, Mitul Limbani mi...@enterux.in wrote: I guess you are looking for event handler, which can be polled programatically n not via manual command entry? Mitul On Oct 18, 2012 8:53 PM, Danny Nicholas da...@debsinc.com wrote: The AMI Command function issues CLI commands, but carry on. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 18, 2012 10:20 AM To: Asterisk Users List Subject: Re: [asterisk-users] Counting calls in progress from AMI I need to do this from the AMI (not the CLI)...I don't *think* a comparable command exists from the AMI. As well, I don't want to poll the system for calls so I'm hoping to trap a call bridged,unbridged type event. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Thursday, October 18, 2012 10:59 AM To: Asterisk Users List Subject: Re: [asterisk-users] Counting calls in progress from AMI The simplest way to accurately do this would be to issue command “core show channels verbose” From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 18, 2012 9:58 AM To: Asterisk Users List Subject: [asterisk-users] Counting calls in progress from AMI I want to track the number of calls up at any given time, through the AMI. I found the Link and Unlink commands as the most likely candidates - is that the right way? Also, a comment on the wiki suggests that Link may be called several times for a single bridge if transcoding is required. That blows up accuracy of my count of course... 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 -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Counting calls in progress from AMI
Hi, Though I've moved this functionality to freeSWITCH but here is the logic. *AMI approach:* Use perl POE and when received Bridged event increment the counter, and similarly on Hangup decrement the counter. Since I see that there are some issues mentioned by OP so I assume using plain Answer/bridge and hangup events aren't working perfectly. *Dialplan approach:* * * The other alternative simple and effective way is use Dial() command with U or M argument. U command triggers a Subroutine and M executes a Macro Just before connecting/Bridging the call between AB party. There in Macro/Subroutine execute a system() with perl script in it say increment.pl That perl script connects to redis(memcache) and increments the callcounter KVP - thats all. Next on hangup execute the same system() application with perl script in it say decrement.pl - this script in terms decrement the callcounter KVP in redis. Any other applications which needs the call-count should now interface to Redis and fetch the callcounter. I hope I made some sense :) Thanks, Sammy On Thu, Oct 18, 2012 at 8:22 PM, Danny Nicholas da...@debsinc.com wrote: The AMI Command function issues CLI commands, but carry on. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Michelle Dupuis *Sent:* Thursday, October 18, 2012 10:20 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Counting calls in progress from AMI ** ** I need to do this from the AMI (not the CLI)...I don't *think* a comparable command exists from the AMI. As well, I don't want to poll the system for calls so I'm hoping to trap a call bridged,unbridged type event. -- *From:* asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [ da...@debsinc.com] *Sent:* Thursday, October 18, 2012 10:59 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Counting calls in progress from AMI The simplest way to accurately do this would be to issue command “core show channels verbose” *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Michelle Dupuis *Sent:* Thursday, October 18, 2012 9:58 AM *To:* Asterisk Users List *Subject:* [asterisk-users] Counting calls in progress from AMI I want to track the number of calls up at any given time, through the AMI. I found the Link and Unlink commands as the most likely candidates - is that the right way? Also, a comment on the wiki suggests that Link may be called several times for a single bridge if transcoding is required. That blows up accuracy of my count of course... 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 -- _ -- 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 as IVR using 3g usb modem
hi, I want to use asterisk as IVR system ,but to make and receive GSM call, i want to use 3g usb modem.(voice enabled)http://www.huaweidevice.co.in/Products/MobileBroadband/E303c.php and i want to install this system on two different machine1 on mac os x -2 raspberry pi- (debian wheezy)--http://www.raspberrypi.org/ thanx in advance.. -- _ -- 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 as IVR using 3g usb modem
Short answer is, its not possible Long answer, why it is not !! U would have to write a dahdi module for this 3G modem to help asterisk understand it as standard gsm channel. Hope that help, Mitul On Oct 18, 2012 9:16 PM, Mahendra Dobariya mahendra_mahen...@hotmail.com wrote: hi, I want to use asterisk as IVR system , but to make and receive GSM call, i want to use 3g usb modem.(voice enabled) http://www.huaweidevice.co.in/Products/MobileBroadband/E303c.php and i want to install this system on two different machine 1 on mac os x - 2 raspberry pi- (debian wheezy)--http://www.raspberrypi.org/ thanx in advance.. -- _ -- 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] Counting calls in progress from AMI
On 18 Oct 2012, at 17:19, Michelle Dupuis wrote:
I need to do this from the AMI (not the CLI)...I don't *think* a comparable
command exists from the AMI.
As well, I don't want to poll the system for calls so I'm hoping to trap a
call bridged,unbridged type event.
Michelle,
if you do not want to poll Asterisk with an AMI 'Status' event (that returns a
list of open channels) counting the bridged, unbridged events is possible, but
not that easy..
