[asterisk-users] pjsip extensions rings but call drop on answer

Hi, I created an IAX2 trunk between my old Asterisk 1.4 server (A) and my new one with v. 16.10.0 (B). The trunk seems to be up, and the calls are initiated, eg. an extension from A can dial an extension in B which rings. However, as soon as the extension in B answers, the call is terminated.

Re: [asterisk-users] Various extensions ring once and go to voicemail

Subject: Re: [asterisk-users] Various extensions ring once and goto voicemail - Thomas Peters >Carlos and Stefan (and other who have helped): >I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling >Asterisk is unrealistic in my position but I wonder

Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters

em 1942 N 17th Street | Milwaukee, WI  53205 Check us out on Facebook & Twitter -Original Message- From: asterisk-users On Behalf Of Stefan Viljoen Sent: Tuesday, January 15, 2019 12:05 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Various extensions ri

Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters

Check us out on Facebook & Twitter -Original Message- From: asterisk-users On Behalf Of Doug Lytle Sent: Tuesday, January 15, 2019 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail - Th

Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters

>>> Carlos and Stefan (and other who have helped): Thomas, You stated that your virtual environment was Oracle, would that equate to VirtualBox? Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters

M To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters Here’s what I get: apbx*CLI> module show like timing Module Description Use Count res_timing_pthread.so

Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters

Here’s what I get: apbx*CLI> module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_dahdi.soDAHDI Timing Interface 4 2 modules

Re: [asterisk-users] Various extensions ring once and go to voicemail

-users@lists.digium.com *Subject:* Re: [asterisk-users] Various extensions ring once and go to voicemail On 1/14/19 4:02 PM, Duncan Turnbull wrote: Sent from my iPad On 15/01/2019, at 10:34 AM, Thomas Peters mailto:tpet...@mcts.org>> wrote: Duncan: You may have it r

Re: [asterisk-users] Various extensions ring once and go to voicemail

demcts.com/> 1942 N 17th Street | Milwaukee, WI 53205 Check us out on Facebook<https://www.facebook.com/mcts> & Twitter <https://twitter.com/RideMCTS> From: asterisk-users On Behalf Of Carlos Chavez Sent: Monday, January 14, 2019 4:08 PM To: asterisk-users@lists.digium.com

Re: [asterisk-users] Various extensions ring once and go to voicemail

On 1/14/19 4:02 PM, Duncan Turnbull wrote: Sent from my iPad On 15/01/2019, at 10:34 AM, Thomas Peters > wrote: Duncan: You may have it right—I took one phone and set the ring time to 60 seconds. I now get about 4 rings on that one. I wonder how I can change

Re: [asterisk-users] Various extensions ring once and go to voicemail

mp; Twitter > > From: asterisk-users On Behalf Of > Duncan > Sent: Monday, January 14, 2019 2:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail > > > > On Tue, J

Re: [asterisk-users] Various extensions ring once and go to voicemail

k-users On Behalf Of Duncan Sent: Monday, January 14, 2019 2:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters mailto:tpet...@mcts.org>> wrote:

Re: [asterisk-users] Various extensions ring once and go to voicemail

On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters wrote: We have an old Asterisk 1.8.7.0 system desperately need to keep alive for another 6 months or so. We had all kinds of hardware problems, so we virtualized it. Thats a while back, I think it tended to use zaptel or dahdi hardware as a

[asterisk-users] Various extensions ring once and go to voicemail

We have an old Asterisk 1.8.7.0 system desperately need to keep alive for another 6 months or so. We had all kinds of hardware problems, so we virtualized it. Now, random extensions ring once and go straight to voicemail. I went to one of the affected extensions and changed the ring time from

Re: [asterisk-users] Long extensions that contain dashes

On Tue, May 29, 2018 at 08:32:39PM -0700, David P wrote: > We would like to use 20-char extension values that use dashes and alphanums > after the first four digits. In order to handle these via pattern-matching, > how can I define a pattern that allows dashes? There seems to be no option > at

[asterisk-users] Long extensions that contain dashes

We would like to use 20-char extension values that use dashes and alphanums after the first four digits. In order to handle these via pattern-matching, how can I define a pattern that allows dashes? There seems to be no option at http://the-asterisk-book.com/1.6/einleitung-regex.html#re

Re: [asterisk-users] Realtime extensions, Multiple SQL lookups?

