[asterisk-users] call hangup after leaving app_queue

can you someone confirm https://issues.asterisk.org/jira/browse/ASTERISK-27065 its easy to replicate Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community

Re: [asterisk-users] Force hangup not working on stuck channel

Also, it looks like in https://issues.asterisk.org/jira/browse/ASTERISK-21762 there might be a workaround (see the last comment at the bottom). Matthew Fredrickson On Fri, Nov 4, 2016 at 2:01 PM, Matt Fredrickson wrote: > On Thu, Nov 3, 2016 at 11:16 AM, Carlos Chavez

Re: [asterisk-users] Force hangup not working on stuck channel

On Thu, Nov 3, 2016 at 11:16 AM, Carlos Chavez wrote: > I am unable to force a hangup on a channel that has been stuck for over two > days: > > IAX2/from-CD-11006 oficina 27701 Up Dial > IAX2/to-CD/2883 3467130007

Re: [asterisk-users] Force hangup not working on stuck channel

I always set a TIMEOUT(absolute) on calls across trunks to something reasonable like 10 hours, that way calls should end in a sane amount of time even if something weird happens. Otherwise I've always had to do a reload when I couldn't hang up from the CLI. On Thu, Nov 3, 2016 at 9:16 AM, Carlos

Re: [asterisk-users] Force hangup not working on stuck channel

Hi Carlos, Did you try with the following CLI command: CLI> channel request hangup CHANNEL_NAME ??? El nov. 3, 2016 1:16 PM, "Carlos Chavez" escribió: > I am unable to force a hangup on a channel that has been stuck for over > two days: > > IAX2/from-CD-11006

[asterisk-users] Force hangup not working on stuck channel

I am unable to force a hangup on a channel that has been stuck for over two days: IAX2/from-CD-11006 oficina 27701 Up Dial IAX2/to-CD/2883 3467130007 46:24:59 Sotelo Sotelo IAX2/to-CD-20713 I have tried "hangup request

[asterisk-users] Call hangup on transfer when originated from a Queue

Hi all, i'm writing because going crazy on this issue i'm unable to solve. My VoIP system is based on OpenSIPS router that forward calls to an Asterisk BOX to have IVR and Queue services. If a call was directed to a queue and operator answer, on transfer to another ext. the call hangup. On

[asterisk-users] Detect hangup due to RTP timeout

Hello, Is there a way to tell in the dialplan, Asterisk AGI, or Asterisk AMI, when a call has been hung up because the SIP rtptimeout has been reached? Thank you, -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019

Re: [asterisk-users] Can't hangup channel from CLI

On Fri, Aug 22, 2014 at 6:00 PM, Steve Edwards asterisk@sedwards.com wrote: Asterisk 11.11.0 on CentOS 6.5, installed from RPMs. OpenSIPS fronting Asterisk from a Tekelec T9000. I'm accumulating stuck channels. snip I haven't identified what callers are doing to reproduce the error

[asterisk-users] Can't hangup channel from CLI

Asterisk 11.11.0 on CentOS 6.5, installed from RPMs. OpenSIPS fronting Asterisk from a Tekelec T9000. I'm accumulating stuck channels. I'm googling now and I recognize that Friday afternoons are the worst time to ask questions, but I'm getting desperate because this is keeping me from

[asterisk-users] Dedicated hangup extension h

Hello, We have a Kamailio SIP Proxy in front of our Asterisk cluster for incoming calls from our carrier. The sip.conf looks like this: [kamailio1] type=friend host=10.0.0.1 context=incoming disallow=all allow=alaw All calls hit the incoming extension. In the extensions.conf we have multiple

Re: [asterisk-users] Dedicated hangup extension h

On 28 Aug 2013, at 09:50, Grant Bagdasarian g...@cm.nl wrote: Hi Grant! I do not know of a way to have multiple 'h' extensions in the same context. But you can easily make an appropriate context for your custom need! exten = _X.,1,Playback(invalid) exten = _X.,n,Hangup exten =

Re: [asterisk-users] Dedicated hangup extension h

10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dedicated hangup extension h On 28 Aug 2013, at 09:50, Grant Bagdasarian g...@cm.nl wrote: Hi Grant! I do not know of a way to have multiple 'h' extensions in the same context. But you can easily