There are two gotchas around this:
One is that a single call, depending of what the dialplan is doing, can involve
several bridged, unbridged events;
The second is that some kind of calls (I call them 'one legged calls': anything
resembling an IVR or a call to get voicemail) do not get any bridged, unbridged
events, at all. The same applies for calls that are sent to a MeetMe
conference: there you see specific MeetMe events for conference rooms.
If you have access to the dialplan (that is if you are in charge of it and you
can modify it) you could add some user defined events, to mark the 'rising' of
a call and its connection to the far side. The dialplan can send pretty much
what you like by using UserEvent.
As an example:
exten = _00.,n,UserEvent(DNIS-Ext,Exten: ${EXTEN},CallerID:
${CALLERID(num)},DNID: ${CALLERID(dnid)},DisplayID:
${PersonalID_Num},ChannelID: ${CHANNEL},RDNIS: ${RDNIS})
The first string (DNIS-Ext) is a marker: anything that you want to receive that
'brands' the user event to your suiting.
The rest is a list of (name, value) pairs, giving the details you might need on
the AMI processing side.
Beware of any loop in the dialplan: it might be that you will get multiple
copies of the same event, if the dialplan execution is such that the same
extension is visited more than once during the processing of a call.
HTH,
Aldo
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[asterisk-users] Question on AMI and ChanIsAvail
I was wanting to call ChanIsAvail from AMI. I logged in and issues command, Action: Command Command: ChanIsAvail DAHDI/1 my response was this: event_list=0 ret=158 Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]No such command 'ChanIsAvail DAHDI/1' Is there any way to tell if a channel is available through AMI? Did I format my request wrong? I am on 1.4.43. Thanks, Jerry -- _ -- 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] Question on AMI and ChanIsAvail
Using AMI you might want to use hints instead of ChanIsAvail since that command puts its' output back into the dialplan as variables. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, October 18, 2012 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question on AMI and ChanIsAvail I was wanting to call ChanIsAvail from AMI. I logged in and issues command, Action: Command Command: ChanIsAvail DAHDI/1 my response was this: event_list=0 ret=158 Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]No such command 'ChanIsAvail DAHDI/1' Is there any way to tell if a channel is available through AMI? Did I format my request wrong? I am on 1.4.43. Thanks, Jerry -- _ -- 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 as IVR using 3g usb modem
On 18 Oct 2012, at 16:50, Mitul Limbani wrote: U would have to write a dahdi module for this 3G modem to help asterisk understand it as standard gsm channel. Look up chan_datacard (i think that's what it's called from memory). Steve -- _ -- 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] Question on AMI and ChanIsAvail
On 18 Oct 2012, at 17:58, Jerry Geis wrote: I was wanting to call ChanIsAvail from AMI. I logged in and issues command, Action: Command Command: ChanIsAvail DAHDI/1 my response was this: event_list=0 ret=158 Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]No such command 'ChanIsAvail DAHDI/1' Is there any way to tell if a channel is available through AMI? Did I format my request wrong? I am on 1.4.43. Thanks, Jerry You can get (on the CLI) the list of AMI commands supported by your Asterisk installation issuing manager show commands. The AMI command that tells you what a terminal is doing at any time is ExtensionState (I guess it's available on 1.4..). But if you add hints to your dialplan, then Asterisk will send an AMI event whenever a terminal in the list of hints is changing state. You would need to listen to all such events and keep track of each line/extension; issuing a list of ExtensionState events only when you launch your tracking process. Cheers, Aldo -- _ -- 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] Counting calls in progress from AMI
Considering this your real question I want to track the number of calls up
at any given time, through the AMI
Comparable Command Exists !!
You can simply run CLI commands *through *AMI and receive the response.
Command used for this is itself Command in AMI. If you still want to do
it by counting the event stream then POE is the best solution if you know
PERL, but still why wasting resources in counting events.If you have other
plans like billing then it is understandable otherwise go easy way and add
a cron job or have a UI in php or whatever.
Manager Command:
Command command,all Execute Asterisk CLI Command
On Thu, Oct 18, 2012 at 8:53 AM, Aldo Bergamini aabe...@gmail.com wrote:
On 18 Oct 2012, at 17:19, Michelle Dupuis wrote:
I need to do this from the AMI (not the CLI)...I don't *think* a
comparable command exists from the AMI.
As well, I don't want to poll the system for calls so I'm hoping to trap
a call bridged,unbridged type event.
Michelle,
if you do not want to poll Asterisk with an AMI 'Status' event (that
returns a list of open channels) counting the bridged, unbridged events is
possible, but not that easy..
There are two gotchas around this:
One is that a single call, depending of what the dialplan is doing, can
involve several bridged, unbridged events;
The second is that some kind of calls (I call them 'one legged calls':
anything resembling an IVR or a call to get voicemail) do not get any
bridged, unbridged events, at all. The same applies for calls that are sent
to a MeetMe conference: there you see specific MeetMe events for conference
rooms.
If you have access to the dialplan (that is if you are in charge of it and
you can modify it) you could add some user defined events, to mark the
'rising' of a call and its connection to the far side. The dialplan can
send pretty much what you like by using UserEvent.