It's been years since I have used real time but can it be because of the GoSub? On Mon, Oct 16, 2017 at 12:39 PM, John Kiniston wrote: > I'm toying with the idea of replacing a statically generated file I > include in my extensions.conf with a realtime lookup against

[asterisk-users] Realtime extensions, Multiple SQL lookups?

I'm toying with the idea of replacing a statically generated file I include in my extensions.conf with a realtime lookup against my database. I've got it working but something seems off in my logs, It looks like I'm getting two lookups for every priority? [Oct 12 16:45:24]

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: and...@telesip.net Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Guys, hi. I have not solved the problem. Outgoing calls to remote extensions drops on 5-20 seconds. Incoming calls work perfectly

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

...@gmail.com Sent: Wednesday, December 18, 2013 9:30 PM To: and...@telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. I set canreinvite=very in the remote

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. I set canreinvite=very in the remote extension, and now the call not drops. Valid solution? On Wed, Dec 18, 2013 at 6:38 PM, Andres and...@telesip.net wrote

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: and...@telesip.net Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Guys, hi. I have not solved the problem. Outgoing calls to remote extensions drops on 5-20 seconds. Incoming calls work perfectly. Hope you

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Monday, March 10, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: and...@telesip.net Subject: Re: [asterisk-users] Remote extensions call drops

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Monday, March 10, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: and...@telesip.net Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Guys, hi. I have not solved the problem. Outgoing calls

[asterisk-users] Remote extensions call drops after 20 seconds.

Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 3:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Remote extensions call drops after 20 seconds. Hello. I have a problem

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

javascript:; Sent: Wednesday, December 18, 2013 3:09 PM To: asterisk-users@lists.digium.com javascript:; Subject: [asterisk-users] Remote extensions call drops after 20 seconds. Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

-users] Remote extensions call drops after 20 seconds. Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f Thank you! -- Allan

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

To: asterisk-users@lists.digium.com Subject: [asterisk-users] Remote extensions call drops after 20 seconds. Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Rodrigo, thanks for reply. 1- RTP ports is forwarded correctly on the NAT router. 2- externip is my public ip. 3- All my extensions have nat=yes by default. 4

Re: [asterisk-users] Remote extensions call drops after 20 seconds.

On 12/18/13, 3:09 PM, alp...@gmail.com wrote: Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f When the call is setup I see

[asterisk-users] Monitor extensions status.

Greetings. I got two extensions on my asterisk that autenticates from outside our network, via internet. Is there a way to monitor, in certain time periods, if they are available (online) and send some sort of notification if they don't? There are two extensions to monitor, they belong to the

Re: [asterisk-users] Monitor extensions status.

2013/1/8 Luis H. Forchesatto luisforchesa...@gmail.com Greetings. I got two extensions on my asterisk that autenticates from outside our network, via internet. Is there a way to monitor, in certain time periods, if they are available (online) and send some sort of notification if they

Re: [asterisk-users] Monitor extensions status.

Top and bottom post in the same email... don't open again the thread :-) #!/bin/bash res=`sudo /usr/sbin/asterisk -rx 'sip show peer $1' | grep Status | cut -d\: -f 2 | cut -d\ -f 2` if [ $res == OK ] then echo OK is registered exit 0 else echo WARNING peer not registered exit 1 2013/1/8 Luis

Re: [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3

-Commercial Discussion Subject: Re: [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3 But when I try to call one extension created with lua I got a message telling that extension doesnt exist on default context. Am I missing something? 2011/2/15 Tilghman Lesher tilgh

[asterisk-users] Lua extensions are not working on asterisk 1.8.2.3

Hi, After compiling a installing asterisk 1.8.2.3 I wanted to play with lua but I noticed that extensions created in extensions.lua was not being registered with asterisk. uga1*CLI dialplan show [ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ] 's' =1. NoOp()

