Re: [asterisk-users] change dialing process on live call

Thanks for your response, this works but we cannot hardcode this in the dialplan, we need this to be done from an external application connected either via manager or stasis. On Sun, Aug 19, 2018, 11:14 AM Doug Lytle wrote: > On 08/19/2018 05:57 AM, Khalil Khamlichi wrote: > > Is th

[asterisk-users] change dialing process on live call

Hi, Is there a way to add another extension to a live dial, for example Dial(PJSIP/1000,,) and after 20 secondes change it to Dial(PJSIP/1000/1001,,) I am open to suggestions such as using manager or stasis. Thanks in advance. Best regards, Kkh --

Re: [asterisk-users] MixMonitor and ChanSpy whisper

i am having same problem on Asterisk 13. MixMonitor does not record whisper or barge audio from ChanSpy. only call audio us recorded. hope someone can help. On Thu, Jul 5, 2018, 11:38 PM Patrick Wakano wrote: > Hello Asterisk list, > Hope you are all doing well! > > We are using the MixMonitor

Re: [asterisk-users] getting real sip status after dial

mporarily Unavailable". > > [caller_hangup] > exten => > s,1,Noop(HANGUPCAUSE(${MASTER_CHANNEL(out_chan)},tech)='${HANGUPCAUSE(${MASTER_CHANNEL(out_chan)},tech)}') > exten => n,Return > > > On 06/09/2018 03:10 PM, Khalil Khamlichi wrote: > > Thanks for your response Er

Re: [asterisk-users] getting real sip status after dial

,tech)}) > > Remember the incoming leg of the call and the outgoing leg of the call > are different channels. Make sure you are giving HANGUPCAUSE the > correct channel. > > On 06/09/2018 02:01 PM, Khalil Khamlichi wrote: > > It seems very weird to me that we cannot access sip co

Re: [asterisk-users] getting real sip status after dial

gt; are different channels. Make sure you are giving HANGUPCAUSE the > correct channel. > > On 06/09/2018 02:01 PM, Khalil Khamlichi wrote: > > It seems very weird to me that we cannot access sip code of a call > > from pjsip which information is actually returned from the provide

Re: [asterisk-users] getting real sip status after dial

It seems very weird to me that we cannot access sip code of a call from pjsip which information is actually returned from the provider, so it is available to asterisk, why does asterisk hide it ? On Sat, Jun 9, 2018 at 5:08 PM Khalil Khamlichi wrote: > > Hi, > > Is there any way I c

[asterisk-users] getting real sip status after dial

Hi, Is there any way I can get exact sip status from pjsip after a dial ? or all we can get is asterisk hangup causes ? Thanks in advance. KKh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

Re: [asterisk-users] Cancel

asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users On Fri, Jun 1, 2018, 11:58 AM Karen Stroebele wrote: > Please cancel my subscription > -- > _ > --

Re: [asterisk-users] multi step auth?

but that doesn't change the SIP header. > > Cheers, > > j > > On 05/08/2018 02:41 PM, Khalil Khamlichi wrote: > > try setting the callerid with > > same => n,Set(CALLERID(all)=17864089672 <17864089672>) > > ofcourse for each customer you will need to pro

Re: [asterisk-users] multi step auth?

try setting the callerid with same => n,Set(CALLERID(all)=17864089672 <17864089672>) ofcourse for each customer you will need to provide his own did. On Tue, May 8, 2018, 8:37 PM Jeff LaCoursiere wrote: > Hi, > > We have been using Voxbone for some time for origination,

Re: [asterisk-users] Which CDR processing for high load ?

what kind of ippbx does 50cps ? I think you must be looking for something like opensips. On Feb 22, 2018 9:13 PM, "Olivier" wrote: 2018-02-22 17:12 GMT+01:00 Dovid Bender : > Have you looked at the proc limits for the Asterisk PID? > > > No I haven't.

Re: [asterisk-users] PJSIP issue - Syntax error exception when parsing

This error is caused by the phone sending an erroneous sip header not by asterisk pjsip stack. solution would be to change the phone. On Wed, Feb 21, 2018 at 8:39 AM, Michele Pinassi wrote: > Hi all, i'm getting this error: > > [Feb 21 09:29:09] ERROR[1250]: pjproject:0

Re: [asterisk-users] username mismatch

try with username=pstn-1270 On Fri, Feb 16, 2018 at 5:50 PM, wrote: > When I have an incoming call I get a "username mismatch": > > WARNING[7459][C-0007]: chan_sip.c:16587 check_auth: username mismatch, > have <55>, digest has > NOTICE[7459][C-0007]:

Re: [asterisk-users] Call picked up from queue and transferred gets disconnected - about 0.01% of calls

maybe extension 3082 has some sort of network issue or maybe hardphone issue. test on another phone / network plug. On Feb 9, 2018 2:48 PM, "Stefan Viljoen" wrote: > Hi Guys > > > > I have an issue where a call is picked up from a queue. The caller asks > the person

Re: [asterisk-users] Running on virtual machine and audio not intelligable

why stick with version 11 ? upgrade to 13 is a starting solution. On Jan 25, 2018 3:28 AM, "Richard Mudgett" wrote: > > > On Wed, Jan 24, 2018 at 7:55 PM, Jerry Geis wrote: > >> >why load or even install dahdi if no cards are used? >> >> I thought

Re: [asterisk-users] Running on virtual machine and audio not intelligable

why load or even install dahdi if no cards are used? On Jan 24, 2018 10:30 PM, "Jerry Geis" wrote: > Hi All, > Running asterisk 11.25.3, the /proc/cpuinfo says > Intel(R) Xeon(R) CPU X5675 @ 3.07GHz > > lsmod | grep dahdi gives > dahdi_transcode16384 1

Re: [asterisk-users] remote Asterisk console

true, here is how to do it https://blog.russellbryant.net/2008/01/30/asterisk-16-features-tls-for-manager-ami-and-http/ On Tue, Jan 16, 2018 at 5:27 PM, Antony Stone wrote: > On Tuesday 16 January 2018 at 18:19:30, Paul Neuwirth wrote: > >> On Tue, 16 Jan

Re: [asterisk-users] remote Asterisk console

Hi, The easiest way would be to use asterisk manager interface (some simple steps to activate it on asterisk are easily found in the docs) https://wiki.asterisk.org/wiki/display/AST/AMI+Examples Now you will need a good python library to make it even easier

Re: [asterisk-users] asterisk queues in off-hook mode ?

thanks Richard, you just solved my problem. On Tue, Dec 19, 2017 at 9:05 PM, Richard Mudgett <rmudg...@digium.com> wrote: > > > On Tue, Dec 19, 2017 at 2:56 PM, Khalil Khamlichi > <khamlichi.kha...@gmail.com> wrote: >> >> Hi, >> >> I am looki

[asterisk-users] asterisk queues in off-hook mode ?

Hi, I am looking to configure asterisk queues in off-hook mode, that is, the agent calls into the system and stays connected to this call, when new customer calls, he is redirected to the queue which should distribute to connected agents. is this possible on teh actual app_queue or we would need