search for: telecomab

...> > IM: mhterres at jabber.mundoopensource.com.br > <mailto:mhterres at jabber.mundoopensource.com.br> > https://www.mundoopensource.com.br > https://twitter.com/mhterres > https://linkedin.com/in/marceloterres > > On 19 July 2017 at 17:03, Carlos Chavez <cursor at telecomab.mx > <mailto:cursor at telecomab.mx>> wrote: > > 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 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 installation. So > far the only message I see is: > > Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 > ip 00007fb2d535723f sp 00007fb...

...Multiple timing modules are loaded. You should only load one. [Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory [Feb 12 12:32:33] WARNING[22261]: chan_iax2.c:10939 build_user: Unable to support trunking on user 'telecomab' without DAHDI timing [Feb 12 12:32:33] WARNING[22261]: chan_iax2.c:10679 build_peer: Unable to support trunking on peer 'telecomab' without a timing interface I am using DAHDI 2.1.0.4, Asterisk 1.6.1-rc1 on a CentOS 5.2 machine with a TDM04 card. These are the modules: Module...

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.

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 packets really leave the > usb adapter? Can asymmetric routing be in effect? > Maybe there wer...

    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:

...aler software for ITSP <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns http://www.ictbroadcast.com/ On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez <cursor at telecomab.mx> wrote: > On 9/26/18 10:20 AM, Matthew Fredrickson wrote: > > > 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, > >>> > >>> T...

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 >...

...roadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> >> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns >> http://www.ictbroadcast.com/ >> >> >> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez <cursor at telecomab.mx> >> wrote: >> >>> On 9/26/18 10:20 AM, Matthew Fredrickson wrote: >>> >>> > 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: >>> >&g...

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

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

On 1/8/18 9:38 AM, Bertrand LUPART - Linkeo.com wrote: > Hello Carlos, > > >> We have a server that records all calls so we set Mixmonitor with the b option to only record calls that are actually bridged. I notice that we have lost of 44 byte files in /var/spool/asterisk/monitor which correspond to calls that were not answered. If a call is not answered I assume it was never

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 configuration. Also, Softphones must be carefully choosen if Deskphone-like quality is expected. Now that WebRTC becomes ubiquitous, it might make sense to trade Softphone features (call history, BLF, ...) for

...r/include/features.h:381:4: warning: #warning _FORTIFY_SOURCE requires compiling with optimization (-O) [-Wcpp]. Anyway, these problems do not happen if you manually build with the simple configure and make commands. Cheers, Patrick Wakano On Fri, 18 Oct 2019 at 11:54, Carlos Chavez <cursor at telecomab.mx> wrote: > They only problem I have found so far is while trying to install > Alembic for SQLAlchemy (for realtime configs). Those are the only packages > that I cannot get working properly. Vanilla Asterisk works fine with the > only extra package needed being libedit-deve...

On Mon, Sep 7, 2020 at 9:35 PM Carlos Chavez <cursor at telecomab.mx> wrote: > 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] br...

    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

    Does anyone know if Chanspy can be used with local channels? Since agents on queues need to use local channels like Local/XXXX at from-queue/n to make sure that all of their registered extensions ring we are now having a problem trying to use ChanSpy to listen to calls.  Since we do not know if the agent is on their desk phone or a softphone (which use different identifiers) we cannot set

I am having a very tough time trying to replace an Elastix 2.X install running as a virtual machine on ESXI 4. I tried using the Freepbx 14 ISO that installs CentOS 6 along with Asterisk 13.16 but I keep getting random segfaults: [175711.476685] asterisk[2942]: segfault at 188 ip 00007fc6c41abffc sp 00007fc608575890 error 4 in libasteriskpj.so.2[7fc6c4144000+14c000] I then proceeded

At the current time, we do not recommend attempting to build Asterisk on CentOS 8. Many packages Asterisk uses are not yet available and would require building from their sources. The Asterisk packages are also not available in the EPEL 8 or CentOS 8 repositories yet for the same reason. We'll update you when we think it's safe. -- *George Joseph* Digium - A Sangoma Company |

Can it happen that the routes lead the traffic through another interface? Did you try a packet capture with tcpdump? Do the packets really leave the usb adapter? Can asymmetric routing be in effect? Maybe there were some static routes that disappeared when the adapter was unplugged... On Thu, Apr 20, 2017, 12:41 AM Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > On