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Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

<br> [memberconnector]<br> ;<br> exten =3D&gt; _XXX,1,Dial(SIP/${peerPrefix}$<u></u>{EXTEN},${TIMERINGQUEUE}= ,)<br> =A0 =A0 =A0 same =3D&gt; n,NoOp(DIALSTATUS=3D${<u></u>DIALSTATUS})<br> <br> As you can see, all status are empty,<div class=3D""><div><br> <br> -- <br>

Since I passed from version 1.0 to the 1.2.3. I have Pb with the callerid. If somebody call with presentation of the number all is well. If somebody make call in masked number, i couldn't send a callerid to the phone. It is in a call center and i use the callerid to present the name of the number called to the operator. Before that went. To identify the sda, I use the assignment of the

Hi, I am looking for a hosted / virtual IPBX *PLATFORM* for service provider.Such hosted IPBX platform is aimed to be as a service, so that final customers don't have to install, maintain, or upgrade any PBX hardware or software. It should have a control panel for end users to create / edit extension, conference rooms , IVR menus, etc Could you please indicate companies that offer such

Hi I am using asterisk version 1.6.0.5 I have build up one utility that will fire Originate Action on Manager... In which, i have define number to call eg. 919912312345 (MobileNumber) How can i know that this number format is true for Indian Number... In originate action, user can enter any international number.. How can I came to know this number format is right for that country...?? IS there

I would like to ask, if someone observe also problem with peer qualify problems, my asterisk log is full with UNREACHABLE/REACHABLE messages, even when two asterisks are in LAN environment, please take a look into this debug, I can't find any problem with packet loss, all qualify requests are replied and acknowledged, I will submit bug report, if you will also not find any problems here...

Hi All, I am trying to get the opus codec working with linphone. I followed the instructions... This shows me its loaded core show translation paths opus --- Translation paths SRC Codec "opus" sample rate 48000 --- opus:48000 To g723:8000 : No Translation Path opus:48000 To ulaw:8000 : (opus at 48000)->(slin at 48000 )->(slin at

Hello people, I've ran into two problem that I can't seem to be able to solve on my own. Here's my scenario (running Asterisk 13.28.1): In short: - Asterisk behaves unexpectedly (at least to me) when negotiating between endpoints             that have a different but intersecting set of codecs (preventing direct media flow).           - Also, when an endpoint sends RTP with an

I''m trying to modify the SaltedHashLoginGenerator to where it separates the users table into two tables: a users table which contains login info: token, salt, etc) and userprofiles table (which contains firstname, lastname, authlevel, etc) I''m trying to use the existent helper for user under app/helpers/user_helper.rb and copied to app/helpers/userprofiles_helper.rb and

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

Just checking the transcoding on our Asterisk boxes and I get the following results. I have the g726, ilbc and lpc10 formats and codecs enabled in 'make menuselect' so I dont understand why its showing as no translation path. Any ideas? I am running certified-asterisk-11.2-cert2 Thanks Gareth > core show translation paths alaw --- Translation paths SRC Codec "alaw"

I installed a hardware g729 codec card in my asterisk, and I'm getting the following error when calling from a g729 sip extension to a SIP trunk also set to g729. The call goes through just fine, but these error messages keep flying by until I disconnect the call. Any ideas? ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

Hi. When I call my RNIS numbers (with a mobile phone for example), I can see 2 incoming calls on the IPBX, which should not happend. I'm not sure if it's a problem with the telco France Telecom and their ISDN setup, or if it's a problem with the MISDN driver on the IPBX itself. I'm stuck ... Any advices for troubleshooting that? Someone provide working configuration files

I have no speex translation ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24 slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name :

On Saturday 13 June 2020 at 17:23:14, Luca Bertoncello wrote: > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > > Try "sip show peer <peername>" for a phone. > bpi*CLI> sip show peer 0049177xxxxxxx > Codecs : > (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin| >

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

Again - when Originate is run from dialplan, i get: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin at 8000)->(slin at 192000) ReadTranscode: No When it's made with a call file (no matter how a call file is created), I see NativeFormats: (slin) WriteFormat: slin ReadFormat: slin WriteTranscode: No ReadTranscode: No Please