As an example:
exten = _00.,n,UserEvent(DNIS-Ext,Exten: ${EXTEN},CallerID:
${CALLERID(num)},DNID: ${CALLERID(dnid)},DisplayID:
${PersonalID_Num},ChannelID: ${CHANNEL},RDNIS: ${RDNIS})
The first string (DNIS-Ext) is a marker: anything that you want to receive
that 'brands' the user event to your suiting.
The rest is a list of (name, value) pairs, giving the details you might
need on the AMI processing side.
Beware of any loop in the dialplan: it might be that you will get multiple
copies of the same event, if the dialplan execution is such that the same
extension is visited more than once during the processing of a call.
HTH,
Aldo
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Regards
**
Muhammad Salman
***
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Re: [asterisk-users] Agents in more than one queue at once
Are there any developers that are familiar with the Queue() app implementation and how it distributes calls? Alex Forster -- _ -- 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] Counting calls in progress from AMI
Considering this your real question I want to track the number of calls up at any given time, through the AMI Comparable Command Exists !! You can simply run CLI commands through AMI and receive the response. Command used for this is itself Command in AMI. If you still want to do it by counting the event stream then POE is the best solution if you know PERL, but still why wasting resources in counting events.If you have other plans like billing then it is understandable otherwise go easy way and add a cron job or have a UI in php or whatever. Manager Command: Command command,all Execute Asterisk CLI Command You should only use the AMI Command action if there is no other AMI action available to do the job. For instance, the AMI CoreShowChannels action is better than using the CLI core show channels command through AMI. manager show command CoreShowChannels Richard -- _ -- 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 as IVR using 3g usb modem
On Thu, 2012-10-18 at 17:18 +0100, Steven Howes wrote: On 18 Oct 2012, at 16:50, Mitul Limbani wrote: U would have to write a dahdi module for this 3G modem to help asterisk understand it as standard gsm channel. Look up chan_datacard (i think that's what it's called from memory). Steve It got renamed, and is now: http://code.google.com/p/asterisk-chan-dongle/ It's just a tgz, no docu, no wiki.. chan_dongle-1.1.r14.tgz chan_dongle version 1.1 revision 14 sources Featured 6 days ago 6 days ago 184 KB 159 Looks a bit fresh... -- _ -- 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 sends wrong fxs 'Idle' hints
It seems fixing this issue and [1] would require significant efforts: https://issues.asterisk.org/jira/browse/ASTERISK-20556 Too bad, it means DAHDI is a no way for me, I will have to switch to SIP DECTs :( Niccolò [1]http://asteriskfaqs.org/2012/10/08/asterisk-users/how-to-avoid-automatic-answer-with-callwaitingyes-on-fxs-channels.html Il 09/10/2012 13:34, Niccolò Belli ha scritto: Hi, I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a remote peer and an fxs phone gets connected and the remote peer hangsup, then asterisk sends the Idle state to notify the watcher before you hangup the fxs phone! Such a way if the user forgets to hangup the fxs phone (which is a cordless for example) then the operators will keep sending calls to him because the light on their function keys switched off! Cheers, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to avoid automatic answer with callwaiting=yes on fxs channels?
It seems fixing this issue and [1] would require significant efforts: https://issues.asterisk.org/jira/browse/ASTERISK-20556 Too bad, it means DAHDI is a no way for me, I will have to switch to SIP DECTs :( Niccolò [1]http://asteriskfaqs.org/2012/10/09/asterisk-users/asterisk-sends-wrong-fxs-idle-hints.html Il 08/10/2012 13:20, Niccolò Belli ha scritto: I will make an example: A is an fxs phone with callwaiting=yes in chan_dahdi.conf X calls A. A answers. Y calls A. A hears the call waiting tone. Now if A hangs up before X, then A rings again (which is what I want). BUT if X hangs up first, then A automatically answers Y without even ringing. Is there a way to avoid it? If X hangs up first I want A to hear the busy tone until it hangs up too. Then I want A to ring again. Otherwise if both A and X hang up at the same time there is no way to know what happened. Thanks, Niccolò -- http://www.linuxsystems.it -- _ -- 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] AstDB with Sqlite
As you may know, asterisk version 10 and high use sqlite. Are any examples or documentation how to use in dialplan? -- _ -- 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] motif and psi - no sound
Hello! I'm trying to use psi+ to conect to asterisk using chan_motif and vise versa. Connection looks good, but no sound. As I see there is some traffic (22.229 is my desktop with psi) 08:38:37.463506 IP 192.168.22.229.8010 192.168.22.19.17012: UDP, length 82 08:38:37.481325 IP 192.168.22.229.8010 192.168.22.19.17012: UDP, length 82 08:38:37.481885 IP 192.168.22.19.17012 192.168.22.229.8010: UDP, length 65 08:38:37.501745 IP 192.168.22.19.17012 192.168.22.229.8010: UDP, length 65 but I hear nothing :-( psi-psi works OK. And I have speex16 allowed in motif.conf Any ideas? Thank you! -- _ -- 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