Re: [asterisk-users] Dedicated hangup extension h

I would set a no-use flag in all extensions that you do not want to use the h, and then test for it in the h extension itself - if it is set you could just run the Hangup application. On 28 Aug 2013 08:51, Grant Bagdasarian g...@cm.nl wrote: Hello, ** ** We have a Kamailio SIP Proxy in

Re: [asterisk-users] Dedicated hangup extension h

...@lists.digium.com] On Behalf Of Grant Bagdasarian Sent: Wednesday, August 28, 2013 3:51 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dedicated hangup extension h Hello, We have a Kamailio SIP Proxy in front of our Asterisk cluster for incoming calls from our carrier

Re: [asterisk-users] Dedicated hangup extension h

...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant Bagdasarian Sent: Wednesday, August 28, 2013 3:51 AM To: asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Dedicated hangup extension h Hello

[asterisk-users] monitoring - hangup channel

How can I monitor channel that hangup? I'm using asterisk 1.8.15.1 and there are many times that nobody is using the line but when I run: asterisk -rx core show channels it show: Channel Location State Application(Data) SIP/pstn--00 (None)

Re: [asterisk-users] AGI HANGUP PROBLEM

Hello, I experience the same problem, and I would really appreciate if someone could give us a hint on that. Hoggins! Le 17/09/2012 19:22, Mehdi Rahimi a écrit : Hi all, I need to handle a problem from AGI please guide me in extensions_custom.conf : exten = s,1,Answer exten =

Re: [asterisk-users] AGI HANGUP PROBLEM

Hi, Just following this thread for few days, I've some basic troubleshooting questions for you. 1- What do you mean by calling from landline? How is your Landline /mobile reaching your asterisk box ? is there a Hardware card ! or a VoIP provider. 2- Enable SIP traces and keep an eye on the

Re: [asterisk-users] AGI HANGUP PROBLEM

ِDear Sammy, Thank you for your following , 1- Land line i mean telco company which is calling to my server , i use FXO VOIP CARD (ATCOM 4 port) and test on a gateway too. 2-please explain me more about Enable SIP traces and keep an eye on the originating BYE request Regards, Mehdi On Tue, Sep

Re: [asterisk-users] AGI HANGUP PROBLEM

In article caehsoweantztyoebdobjchoeszhfk_z9sigaujsij15xx-u...@mail.gmail.com, Mehdi Rahimi mrm.ci...@gmail.com wrote: Hi all, I need to handle a problem from AGI please guide me in extensions_custom.conf : exten = s,1,Answer exten = s,n,AGI(hang.php) exten = s,n,Hangup in

Re: [asterisk-users] AGI HANGUP PROBLEM

Hi, So basically the FXO cards configurations need to be tweaked i.e hanguponpolarityinverse=yes etc. Since this is a Hangup request initiated by the SIP client, Asterisk then atleast it should close all the media streams and channel should get deleted. Keeping an eye on BYE : *CLI sip set debug

Re: [asterisk-users] AGI HANGUP PROBLEM

In article cajujwtig7yzk4+kb3c6sdu6zhb_+vwsg-oy0pibw0maeeed...@mail.gmail.com, SamyGo govoi...@gmail.com wrote: So basically the FXO cards configurations need to be tweaked i.e hanguponpolarityinverse=yes etc. Since this is a Hangup request initiated by the SIP client, Asterisk then atleast

Re: [asterisk-users] AGI HANGUP PROBLEM

Hi Tony, Thank you for your attention , and appreciate your contribution . You are right we can not do anything till the caller hangup BUT how can we prevent to hearing DTMF when someone else is trying on another extension ? to clearance : someone calls (from landlines os mobile , no difference)

Re: [asterisk-users] AGI HANGUP PROBLEM

On Tuesday 18 September 2012, Mehdi Rahimi wrote: Hi Tony, Thank you for your attention , and appreciate your contribution . You are right we can not do anything till the caller hangup BUT how can we prevent to hearing DTMF when someone else is trying on another extension ? to clearance :

[asterisk-users] AGI HANGUP PROBLEM

Hi all, I need to handle a problem from AGI please guide me in extensions_custom.conf : exten = s,1,Answer exten = s,n,AGI(hang.php) exten = s,n,Hangup in hang.php : #!/usr/bin/php -q ? set_time_limit(30); require('phpagi.php'); error_reporting(E_ALL); $agi = new AGI();

