similar to: Using Codec G-726

Hi all, i'm trouble with codec setup on an asterisk machine 1.4.18 and some Thomson ST2030 as extensions. In the users.conf file for internal extension i have: disallow=all allow=g729 allow=alaw allow=ulaw Without any codec installed (i mean with original g729 of asterisk) all go fine, calling from an extension to one other: Peer User/ANR Call ID Seq (Tx/Rx) Format

Hi, I am trying to post this again as I am getting no answers and the support@digium.com bounces... (I have searched the whole list and can't find the answer either) I have installed a 5 user license for G.729 and want to route calls through Asterisk from my G.729 phone to Cisco AS5300 also using G729. Both Cisco and the phone connect using this codec if I do not force the call to go

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>

Hi, I purchased a G.729 1 channel codec license from digium. And installed as per the documentation. Then configured the sip.conf to use the new codec. For that, I am added the following entries in sip.conf (via web interface, as i am using asterisknow 1.5) disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm After that, when try to call through the PSTN line I can hear the voice of

Dear all, I am doing an interop testing with asterisk-1.6.0.5 now, and I have a question about the G.726 codec on asterisk. While my IAD supportes G.726-16,24,32 and 40 codecs, when doing a testing about G.726-40, I found that asterisk removed the G.72-40 sdp attrib when transmitting the INVITE with SDP. I modified sip.conf in order to solve the problem, G.726-32 is ok when allow=g726, but

I have 2 Asterisk servers connected with IAX. It's working fine I can call an extension from one phone in an office to another phone in the other office. The only problem I have is lagging. What codec should I use? I have Cisco phones 7960 on both end. Currently in the IAX trunk I configured it to disallow all and use GSM only. In my sip config of each phone I use disallow all and allow

I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the

The x86 and x86_64 Digium G.729 codec binaries have been updated for use with the Asterisk 1.4 beta (which should also work on current svn trunk). Anybody that is using the older modules with the 1.4 beta (or svn trunk newer than several days ago) is strongly encouraged to upgrade immediately, to avoid potential issues. -- Jason Parker Digium

The x86 and x86_64 Digium G.729 codec binaries have been updated for use with the Asterisk 1.4 beta (which should also work on current svn trunk). Anybody that is using the older modules with the 1.4 beta (or svn trunk newer than several days ago) is strongly encouraged to upgrade immediately, to avoid potential issues. -- Jason Parker Digium

Hi all, i have purchased the g.729 codec from digium. The registration was successful. (with the "old" binary) But there're a few questions: - should not the codec listed in the codec list when i enter "show codecs" ? - the codec is named with g729b but if i enter show codecs there is a codec g729a listed also the g729b is not installed. what is the difference between

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

Dear all, I have Trixbox 2.6 (Asterisk 1.4) installed in my voip server. I have the following short questions about the usage of G.729 codec: 1) Does Asterisk have installed the G.729 codec by default ??? 2) If I don't want to pay for a codec license, using Asterisk in "pass-through" mode for G.729 voice communications, do I just have to download the open source version of the G.729

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Hello, My problem is, outgoing calls (from asterisk to CCM) work fine but incoming (from CCM to Asterisk) does not work because of CCM is trying to use g729 over SIP trunk. I have found that link after a quick search. Problem is the same as in link below (However my Asterisk version is 1.8.13) and solution seems to have H323 trunk between CCM and Asterisk for using g729 codec. The post was

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

Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. I've got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I've set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows "(ULAW)" (G.711) as the codec in use. I'm a newbie at Asterisk, can anybody suggest what I might be

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems

Hi I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've allowed ulaw and g729. I want to change the codec for outbond calls. Please help not able to find anything using search. thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070501/af78de7a/attachment.htm

As of a few minutes ago, the Digium FTP servers at ftp.digium.com contain a new set of Linux X86-32 and X86-64 G.729 codec binaries, along with a new registration utility. The new codec binaries were produced using GCC 4.1, and are more highly optimized than the previous versions. In addition, there are now versions for both Asterisk 1.2 (and previous releases) and the soon-to-be-released