similar to: Re: Codec Issue on IAX trunk? (Solved)

Hi Rich - > Those type changes to iax.conf require a full stop of > asterisk followed by a cold asterisk startup. A restart > from the CLI won't cut it. Ahh! That's a very important piece of information! > Were you previously doing the CLI restart? I did lots a CLI "reloads", and few cold restarts to the ast33 machine, but no cold restarts on the ast551

Hi All - Well, after happily existing in a one office environment with asterisk for a few months, I've now decided to start adding in our other offices with their own * boxes and IAX connections (over VPN). Unfortunately, I'm an idiot and I can't get it to work. I'm having some kind of problem with codecs, I guess, but I don't understand what or why. When trying to use

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) --

I seem to be having a problem that I have narrowed down to a disagreement on codec negotiation or codec setup of some kind in an IAX peering arrangement. Here's a non-ASCII art version of the setup: DID origination provider via SIP/gsm to Call routing asterisk server via IAX/gsm to Client asterisk server via SIP/ulaw to Polycom 501 UA The problem that occurs

Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW i'm working

Hi there, I had a problem, basically, I have 4 different types of end users (gsm, ilbc, g729, ulaw). However, I only have one user with my DID provider. My provider supports all 4 codecs. The issue is then: When an incoming call comes in, a codec is negotiated (usually ULAW), later on, when the extension is dialed, we'll see we're doing GSM, and thus transcode. Here's an example

I didn't know how else to caption this. I'm trying to play around with codec pass-through. I have two SIP phones, both with g729, behind two Asterisk servers. I set all the configs, SIP and IAX, to "disallow=all; allow=g729" on both servers. But the originating server won't even try to call the destination server: -- Executing Dial("SIP/btel-c7d7",

Hi, I have strange problem. I have 1 SIP client (bt100) and 1 Iax2 client (IAXPhone): - when I call from Iax to SIP sound works - when I call from Sip to Iax sound doesn't work, I get : Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm

Hi I've two servers : 1. UK 2. Pakistan Pakistan * server has ISDN30. Pakistan(ISDN30) <====> UK ===> User Im planning to setup an IAX2 trunk between these two server ? so , how much bandwidth I need for 30 simul. calls ? Im planning to use G729 on both my server ? to support 30 calls over IAX2 do I've to change some setting during compile time or not ? pls suggest.

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

I am having some problems getting TelIax service to work with *. Outbound calls work just fine. When I try an inbound call the phone rings and there is no audio. Upon further investigation "iax2 show channels" indicates that the codec is "unknown" The provider confirmed that they are set for ulaw and so am I. Does anyone have an idea what could be causing the codecs to

Hello, I am new to Asterisk; I did go through a lot of documentation, wikis, and the O'Reilly book and have most of what I need now working well. I do have a problem that I keep bumping heads against, however: I can dial out very well through a IAX2 trunk (9 followed by number), but if I specify the same IAX2 trunk in the Channel of a .call file, the call does not go through. Here is my

Hi - I'm running CVS-HEAD from 2005-08-11 20:17:17 UTC, and I'm trying to set up some redundancy on IAX connections between locations. I have two IAX peers set up that work correctly by themselves: "ast551-out" and "ast551-out-backup": [ast551-out] type=peer secret=secret username=ast551 host=X.X.X.X qualify=1000 disallow=all allow=gsm allow=ulaw trunk=no

I'm testing a few IAX trunk scenarios in a controlled lab. From server2 extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial 6099 it also plays tt-weasels as it's supposed to, but it's not the tt-weasels under its

Dale, Sorry for taking so long to answer, I've been traveling. Thanks so much for the suggestion, your solution worked perfectly. I'm not sure why I didn't notice that the IAX trunk was working in the other direction. Once again, thanks for your help. Mitch Date: Mon, 25 Jun 2012 05:44:37 -0500 From: Dale Noll <dnoll at wi.rr.com> Subject: Re: [asterisk-users] IAX Trunk

    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

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> Oh is Gecko == Mozilla? Oh well, can''t keep them straight. As far as I''m > concerned there are only 2 browsers worth developing for, IE6 and FF > > 1.07. Remove IE 6 and I agree, hahahaha. Actually, I hate Mozilla''s box model (I know it''s the w3c recommended one, but it''s flawed. Width and height should include padding and borders.