similar to: G.729 Pass-Thru & Voicemail

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download

List, I have some cisco phones (7940) and asterisk 1.4 running nicely.. Communication between the phones is G.729, and my sip.conf looks like this: disallow=all ; First disallow all codecs allow=g729 ; allow=gsm allow=ulaw allow=alaw However, I cannot call voicemail - I get the following error: [Apr 20 14:58:31] WARNING[87184]: channel.c:2816

Hi list! My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a voicemail. On two of these numbers the voicemail works without any problem, on the other it doesn't... I get this error: [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) [Oct 17 17:01:29] WARNING[14700]: file.c:957

Justin Killen wrote: <snip> > > Whenever I try to copy this callfile into /var/spool/asterisk/outgoing/ > I get these 3 lines repeating over and over (I?m not 100% sure which > entry is first): > > [2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353 > set_format: Unable to find a codec translation path from (nothing) to (slin) > > [2015-02-16

Hi, I copied a setup from an older 1.8.5 installation to an 11.15 installation, and I'm having problems getting call files to work. Here is the extension setup I'm using: [outbound-swift] exten => _[a-zA-Z].,1,Answer exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure) ;exten => _[a-zA-Z].,1,Swift("${EXTEN}") exten => _[a-zA-Z].,n,Goto(1) [mis-phone] exten =>

Joshua, If I'm understanding this correctly, you're saying that the Playback is failing because it isn't connected to anything on the other end, because the Dial() failed. When the channel is created on the "OutgoingSpoolFailed" extension, what context is it created in, one of the origin legs? Is there a way detect this condition in the target context ([outbound-swift]),

Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark

On 2017-04-18 08:17 PM, Pete Mundy wrote: >> On 19/04/2017, at 7:58 am, D'Arcy Cain <darcy at VybeNetworks.com >> <mailto:darcy at VybeNetworks.com>> wrote: >> >> <snip> >> Everything looks the same as another one that works except for two >> things. The one that works doesn't have the "Probation passed" lines. >> I am

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

Hi All, I have this codec problem, I use "gsm" in my iax.conf file and in teliax settings also, but the error is still appearing as below. please help me. Kumara Starting simple switch on 'Zap/1-1' -- Executing Dial("Zap/1-1","IAX2/kumara@teliax/01194777070239|30|tr") in new stack -- Called kumara@teliax/01194777070239 -- Call accepted by

Hi List. Don't know if I already posted about this problem but, if I have I apologize for the double post. I am trying to test a time of day extension dialing 80, all I'm trying to test is if is morning I would like asterisk to say "Good Morning" but, when I run the test I get the following error message saying that the file doesn't exist and it does: Night..............

Hello. I have a question regarind sound files in asterisk 1.6. I have a sound package in ulaw format and I would like to know if I have a sip extension with allow=alaw would asterisk convert that file to the codec the user is allowed to? I am having a problem playing a file that exist in /var/lib/asterisk/sounds/es/good.ulaw but asterisk is telling me it doesn't. Here's what I get when

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

Hi everyone, I'm having an odd issue. I've been doing some testing over the past couple weeks on some Asterisk modules / utilities, but have bumped into a problem which I can't seem to resolve. Asterisk can't seem to play the default sound files (ULAW) in my environment. All necessary debugging information is included below. I'd love to get anyone else's thoughts on this,

Hi! I have a strange problem. In an AGI I tell Asterisk to playback a number, for example 31. I then use the AGI SAY NUMBER command and I only hear "thirty" and then get: -- Playing 'digits/30' (language 'de') Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable

i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug

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 =

part of extensions.conf: exten => 11,1,Answer() exten => 11,n,NoOp(CallerID : ${CALLERID(all)}) exten => 11,n,Playback(/tmp/welkom-tcs.alaw) exten => 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1) ; wordt doorgerouteerd naar context open, maar indien gesloten : exten => 11,n,NoOp(Oproep tijdens winkel gesloten) exten => 11,n,Playback(/tmp/winkel-gesloten.alaw) exten =>

Greetings! to all Now, with some hard time and help from many genurous people's in the list, I have come to this point with my TDM20B card & my teliax's IAX2 account. I hope someone may help me with this issue mentioned below. I have already selected my codec as gms in my iax.conf as well as in teliax's "my account page" but still i have the same error when I attempt

I've got the call cuing all setup and working, but im trying to get the Callswaiting,you are caller #, etc, and its not working. I have the following inthere as stated: queue-youarenext = "queue-youarenext" ("You are now first in line.") queue-thereare = "queue-thereare" ("There are") queue-callswaiting = "queue-callswaiting" ("calls