similar to: Asterisk and SER - choppy sound with G.729

Hello, When using Asterisk version 0.7.2, FreeBSD port with Mediatrix 1124 gateway, there is problem with DTMF "out-of-band". See debug below: Mediatrix forces (*) to use Payload Type as 96: [...] a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [...] Then we've got this nice debug from (*): May

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira

Hi, We are currently testing Asterisk with Wildcard 100XP and serveral Cisco ATA Box. Everything works great except that the card does not detect the hangup signal. We are using a standard Belgian PSTN line. I have not found anything about a be zone (only us, fr, de, nl, ...). Does someone experience the same problem? Do I need to create a new zone be (and how to do that)? Another small

Hi, I'm currently making some configuration with Asterisk and I wonder is someone has already a sample settings of indications.conf for Belgium. Thanks. Emmanuel Bergmans ---------------------------------------------------------- Perceval Technologies sa/nv Rue Tenbosch, 9 B-1000 Brussel Tel: +32-2-6409194 Fax: +32-2-6403154 General:info@perceval.net -

Dear All : I need to link my HQ to some Remote Sites - I need a Router which supports VOIP , and VPN Also the Router Should has 3 FXS ports and 1 FXO ... The call should be routed from the Remote Site to the HQ through a VPN tunnel ( 3DES ) ... Any Advise ? Mohamed Farid ,, Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are

Hey everyone, I saw over on the Vorbis-Dev list that they ported Ogg Vorbis over to Flash. Just wondering if this would be possible for Speex, and also if you have considered an iPhone implementation. I know that Apple is really hesitant to anger AT&T, but with Google's Android, VOIP on cell phones is just around the corner. Ironically voice bandwidth is set to go way down

Arek, This is awesome information. Why not chuck it on the web page at http://xiph.org/quicktime/development.html ? Cheers, Silvia. On Mon, Aug 23, 2010 at 7:07 PM, Arek Korbik <arkadini at gmail.com> wrote: > Hi, > > On Sun, Aug 22, 2010 at 10:28 PM, G S <stokestack at gmail.com> wrote: > > Hi all. > > > > Maybe there's a document about the build

Hi, I wanted to let you know that I have just made available the sources to the ogg + vorbis implementation in haXe, which I've been working on for last couple of weeks. The code compiles to an swf file playable in Flash Player 10. A demo of a simple player implementation (latest Flash 10 required): http://people.xiph.org/~arek/pg/hx/test.html and the sources, in a bzr branch, currently

Hi, I wanted to let you know that I have just made available the sources to the ogg + vorbis implementation in haXe, which I've been working on for last couple of weeks. The code compiles to an swf file playable in Flash Player 10. A demo of a simple player implementation (latest Flash 10 required): http://people.xiph.org/~arek/pg/hx/test.html and the sources, in a bzr branch, currently

I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec) when bridging IAX->IAX peers or SIP->SIP peers. My timing source is

Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw => ulaw is choppy, ulaw => alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit-

Hi there, We're having some complaints of choppy audio from our SIP customers. Asterisk is showing no errors, but I'm getting a lot of these in my syslog: Feb 17 13:34:31 ntop[2863]: **WARNING** packet truncated (14654->8232) The first number varies, but the last number is always 8232. I've read that this is a common MTU size, but none of our interfaces have an MTU of 8232.

Hello list, I've been experiencing choppy sound as well. The version on Asterisk I was using originally was dated 10/24/03 (I think), the problem appeared after I updated from that version. My setup is a little different though. I'm having choppy sound only on some incoming calls -- from PSTN->PBX (between spans on a TE410) and PSTN->SIP. We use Cisco 7940 handsets and we also

I have had no success in getting the voicemail working on Asterisk 1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried with or without ztdummy device, renice -20 on asterisk process and even real-time priority on the host Windows XP box for the vmware process. I am running on an AMD Athlon 64 X2 4600+. The behaviour is when the voicemail answer, the voice sound ok but when

Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)<==sip==>(asterisk 1)<====iax2 trunk====>(asterisk 2) <===alaw==>(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i

Hi! I have Asterisk 1.6.1 installed on OpenSuSE 11.0 running with choppy sound. One specialist on the forums asked me if I have DAHDI configured, he assumed that this could be cause of choppy sound problem. > dahdi_test Unable to open dahdi interface: No such file or directory Do I need to configure DAHDI even if I do not have any Zaptel devices? Is there any guide for configuring

Hey. I'm new to Linux based operating systems. A friend of mine introduced me a year ago and I finally decided to try it so try to bear with me here. The problem I've been having is that I've managed to get Amnesia to work under the latest Wine version (1.3.31) but the game is really choppy. I've tried messing with the Wine configuration and the in-game configuration but it

Hi All, Whenever a call comes in via the ISDN and somebody leaves a voicemail, the sound file recorded is very choppy. If I actually take the call, the sound is not choppy so it's obviously something to do with the Asterisk box itself having to do the recording. Perhaps the sound card drivers? I'm using the stock i810_audio (OSS) drivers on Fedora Core 1. If I call from a local VoIP

Hello All, We recently changed our asterisk system to begin using G.729a as the primary codec. We have a Cisco 1700-series router which connects to the PSTN via FXO ports, along with Cisco 7940 SIP phones. Everything is working great, except... When an inbound caller calls into our system, they hear an IVR. When the caller dials an ext (SIP phone), the ringing progress tone is

Hi After a day of running asterisk, I got choppy sound when fw ip->pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091009/be8bbf2f/attachment.htm