[asterisk-users] $agi-hangup() Does not hang up the channel

Hello All, I need to use agi to handle some issue , after finishing agi i want to hang up the channel , if i call from an extension there is no problem but i want to be the same for PSTN (outside) caller , if someone call asterisk show the hang up channel but the caller is not disconnected and if

Re: [asterisk-users] $agi-hangup() Does not hang up the channel

On Monday 17 Sep 2012, Mehdi Rahimi wrote: I need to use agi to handle some issue , after finishing agi i want to hang up the channel , if i call from an extension there is no problem but i want to be the same for PSTN (outside) caller , if someone call asterisk show the hang up channel but

Re: [asterisk-users] $agi-hangup() Does not hang up the channel

Thank you for your reply i did it in both ways (AGI and DIALPLAN) but not working. so you mean it is because of telco ? what about digital lines such as E1 ? Regards, Mehdi On Mon, Sep 17, 2012 at 8:57 AM, Raj Mathur (राज माथुर) r...@linux-delhi.org wrote: On Monday 17 Sep 2012, Mehdi Rahimi

Re: [asterisk-users] $agi-hangup() Does not hang up the channel

This is happen whenever caller calls from mobile phone and if the caller calls from analog line i can handle with : ;exten = s,n,Playtones(congestion) ; send the audio sequence that humans understand means congestion ;exten = s,n,Congestion(5) ; signal the other end of congestion. Wait for

Re: [asterisk-users] $agi-hangup() Does not hang up the channel

On Monday 17 Sep 2012, Mehdi Rahimi wrote: Thank you for your reply i did it in both ways (AGI and DIALPLAN) but not working. so you mean it is because of telco ? what about digital lines such as E1 ? From my experience: the call gets disconnected if the called party executes HangUp on a

[asterisk-users] Asterisk HangUp not breaking incoming call for caller

Hi, Asterisk 1.8.13 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) When Asterisk executes HangUp() on an incoming call, the line remains connected for the caller. Zone = in, opermode = INDIA. Line set to fxsks. Any help on where to start

[asterisk-users] Asterisk HangUp not breaking incoming call for caller

Hi, Asterisk 1.8.13 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) When Asterisk executes HangUp() on an incoming call, the line remains connected for the caller. Zone = in, opermode = INDIA. Line set to fxsks. Any help on where to start

Re: [asterisk-users] Asterisk HangUp not breaking incoming call for caller

Raj, I am just confirming it happens here as well. CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1. Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) Loadzone = us The problem started manifesting itself after I switched to 1.8.x from 1.6.2.x Typical scenario: a caller apparently hangs up,

Re: [asterisk-users] Asterisk HangUp not breaking incoming call for caller

This has been happening since the asterisk 1.2 days, makes me believe it has something to do with Analog FXO ckts provided. Mitul Limbani On Sep 12, 2012 10:18 AM, Vladimir Mikhelson v...@mikhelson.com wrote: Raj, I am just confirming it happens here as well. CentOS 5.7. Asterisk 1.8.15.1.

Re: [asterisk-users] Asterisk HangUp not breaking incoming call for caller

On Wednesday 12 Sep 2012, Vladimir Mikhelson wrote: Raj, I am just confirming it happens here as well. CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1. Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) Loadzone = us The problem started manifesting itself after I switched to 1.8.x

Re: [asterisk-users] Asterisk HangUp not breaking incoming call for caller

Raj, Problem 1 is where asterisk times out and line gets free However for problem 2 There is no way that this line frees up, as it depends upon remote side infra of caller, if they calling from pri ckt they possibly could identify our hangup signal, but if they calling from Analog exchange this

Re: [asterisk-users] Asterisk HangUp not breaking incoming call for caller

On 9/12/2012 12:05 AM, Raj Mathur (??? ?) wrote: On Wednesday 12 Sep 2012, Vladimir Mikhelson wrote: Raj, I am just confirming it happens here as well. CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1. Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) Loadzone = us The problem

Re: [asterisk-users] FXS hangup issues

I currently have an Asterisk 1.8.8.1 system set up with SIP accounts as well as a Wildcard TDM400P REV I card with both FXS and FXO ports - FXO is connected to outside lines, FXS connected to inside analog phones. Everything about the setup works fine except one thing - after making calls to

