similar to: Can ShanSpy be used on Local Channels?

    Some users have complained that their calls drop after about 30 seconds.  Not all, just some.  After looking at the log files the only difference I can find from the dropped calls is the following line: [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge 14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge technology to native_rtp     Most calls just do:

I have tried both by hand and hitting tab to auto complete: *CLI> channel request hangup Message/ast_msg_queue Message/ast_msg_queue is not a known channel On 31/10/19 14:18, Sean Bright wrote: > On 10/31/2019 2:13 PM, Carlos Chavez wrote: >> I assume this is something created by Freepbx.  If I do a "channel >> request hangup" it tells me the channel does not exist.

    Is there a solution to dial multiple contacts for a Queue agent?  Since the pandemic started many of our customers have begun to move agents off site.  Since most of them were using softphones we did not have any problems but now we have one case where the agents have a desk phone in their cubicle and are using a softphone from home.  For regular calls there is no problem as

Hi Olivior, We have recently worked on a WebRTC based agent panel. As based on my experience I think that WebRTC based phones are far better and cheaper then those soft / sip phone. the big plus is that they are easy to customize and developer can use the power of browser and web to build / offer features which are not possible with regular phones. Regarding your concern about BLF or call

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

    I am trying to troubleshoot two Asterisk servers that have an IAX2 trunk between them.  Calls come and go but there is no CallerID from the remote server either way.  One of the servers is running Asterisk 16 and the other is an older 1.8 install (I know, I am trying to get permission to update).  The trunk between servers is very simple.  Something like: Server 1 (Mexico) [panama]

    Not these particular two servers. On 02/03/20 12:16, Doug Lytle wrote: >>>> I am trying to troubleshoot two Asterisk servers that have an IAX2 >>>> trunk between them. > Carlos, > > Had caller-id ever worked between these two systems? > > Doug > -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161

    Since yesterday I have a stuck channel on my Asterisk server and I do not know how to eliminate it: Message/ast_msg_queu macro-dial-one       s                  59 Up      Dial         PJSIP/1218/sip:1218 at 192.1 17:24:07     I assume this is something created by Freepbx.  If I do a "channel request hangup" it tells me the channel does not exist. Any ideas? --

    Dis you set "outbound_auth" in your endpoint configuration to Twilio? On 21/06/23 11:19, TTT wrote: > I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP > (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: > > [Twilio] > type=auth > auth_type=userpass > password=mysecret > username=myun > > However, my calls

You need to put your external IP in the transport configuration: external_media_address=X.X.X.X external_signaling_address=X.X.X.X external_signaling_port=5060 On 21/06/23 12:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction:

    I am having a strange problem.  We have an Asterisk 16.12.0 server (we have upgraded at least two versions since we found the problem) where users complain that confbridge calls end after about 30 minutes or so.  The problem is that according to Asterisk the calls are still active.  All users are cut off at the same time but a "core show channels verbose" still shows channels as

On 1/14/19 4:02 PM, Duncan Turnbull wrote: > > > Sent from my iPad > > On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org > <mailto:TPeters at mcts.org>> 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

On 7/19/17 2:37 AM, Marcelo Terres wrote: > This is the pjsip library. > > Is it possible to you to update pjsip for the latest version to test > if it solves the problem? > > On 18 Jul 2017 3:52 pm, "Carlos Chavez" <cursor at telecomab.mx > <mailto:cursor at telecomab.mx>> wrote: > > I am getting frequent segfaults on a new Asterisk

    I am having a strange problem with a new provider.  We already have a couple SIP trunks working fine.  We are trying a new provider but we are having one way audio problems with outgoing calls.  Incoming calls do have two way audio, only outgoing calls have this problem.  I do not see anything odd with a packet capture and using PJSIP history to check.  The provider says that on outgoing

    I am switching many of my scripts to python and I found pyst2 in my search for an Asterisk library.  While it seems to work well for AGI acripts it seems very broken when using it for AMI.  Can anyone recommend a good and currently supported AMI library for python? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161

I am getting frequent segfaults on a new Asterisk installation. So far the only message I see is: Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 ip 00007fb2d535723f sp 00007fb25a11b5c0 error 4 in libasteriskpj.so.2[7fb2d52e5000+180000] Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault at 188 ip 00007f4afea0c23f sp 00007f4a7f7e35c0 error 4 in

On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx> wrote: > > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and

On 7/20/17 8:47 AM, Marcelo Terres wrote: > Which version of Asterisk are you using? Are you compiling it with the > bundle pjproject ? > > --with-pjproject-bundled > > Regards, > > Marcelo H. Terres <mhterres at gmail.com <mailto:mhterres at gmail.com>> > IM: mhterres at jabber.mundoopensource.com.br > <mailto:mhterres at

If SIP goes to the same provider then yes. Still I would check a packet capture for better understanding. BTW, did you try iax debug? ??, 20 ???. 2017 ?. ? 19:46, Carlos Chavez <cursor at telecomab.mx>: > On 4/20/17 12:45 AM, Kseniya Blashchuk wrote: > > Can it happen that the routes lead the traffic through another interface? > Did you try a packet capture with tcpdump? Do the

I want to try using google for speech recognition in Asterisk and I found a ready made AGI: http://zaf.github.io/asterisk-speech-recog/ I have followed all the steps listed in the web site but I keep getting this error: <PJSIP/2001-0000006e>AGI Tx >> 200 result=99981 (timeout) endpos=22720 <PJSIP/2001-0000006e>AGI Rx << VERBOSE "Unable to get recognition