Re: [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3

On Tuesday 15 February 2011 11:06:32 Carlo Pires wrote: Hi, After compiling a installing asterisk 1.8.2.3 I wanted to play with lua but I noticed that extensions created in extensions.lua was not being registered with asterisk. uga1*CLI dialplan show [ Context

Re: [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3

But when I try to call one extension created with lua I got a message telling that extension doesnt exist on default context. Am I missing something? 2011/2/15 Tilghman Lesher tilgh...@meg.abyt.es: On Tuesday 15 February 2011 11:06:32 Carlo Pires wrote: Hi, After compiling a installing

Re: [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3

Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3 But when I try to call one extension created with lua I got a message telling that extension doesnt exist on default context. Am I missing something? 2011/2/15 Tilghman Lesher

Re: [asterisk-users] SIP Extensions and loss of Internet connection

. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Extensions and loss of Internet connection On Mon, Nov 22, 2010 at 11:44 AM, Daniel Bareiro daniel-lis...@gmx.net wrote: Hi, Alejandro. A few days I have problems connecting to the Internet

Re: [asterisk-users] SIP Extensions and loss of Internet connection

- Original Message - Hi all! A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I found that it

Re: [asterisk-users] SIP Extensions and loss of Internet connection

Hi, Phil. A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I found that it can be related to a bug of

Re: [asterisk-users] SIP Extensions and loss of Internet connection

- Original Message - Hi, Phil. A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I

Re: [asterisk-users] SIP Extensions and loss of Internet connection

Does you Asterisk server point to an internal DNS or to your router ? The /etc/resolv.conf of the host on which I installed Asterisk points to an internal DNS. Is there a parameter in the Asterisk configuration where also I have to force the use of an internal DNS server? Do your

Re: [asterisk-users] SIP Extensions and loss of Internet connection

On Sun, Nov 21, 2010 at 8:14 PM, Daniel Bareiro daniel-lis...@gmx.net wrote: Hi all! A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. You have to be a bit more specific. For

Re: [asterisk-users] SIP Extensions and loss of Internet connection

Hi, Alejandro. A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. You have to be a bit more specific. For example is your Asterisk box behind a router/nat? Or does your

Re: [asterisk-users] SIP Extensions and loss of Internet connection

On Mon, Nov 22, 2010 at 11:44 AM, Daniel Bareiro daniel-lis...@gmx.net wrote: Hi, Alejandro. A few days I have problems connecting to the Internet on my house [...] It would appear that the server for some reason was 'locked'. For example, when I try to register from Twinkle softphone, I get

[asterisk-users] SIP Extensions and loss of Internet connection

Hi all! A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I found that it can be related to a bug of chan_sip, can

[asterisk-users] Multiple extensions - same context

Hey Everyone; I inherited an Asterisk box where the dialplan is a real mess. ( I would actually be embarrassed to post some of the stuff!) So, here is what I need to do - and again, I am looking for fishing nets and places to cast them - if I don't figure it out, I will never freakin' learn!

Re: [asterisk-users] Multiple extensions - same context

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Silver Thorne Sent: Thursday, November 04, 2010 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Multiple extensions

Re: [asterisk-users] Multiple extensions - same context

On Thu, 2010-11-04 at 12:07 -0400, Silver Thorne wrote: Hey Everyone; I inherited an Asterisk box where the dialplan is a real mess. ( I would actually be embarrassed to post some of the stuff!) So, here is what I need to do - and again, I am looking for fishing nets and places to cast

[asterisk-users] run extensions after call moved to queue and answered by member

Hello. Can You provide example, how can i run specific extension after incoming call going into queue and answered (but not hangup). (i want to use System(echo .) after member of specific queue answered a call); Thank You. -- Vasiliy G Tolstov v.tols...@selfip.ru Selfip.Ru --

Re: [asterisk-users] run extensions after call moved to queue and answered by member

Which version of asterisk are you running? Older versions allowed for an AGI to be called when the queued call got connected with an agent. core show application queue Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI]) The optional AGI parameter will setup an AGI script to be

Re: [asterisk-users] run extensions after call moved to queue and answered by member