[asterisk-users] FXS hangup issues

Greetings, I currently have an Asterisk 1.8.8.1 system set up with SIP accounts as well as a Wildcard TDM400P REV I card with both FXS and FXO ports - FXO is connected to outside lines, FXS connected to inside analog phones. Everything about the setup works fine except one thing - after making

[asterisk-users] Request hangup on local channel

show channels is giving me a line like: Local/call_out@playfile-981a,2!local-playfile!call_out!4!Up!MeetMe!PA0008|1qt!3175661010!!3!2!(None) I extract from that the following ID: Local/call_out@smvoice-local-public-address-playfile-981a I am trying to do the AMI Action: Hangup command (which I

Re: [asterisk-users] Request hangup on local channel

In article 4e9c3cbf.1070...@pagestation.com, Jerry Geis ge...@pagestation.com wrote: show channels is giving me a line like: Local/call_out@playfile-981a,2!local-playfile!call_out!4!Up!MeetMe!PA0008|1qt!3175661010!!3!2!(None) I extract from that the following ID:

Re: [asterisk-users] Request hangup on local channel

On 10/17/2011 10:33 AM, Jerry Geis wrote: show channels is giving me a line like: Local/call_out@playfile-981a,2!local-playfile!call_out!4!Up!MeetMe!PA0008|1qt!3175661010!!3!2!(None) I extract from that the following ID: Local/call_out@smvoice-local-public-address-playfile-981a I am

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On 06/01/2011 06:28 PM, Steve Davies wrote: On 1 June 2011 15:10, randall rand...@songshu.org wrote: On 06/01/2011 03:55 PM, randall wrote: On 06/01/2011 03:41 PM, Tzafrir Cohen wrote: On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote: Hi all, After running fine for a few months now

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On 06/01/2011 01:12 PM, Karsten Wemheuer wrote: Hi randall, Am Mittwoch, den 01.06.2011, 10:00 +0200 schrieb randall: i get the following errors: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 Your telco provider

Re: [asterisk-users] PRI hangup request, cause 18

satish patel wrote: We are getting hangup cause 18 http://networking.ringofsaturn.com/Routers/isdncausecodes.php *Cause No. 18 - no user responding.* This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within

Re: [asterisk-users] PRI hangup request, cause 18

Thanks for reply, But I'm able to call those number from my cell phone and othere pri. I'm only having this issue on 2 pri line rest are working ? -- Sent from my iPhone On Jun 8, 2011, at 5:44 AM, Doug Lytle supp...@drdos.info wrote: satish patel wrote: We are getting hangup cause 18

Re: [asterisk-users] PRI hangup request, cause 18

Ist the same operator connected to the pri-line? Perhaps another telco-operator can not connect to the desired destination - for whatever reason. Am 08.06.2011 12:55, schrieb Satish Patel: Thanks for reply, But I'm able to call those number from my cell phone and othere pri. I'm only having

Re: [asterisk-users] PRI hangup request, cause 18

Ist the same operator connected to the pri-line? Perhaps another telco-operator can not connect to the desired destination - for whatever reason. Am 08.06.2011 12:55, schrieb Satish Patel: Thanks for reply, But I'm able to call those number from my cell phone and othere pri. I'm only having

Re: [asterisk-users] PRI hangup request, cause 18

Date: Wed, 8 Jun 2011 15:41:04 +0200 From: t...@ovm-group.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI hangup request, cause 18 Ist the same operator connected to the pri-line? Perhaps another telco-operator can not connect to the desired destination

[asterisk-users] PRI hangup request, cause 18

We have 2 PRI from ATT And all is well but only few numbers having following issue. We are getting hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and this issue raised [Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got hangup request,

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On 06/01/2011 05:42 PM, Tzafrir Cohen wrote: On Wed, Jun 01, 2011 at 04:10:34PM +0200, randall wrote: On 06/01/2011 03:55 PM, randall wrote: On 06/01/2011 03:41 PM, Tzafrir Cohen wrote: On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote: Hi all, After running fine for a few months now

[asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

Hi all, After running fine for a few months now asterisk seems to hang frequently , still functioning but the DAHDI channels seem busy (users report a busy signal when calling or being called) A reboot will allow it to run for another day or maybe 2 or 3 till the problem occurs again.

