search for: g729

Hi all, I have a lot of call in wich I found that my Asterisk doesn't answer the BYE message, then the BYEs are retransmitted, but the call ends, when the Asterisk sends a BYE. Time AS.TE.RI.SK CA.RR.IE.R1 0 INVITE SDP ( g729 g711A g711U telephone-event) SIP From: sip:1265666072 at 81.209.186.14 <sip%3A1265666072 at 81.209.186.14>To:sip:1234567890 at CA.RR.IE.R1 (5060) ------------------>71(5060)1U telephone-event) 0.09 (5060) 100 Trying-------->71(5060)1U telephone-event) (5060) <------------------7...

...ot of "WARNING[20224]: udptl.c:819 ast_udptl_new_with_bindaddr: No UDPTL ports remaining" errors - is this related to number 2 above? Thanks, MD Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 172.16.116.29 2897516#15 1db77942648 00102/00000 g729 No Init: INVITE 172.16.116.29 2897516#13 585240b13ef 00102/00000 g729 No Init: INVITE 172.16.116.29 2897516#13 0244b8e668d 00102/00000 g729 No Init: INVITE 172.16.116.29 2897516#16 46d0960e602 00102/00000 g729 No I...

Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI> core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CL...

We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729. The problem is that I want to have my PSTN provider configured to allow ULAW as a first priorit...

I am trying to figure out the best way to deal with this. I want all of the calls in the network to be G729 and this is working. I do have hardware that provides me 30 g729 licenses. I am setting each extensions to disallow=all and allow=g729. However when I have this setup, I get no voice mail prompts. I tried setting to disallow=all and allow=g729,alaw and I still have no audio when calling voice mail....

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 has "disallow=all & allow=g729" - h323.conf has "disallow=all & allow=g729" The problem...

hi there .. I asked sometime ago regarding getting a Budgettone working with Asterisk over G729. My system is quite simple, Asterisk server with 1 G 729 license installed, and 10 Grandstream phones. Only one of them needs G729, because it's on a remote link via an ADSL bridge. The rest run happily on G711 on a local network. I added the lines disallow=all allow=g729 to the sip.conf en...

...thod. Here is the output from iax2 show channels: *CLI> iax2 show channels Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None) 111.111.111.111 xxxxxx 00003/00020 00001/00003 00070ms 0001ms 0071ms g729 (None) 111.111.111.111 xxxxxx 00004/00005 00004/00005 00070ms 0000ms 0070ms g729 (None) 111.111.111.111 zzzzzz 00005/00004 00002/00004 00000ms 0007ms 0071ms g729 (None) 111.111.111.111 xxxxxx 00007/00042 00001/...

I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow=ulaw allow=alaw allow=g729 There's the codec configs for the gateway and...

Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all and allow=ulaw. told asterisk to do a reload, and tried dialing out on a zap line. It was obvious from the call quality...

hello all, i have problem in using g729 codec. my asterisk version is 1.8.22. when i run "core show codecs" in asterisk, there is a g729 codec in the list so i assume that i can use it for my channels. but connection can not be set when i use it for my h323 channel. i read somewhere that codec g729 is a commercial codec and i...

I've purchsed 5 g729 licenses from digium, but am a little confused about why things are acting the way they are. When I do show g729 I see: 0/0 encoders/decoders of 5 licensed channels are currently in use My sip.conf starts out: [general] disallow=all allow=ulaw allow=g726 then my hosts look like this: [6016716]...

Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following...

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 Hold Last Message 10.9.9.9 202 11a6323d2ff 00102/00000 0x100 (g729)...

I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. "show translations" verifies that the registration took place. When I place a call, having "allow=g729" as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotia...

Hi users, I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file to G729 using file convert, but I am facing error as follows, file convert /tmp/welcome.gsm /tmp/welcome.g729 Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729! Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed. [Dec 20 17:24:18] WARNING[2221]: translate.c:256 ast_translator_b...

I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailb...

How does the g729 encoder/decoder count in regards to the total number of licenses and how does it count an encoder/decoder? I looked on the wiki and don't really see anything that explains it. In other words, how do the calls below count (assume no reinvite)? g729 phone calls into voicemail g729 phone ca...

I'm sitting here with my dunce cap on. My weak excuse is that I haven't ever played with g729 before. I have a Sipura 841. I have the phone config set to use g729. Its appropriate sip.conf entry, and the IAX stanza for my ITSP all set to disallow=all, allow=g729. But as soon as I dial, I get a complaint from the server: -- Call accepted by 66.225.202.72 (format g729) -- Fo...

I installed G729 from Diguim and I was expecting the sound quality on my i686 machine to be better than gsm. Compared to gsm, G729 sounds closer and a little robotic. Is this what is supposed to be or am I missing something? I am interested in G729 because the internet in my country is very expensive and I want t...