В Чтв, 20/05/2010 в 05:49 -0700, Jim Dickenson пишет: Which version of asterisk are you running? Thank's for answer. One minute before i found answer - add membermacro to quesues.conf I'm use asterisk 1.6 -- Vasiliy G Tolstov v.tols...@selfip.ru Selfip.Ru --

Re: [asterisk-users] Realtime extensions

On 02/20/2010 01:53 AM, jonas kellens wrote: I have read on this list that people do not get a reply if they ask stupid questions. Is this then a stupid question that I ask ? If nobody has ever combined extensions.conf and realtime in a way that I want to do, I wanna hear it too. Even if

Re: [asterisk-users] Realtime extensions

On 19/02/10 8:15 AM, jonas kellens wrote: How about something like : [mycontext] exten = 100,1,NoOp(calling 100) exten = 100,n,NoOp(going realtime) switch = Realtime/mycont...@realtime_extensions mailto:mycont...@realtime_extensions ; from here on we use realtime And then my MySQL-DB

Re: [asterisk-users] Realtime extensions

On 20/02/10 10:53 PM, jonas kellens wrote: I have read on this list that people do not get a reply if they ask stupid questions. Is this then a stupid question that I ask ? If nobody has ever combined extensions.conf and realtime in a way that I want to do, I wanna hear it too. Even if this

Re: [asterisk-users] Realtime extensions

I have read on this list that people do not get a reply if they ask stupid questions. Is this then a stupid question that I ask ? If nobody has ever combined extensions.conf and realtime in a way that I want to do, I wanna hear it too. Even if this means no solution for me. Then I know it's not

Re: [asterisk-users] Realtime extensions

Anyone know if my example of combining extensions.conf and realtime extensions is doable ?? Kind regards, Jonas. On Thu, 2010-02-18 at 20:15 +0100, jonas kellens wrote: How about something like : [mycontext] exten = 100,1,NoOp(calling 100) exten = 100,n,NoOp(going realtime) switch =

[asterisk-users] Realtime extensions

Hello list ! Can realtime dialplan be combined with 'hardcoded' dialplan in extensions.conf ?? Does a context need completely be written or in extensions.conf or in the mysql-table 'extensions_table' ? Or can I combine the two with the 'switch'-statement ?? Kind regards, Jonas. --

Re: [asterisk-users] Realtime extensions

On Thu, 2010-02-18 at 19:46 +0100, jonas kellens wrote: Does a context need completely be written or in extensions.conf or in the mysql-table 'extensions_table' ? Or can I combine the two with the 'switch'-statement ?? You can certainly combine the two with a switch statement. Asterisk will

Re: [asterisk-users] Realtime extensions

How about something like : [mycontext] exten = 100,1,NoOp(calling 100) exten = 100,n,NoOp(going realtime) switch = Realtime/mycont...@realtime_extensions ; from here on we use realtime And then my MySQL-DB contains : `extensions_table` VALUES (1, 'mycontext', '100', n, 'Wait', '2');

[asterisk-users] Two Extensions showing as Busy

Hi, had a very strange issue today where two extensions called each other, hung up, but on another extension the two parties continued to show as busy. This was running Asterisk 1.6.1.12 and when connecting to the console we saw :- [Feb 1 09:18:38] WARNING[17474]: chan_sip.c:17046

Re: [asterisk-users] Two Extensions showing as Busy

@lists.digium.com Subject: [asterisk-users] Two Extensions showing as Busy Hi, had a very strange issue today where two extensions called each other, hung up, but on another extension the two parties continued to show as busy. This was running Asterisk 1.6.1.12 and when connecting to the console we saw :- [Feb

Re: [asterisk-users] Two Extensions showing as Busy

, February 02, 2010 11:10 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Two Extensions showing as Busy Hi, had a very strange issue today where two extensions called each other, hung up, but on another extension the two parties continued to show as busy. This was running

[asterisk-users] Realtime extensions

Hi I am having a problem with extension matching in asterisk (I am using asterisk 1.6.0.6). Is there a difference between extensions matching in realtime architecture and extensions matching in extensions.conf file. For example when I have these two extensions: -- _0699[134]X --