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On Wed, Jun 1, 2011 at 11:36 AM, randall rand...@songshu.org wrote: Hi all, After running fine for a few months now asterisk seems to hang frequently , still functioning but the DAHDI channels seem busy (users report a busy signal when calling or being called) A reboot will allow it to

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On 06/01/2011 09:04 AM, mahesh katta wrote: On Wed, Jun 1, 2011 at 11:36 AM, randall rand...@songshu.org mailto:rand...@songshu.org wrote: Hi all, After running fine for a few months now asterisk seems to hang frequently , still functioning but the DAHDI channels seem busy

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On Wed, Jun 1, 2011 at 1:07 PM, randall rand...@songshu.org wrote: On 06/01/2011 09:04 AM, mahesh katta wrote: On Wed, Jun 1, 2011 at 11:36 AM, randall rand...@songshu.org mailto:rand...@songshu.org wrote: Hi all, After running fine for a few months now asterisk seems to

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

i get the following errors: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 Your telco provider has crc on or off , that is not matching with your server cross check with them. and this problem solve 4

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On Wed, Jun 1, 2011 at 1:30 PM, randall rand...@songshu.org wrote: i get the following errors: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 Your telco provider has crc on or off , that is not matching

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On 06/01/2011 10:07 AM, mahesh katta wrote: On Wed, Jun 1, 2011 at 1:30 PM, randall rand...@songshu.org mailto:rand...@songshu.org wrote: i get the following errors: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

Hi randall, Am Mittwoch, den 01.06.2011, 10:00 +0200 schrieb randall: i get the following errors: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 Your telco provider has crc on or off , that is not matching

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote: Hi all, After running fine for a few months now asterisk seems to hang frequently , still functioning but the DAHDI channels seem busy (users report a busy signal when calling or being called) A reboot will allow it to run for

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On 06/01/2011 03:41 PM, Tzafrir Cohen wrote: On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote: Hi all, After running fine for a few months now asterisk seems to hang frequently , still functioning but the DAHDI channels seem busy (users report a busy signal when calling or being

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On 06/01/2011 03:55 PM, randall wrote: On 06/01/2011 03:41 PM, Tzafrir Cohen wrote: On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote: Hi all, After running fine for a few months now asterisk seems to hang frequently , still functioning but the DAHDI channels seem busy (users report

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On Wed, Jun 01, 2011 at 04:10:34PM +0200, randall wrote: On 06/01/2011 03:55 PM, randall wrote: On 06/01/2011 03:41 PM, Tzafrir Cohen wrote: On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote: Hi all, After running fine for a few months now asterisk seems to hang frequently ,

Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

On 1 June 2011 15:10, randall rand...@songshu.org wrote: On 06/01/2011 03:55 PM, randall wrote: On 06/01/2011 03:41 PM, Tzafrir Cohen wrote: On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote: Hi all, After running fine for a few months now asterisk seems to hang frequently , still

[asterisk-users] Sip Hangup after critical packet

  HI   I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn)   Out going calls from asterisk to the ata works fine Incoming calls from the ata to asterisk cuts off with the error msg   Maximum retries exceeded on transmission 70854efe-4157e...@10.168.7.103 for seqno 102

[asterisk-users] Asterisk Hangup Issue in Ringing State with Incoming call

Hi, I have simulated “Chan phone” driver according to my own driver code and I am able to make internal and external [trunk] Asterisk calls. Only issue I am facing is with hangup in ringing state of incoming call. (1) Make a call from external X-lite to FXS and FXS is in ringing state now

Re: [asterisk-users] Asterisk hangup all outging calls after 32 seconds

Hi again... No ideas? El mar, 30-03-2010 a las 20:05 -0500, Ing CIP. Alejandro Celi Mariátegui escribió: (Sorry, but my english is not good) Hi, I have a problem with my new asterisk instalation. I search in google but I couldn't find nothing. Here's the thing. Before, we have 2

Re: [asterisk-users] Asterisk hangup all outging calls after32 seconds

Discussion Subject: Re: [asterisk-users] Asterisk hangup all outging calls after32 seconds Hi again... No ideas? El mar, 30-03-2010 a las 20:05 -0500, Ing CIP. Alejandro Celi Mariátegui escribió: (Sorry, but my english is not good) Hi, I have a problem with my new asterisk instalation. I

