similar to: Odd internal vs. External dialplan issue

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

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

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

Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723

Greetings everyone, I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream. In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information. I have noticed that when I do a MULTICAST page and send data

Mar 15 12:54:01 NOTICE[24269] channel.c: Dropping incompatible voice frame on Local/[removed number]@context-5c3e,2 of format ulaw since our native format has changed to slin Can anyone provide an English translation of what this means? The extension is a Polycom IP 501 The only allowed formats are g.711u MOH is MP3 files (obvious) All prompts have been re-recorded in .ul uLaw

Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP

After an upgrade to Asterisk 12, I'm "collecting" channels. When I enter and then exit a conference room, I see: -- <CBAnn/207-0000067f;1> Playing 'confbridge-leave.slin' (language 'en') -- Channel CBAnn/207-0000067f;2 joined 'softmix' base-bridge <5edb1920-3774-4ba3-8c4d-23e8fd04519c> -- Channel CBAnn/207-0000067f;2 left

Hello! Client is using ulaw, however server sometimes fills the log with following: [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

Hello list, I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom or Asterisk that makes the trouble. I'm playing a prompt, then make a choice for "2" : [Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] -- <SIP/test1-00000701> Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (language 'nl') [Jan

This is a serious problem! I have brought up this issue in four previous attempts to get some feedback. I find it hard to believe that no one else is having this same problem. Apr 11 13:27:36 NOTICE[4446]: channel.c:1906 ast_read: Dropping incompatible voice frame on Local/103@sip-00f3,2 of format alaw since our native format has changed to slin Apr 11 13:27:36 NOTICE[4446]: channel.c:1906

I'm doing some final check-outs before upgrading from 1.4.x to 1.6.x and I've encountered a problem playing back a .wav file to an Ekiga client: My dialplan looks like: exten => 730,1,answer exten => 730,n,playback(/home/phones/common/moh/moha/Sovereign) exten => 730,n,hangup Sovereign.wav is a .wav file that plays nicely on my 1.4 server. Here is what the console displays:

Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten =

Hi list, I'm trying to test an IVR system with recorded prompts and would like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234 ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3 is slin; Need it the other way so I can do DAHDI--> IAX testing. Any ideas? Google wasn't really helpful on this one. Danny Nicholas

I upgraded my server from Fedora Core 4 to Fedora Core 5. I was wondering if anybody else has run into the problem and know's the fix? I recompiled asterisk and if I don't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] => (Sound File

ive installed asterisk but below is what am getting proces gets killed.please help [root at localhost sounds]# asterisk -vvvvc Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under

Hi Sir, We have integrated Skype with Asterisk (skype user id:- rexesbposolutions). Each call which is coming to skype account is getting transfered to Asterisk Queue. It has following two cases: case 1: When we call from normal skype account to skype account (rexesbposolutions), everything is working fine. case 2: This skype account (rexesbposolutions) has been assigned with a online virtual