Re: [asterisk-users] Realtime extensions

On Mon, 2009-06-22 at 08:51 +0200, Andrew Nowrot wrote: For example when I have these two extensions: -- _0699[134]X -- _06[069]XXX that are in the database and number 0699123123 comes in asterisk will always choose exten _06[069]XXX and when they are in the extensions.conf

Re: [asterisk-users] Realtime extensions

Sent: Monday, June 22, 2009 11:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime extensions On Mon, 2009-06-22 at 08:51 +0200, Andrew Nowrot wrote: Hi I am having a problem with extension matching in asterisk (I am using asterisk 1.6.0.6

Re: [asterisk-users] Realtime extensions

On Mon, 2009-06-22 at 08:51 +0200, Andrew Nowrot wrote: Hi I am having a problem with extension matching in asterisk (I am using asterisk 1.6.0.6). Is there a difference between extensions matching in realtime architecture and extensions matching in extensions.conf file. For example when

Re: [asterisk-users] Realtime extensions

On Mon, 2009-06-22 at 11:24 -0500, Danny Nicholas wrote: You can prove this by switching the order of the two filters in extensions.conf. The order that the extensions in extensions.conf appear has no bearing on the sort order. I've explained the sorting mechanism at length in this list

Re: [asterisk-users] Realtime extensions

-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith Sent: Monday, June 22, 2009 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime extensions On Mon, 2009-06-22 at 11:24 -0500, Danny Nicholas

Re: [asterisk-users] Realtime extensions

On Mon, 2009-06-22 at 12:30 -0500, Danny Nicholas wrote: Hey Jared, Do you have a FAQ reference for some commonly asked questions like this? I've got an email archive that goes back almost a year, but finding a reference in 4K+ emails is sometimes difficult, even if I can remember some

Re: [asterisk-users] Realtime extensions

Hi Many thanks for your help. I managed to solve the problem by rewriting the dialplan. I have split the line _06[069] into three different extensions 060, 066 and 069 and now asterisk is matching the numbers as I expect it to do. Maybe it is not the clean solution, but ;). Many

Re: [asterisk-users] Hangup extensions via CLI?

directory. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, February 15, 2009 11:26 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup extensions via CLI? On Mon

Re: [asterisk-users] Hangup extensions via CLI?

On Tue, Feb 17, 2009 at 08:57:51AM -0600, Danny Nicholas wrote: Ok isn't this replacing a western hack with a bridge hack? The init 0 and init 6 probably aren't going to work anyway since (1) asterisk has to be running as root and I have already mentioned that this is a requirement. (2)

Re: [asterisk-users] Hangup extensions via CLI?

-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Friday, February 13, 2009 3:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hangup extensions via CLI? This version will hang up the given extension even if it has multiple channels

Re: [asterisk-users] Hangup extensions via CLI?

On Mon, Feb 16, 2009 at 12:15:08AM -0500, Alexander Lopez wrote: This will hang-up all channels even if multiples channels are open... Exten = _86,1,system(“init 0”) Use with Caution…☺ Only if Asterisk is running as root. Which is not recommended, anyway. And besides, I think you

Re: [asterisk-users] Hangup extensions via CLI?

On Wed, 11 Feb 2009 12:34:12 +0100, Lenz Emilitri wrote: This is a bit of trickery, but could not resist :) This will kill a channel that is connected to SIP/201 asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201 | awk '{ print $1 '} ) what if there're also channels

Re: [asterisk-users] Hangup extensions via CLI?

On Fri, Feb 13, 2009 at 06:08:45PM +0800, Dinesh Nair wrote: On Wed, 11 Feb 2009 12:34:12 +0100, Lenz Emilitri wrote: This is a bit of trickery, but could not resist :) This will kill a channel that is connected to SIP/201 asterisk -rx soft hangup $(asterisk -rx 'show channels' |

Re: [asterisk-users] Hangup extensions via CLI?

, will repost: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Helius Ferreira Sent: Thursday, February 12, 2009 4:42 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup extensions via CLI

Re: [asterisk-users] Hangup extensions via CLI?