Re: [asterisk-users] Asterisk hangup all outging calls after 32 seconds

Have you modified extensions_custom.conf? From the bash command line try: grep TIMEOUT\(absolute\) /etc/asterisk/extensions_*.conf and post the results. Greg -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Asterisk hangup all outging calls after 32 seconds

(Sorry, but my english is not good) Hi, I have a problem with my new asterisk instalation. I search in google but I couldn't find nothing. Here's the thing. Before, we have 2 asterisk servers, each one with a E1 card. one with a Digium TE105 and the another with a A104 and we have a very

Re: [asterisk-users] Asterisk hangup all incoming calls after 10 seconds

Hi Bruno, I remember one of our customer had a similar problem with tellfree in Brazil. Their IT technician told me it was due to a g729 codec problem...they installed it and the problem disappeared. I never checked, I could only trust their man. Maybe it can help. Giorgio P.S.: let me know

Re: [asterisk-users] Asterisk hangup all incoming calls after 10 seconds

Hi Giorgio, So it means that Asterisk has no native support for g729 ? Thanks On Wed, Mar 17, 2010 at 7:04 AM, Giorgio Incantalupo gincantal...@fgasoftware.com wrote: Hi Bruno, I remember one of our customer had a similar problem with tellfree in Brazil. Their IT technician told me it was

Re: [asterisk-users] Asterisk hangup all incoming calls after 10 seconds

On Mar 17, 2010, at 1:05 PM, Bruno Camargo wrote: Hi Giorgio, So it means that Asterisk has no native support for g729 ? Thanks -- BrCaBadT -- Depends on your definition of support. It supports passthrough... but if you're using it locally on a bridge on transcoding, you'll need

Re: [asterisk-users] Asterisk hangup all incoming calls after 10 seconds

Hello Gentleman, I guess the problem was the codec. I have allowed only g711u for testing purposes and the incoming call endured for 1 minute, until the caller hanged. Thanks a lot for the support but there are tons of questions yet to be answered! Thanks On Wed, Mar 17, 2010 at 2:12 PM,

Re: [asterisk-users] Asterisk hangup all incoming calls after 10 seconds

Too early... call droped after 11 seconds now... different log. [Mar 17 22:19:17] DEBUG[2783] chan_sip.c: SIP TIMER: Rescheduling retransmission #13781 (6) SIP/2.0 - 1 [Mar 17 22:19:17] DEBUG[2783] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 100 ms (Retrans id

[asterisk-users] Asterisk hangup all incoming calls after 10 seconds

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very

[asterisk-users] AMD: HANGUP

*Code:* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Playback(Local/91441425477...@default-b9f2,1, sip-silence) in new stack -- Playing 'sip-silence' (language 'en') -- Executing AGI(Local/91441425477...@default-b9f2,1, agi:// 127.0.0.1:4577/call_log) in new stack

Re: [asterisk-users] AMD: HANGUP

It looks like your channel has been hungup during the AMD application, not that the AMD application is hanging up the call. The source is your friend (http://www.asterisk.org/doxygen/asterisk1.4/app__amd_8c.html): 00205 /* If we fail to read in a frame, that means they hung up */ 00206

Re: [asterisk-users] AMD: HANGUP

I changed my VOIP, and now things are ok. But didnt understand, how can VOIP can affect it ? On Wed, Feb 24, 2010 at 11:53 PM, David @ULC ucoms2...@gmail.com wrote: *Code:* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Playback(Local/91441425477...@default-b9f2,1,

[asterisk-users] queue hangup

hi there, How can we track that the calls within queue has been hang up or disposed within extension.conf ? I am trying to run agi script once the call within queue has been finished. Please advice. amir ___ -- Bandwidth and Colocation Provided by

[asterisk-users] DAHDI hangup detection

I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled from source on a Debian Lenny box, also running FreePBX 2.5.2. I also have an OpenVOX TDM400 card installed with an FXO port on port 1, and an FXS port on port 2. I have a POTS line installed and working on the FXO port.

Re: [asterisk-users] DAHDI hangup detection

. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Brown Sent: Tuesday, September 15, 2009 11:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI hangup detection I am running Asterisk

Re: [asterisk-users] DAHDI hangup detection

Of Stephen Brown Sent: Tuesday, September 15, 2009 11:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI hangup detection I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled from source on a Debian Lenny box, also running FreePBX 2.5.2. I also have an OpenVOX

Re: [asterisk-users] DAHDI hangup detection

-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Brown Jr Sent: Tuesday, September 15, 2009 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI hangup detection Ok on the workaround, how

Re: [asterisk-users] DAHDI hangup detection

Linetest.agi would run an AMI session to see if DAHDI-X was in use and return a variable as AVAIL or undef. If the line is in use, record voicemail, else hangup. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk

Re: [asterisk-users] DAHDI hangup detection

On Tue, 2009-09-15 at 11:23 -0500, Danny Nicholas wrote: The issue is that POTS as a technology does not have Answer/Hangup Supervision control (This is per the good folks at Digium). This is incorrect. Asterisk *does* support far-end disconnect supervision, if you're using Kewlstart

Re: [asterisk-users] DAHDI hangup detection

Following up on this with some testing I have done this afternoon. I have lengthened the default ringtime to 25 seconds from 15, and have tried both kewlstart and loopstart. Kewlstart appears to be the most functional, if I hang up before it hits app_voicemail, it appears to work as intended,

Re: [asterisk-users] Writing Hangup causes to CDR record

Hi guys, I'm trying to write hangup causes from asterisk into the CDR record. Using version 1.4.24.1 at the moment, but no joy so far. Has anyone implemented this? Neeraj Chand Support Analyst Fiji Islands Australia T: +6793342526 T: +61388924326

[asterisk-users] Numeric Hangup Code

I am sending SIP or H323 calls to a carrier, and I need to store in the CDR why the calls are rejected or why they hang up. In SIP, it can be code 503, 500, 488, etc. How do I get the information in my dialplan? I don't mean $(DIALSTATUS}, but the real numeric code F.Alves

[asterisk-users] API hangup command

If a call is established to a destination device Phone to other device (phone or something). Then I issue the monitor command to record the person speaking. Now I want to STOP the end device call --- BUT I want to continue to record the person speaking and sometime later deliver the

Re: [asterisk-users] API hangup command

To: asterisk-users@lists.digium.com Subject: [asterisk-users] API hangup command If a call is established to a destination device Phone to other device (phone or something). Then I issue the monitor command to record the person speaking. Now I want to STOP the end device call --- BUT I want

[asterisk-users] Caller Hangup detection

Hello, Is here any dial plan variable which could help me to identify that call was dropped (when still not connected) by caller? HANGUPCAUSE returns 0 DIALSTATUS returns NOANSWER How to identify such situation? Related question - how to know which end (caller or callee) ended the

[asterisk-users] Slow hangup - Australia - analog - incoming calls

I have had to install a TDM800 in a site, as the telco has held off installing ISDN indefinitely.. It's all fine except for the fact that it takes ages to hang up the line (6 or more rings), and sometimes doesn't even bother. This is only on incoming calls - outgoing calls work perfectly. Is

[asterisk-users] AGI Hangup

When I sent Hangup using AGI application, Asterisk always return -1: AGI Rx EXEC HANGUP -- AGI Script Executing Application: (HANGUP) Options: ((null)) AGI Tx 200 result=-1 But only send Bye ( Sip message ) when the script is fisnish, not when I send EXEC HANGUP. Why? Thx

[asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP

We have been using asterisk for a while now but have recently needed to install a second server in a remote office and set up a iax trunk between the 2 servers. The dial plan seems to work well when I tested it on the same LAN. However this afternoon I connected the system at the remote office and

Re: [asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP

In article [EMAIL PROTECTED], Nathan Dennis [EMAIL PROTECTED] wrote: We have been using asterisk for a while now but have recently needed to install a second server in a remote office and set up a iax trunk between the 2 servers. The dial plan seems to work well when I tested it on the same

Re: [asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP

this in error, please contact the sender and delete the material from any computer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Wednesday, 24 September 2008 8:03 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users

[asterisk-users] soft hangup (was: Re: (no subject))

Bill Andersen schrieb: V 1.4 When I do a show channels I get the following. CLI show channels Channel Location State Application(Data) SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 SIP/7110-afd286e0[EMAIL

[asterisk-users] Asterisk hangup not working on inbound calls

Hi, I have an Asterisk 1.2.18 box with a TDM400P card. If I make a call and then I hangup the phone, the call ends correctly but if I receive a call and I hangup the phone the other party does not get the hangup signal from Asterisk. Is there anybody who can explain this strange behaviour?

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