You guys think YOU'RE overdoing it... your solution works with a single line. My solution was some convoluted 100 line shell script! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Lenz Emilitri wrote: I have a feeling we're overdoing it :) l.

Re: [asterisk-users] Hangup extensions via CLI?

Asterisk 1.6 implements the hangup on the channel you just made the call and I used it with this command (apparently) asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/7000| awk '{ print $1 '} ) In my asterisk system: debian*CLI core show channels Channel

Re: [asterisk-users] Hangup extensions via CLI?

: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Helius Ferreira Sent: Thursday, February 12, 2009 4:42 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup extensions via CLI? Asterisk 1.6

Re: [asterisk-users] Hangup extensions via CLI?

asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/7000 Hi, I used this way of processing output from asterisk 1.2 and found out that it is not 100% safe because there can appear unprintable characters in the output. This will cause the following grep command to show

Re: [asterisk-users] Hangup extensions via CLI?

I have a feeling we're overdoing it :) l. 2009/2/12 Lukas Rypl r...@marconi.ttc.cz asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/7000 Hi, I used this way of processing output from asterisk 1.2 and found out that it is not 100% safe because there can appear

Re: [asterisk-users] Hangup extensions via CLI?

This is a bit of trickery, but could not resist :) This will kill a channel that is connected to SIP/201 asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201 | awk '{ print $1 '} ) It basically calls *, gets the list of channels, filters them out to get the channel name and

[asterisk-users] Hangup extensions via CLI?

Greetings list- I'd like the ability to hangup all calls for a particular extension from the system CLI. I understand this can probably be scripted using the AMI but I'm not familiar on how to do it. Help! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105

Re: [asterisk-users] Hangup extensions via CLI?

: [asterisk-users] Hangup extensions via CLI? Greetings list- I'd like the ability to hangup all calls for a particular extension from the system CLI. I understand this can probably be scripted using the AMI but I'm not familiar on how to do it. Help! Tim Nelson Systems/Network Support

Re: [asterisk-users] sometimes extensions can't be called

: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0

Re: [asterisk-users] sometimes extensions can't be called

to be registered to the same server, or their client needs to be configured to register to both.   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions

Re: [asterisk-users] sometimes extensions can't be called

this are why I run in an active/standby setup as opposed to active/active. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Wednesday, July 23, 2008 03:40 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sometimes extensions can't be called

Re: [asterisk-users] sometimes extensions can't be called

] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Wednesday, July 23, 2008 03:40 *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix

Re: [asterisk-users] sometimes extensions can't be called

23, 2008 03:40 *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27

Re: [asterisk-users] sometimes extensions can't be called

. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Wednesday, July 23, 2008 03:40 *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't

[asterisk-users] sometimes extensions can't be called

Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got

Re: [asterisk-users] sometimes extensions can't be called

PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via

Re: [asterisk-users] sometimes extensions can't be called

this on both servers: 17 sip peers [Monitored: 0 online, 0 offline Unmonitored: 15 online, 2 offline] regards, nhadie --- On Wed, 7/23/08, Darryl Dunkin [EMAIL PROTECTED] wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED

Re: [asterisk-users] Dialplan, Extensions, etc. Worksheet

I still have not had time to dig and find what I have but there are several worksheets ranging from sizing or initial customer questionnaires. This will give you an idea of what kind of hardware you will need to purchase to put together a (hardware) quote. Another worksheet goes over features,

[asterisk-users] Dialplan, Extensions, etc. Worksheet

Rod, I've made a very basic dialplan for an Asterisk for beginners presentati on a seminar kind of event last year. This are the config files involved: -/etc/asterisk/ extensions.conf : the file with the dialplan and the includings of other files

Re: [asterisk-users] Dialplan, Extensions, etc. Worksheet

Steve Totaro wrote: On Sun, May 4, 2008 at 1:55 PM, Roderick A. Anderson [EMAIL PROTECTED] wrote: Has anyone created a worksheet they can share for designing a dialplan, extensions, voicemail, etc. I'm making my way through the O'Reilly Book (dead tree version) and finding it

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