similar to: sip <> zap calls choppy, where to setup the jbuffer?

Hi all, I am playing with asterisk forwarding my calls to my cell phone. I have 2 X101P boards. When I call in, type the extension number that calls my cell #, all works as expected. Except, the audio is terrible. It is very choppy. I played with jitterbuffers, and that helped a little bit, but still the audio is very bad. I have echo problems on both lines, and haven't had

I believe the CorePlayer will play back FLAC. Atamido ----- Original Message ----- From: "Josh Coalson" <xflac@yahoo.com> To: "Harry Sack" <tranzedude@gmail.com>; <flac-dev@xiph.org> Sent: Thursday, March 22, 2007 6:26 PM Subject: Re: [Flac-dev] FLAC: players for Pocket PC > http://www.google.com/search?q=flac+pocket+pc > > e.g. > >

If you are having problems compiling NVFaxDetect (app_nv_faxdetect.c) or other Newman Telecom applications on Asterisk 1.4, please look at Steve's comments at: http://www.voip-info.org/wiki/view/NewmanTelOnAsterisk14 Several changes to Asterisk prevents NVFaxDetect and other apps from registering. Some changes needed. He also has copies of the code if you need it... Justin Newman

Hi List; Can someone advise me which IP Phone model that has buttons that can be assigned to do specific functionalities (call pickup, call formward, call appearance) and a transfer button and hold button? Which is the best of the following (that has buttons can be assigned to specific functions): Cisco 7970 or 7960 Polycom 501 Grandsream IP Phone Budge Tone 1001 or 1002 Linksys SPA 942 or 922

Hello, MOH always is choppy when someone calls from a cell phone to my pots or nufone 866. It sounds fine when it originates from a land line. I use zaptel hardware, and plenty of resources. I have tried to use different songs. None have the id3 tags, I tried the custom settings with -q -r 8000 -f 8192 -b 2048 --mono -s. Tried permanent resampling to 8khz, 16bit, filterd with lame -q1. I

Hello, I'm trying to set the 3rd span of a new digium quad card as a E&M T1 for Faxes to a Hylafax server. The 1st and 2nd spans are working as PRIs. When I start asterisk, the logs show a signaling error and chan_zap.c dies. I also get an error that it can't read the gains but they are the standard shown below. 2.6 kernel, Debian Stable, * 1.2 svn from feb 2007 my procedure: make

In the documentation, there is an example such as this: case $operatingsystem { sunos: { include solaris } # apply the solaris class redhat: { include redhat } # apply the redhat class default: { include generic } # apply the generic class } I''ve seen the $operatingsystem variable used elsewhere. What actually puts the value into the $operatingsystem variable? Does

Chris Mason (Lists) wrote: > Cory Andrews wrote: >> >> >> >> >> IP430, will sit between the IP301 and IP501, IP430 will have dual >> Ethernet, PoE, and full duplex speakerphone. List price (MSRP) $239 >> street price should fall likely between IP301 and IP501. >> > That looks great, the 301 is almost useless due to the lack of speaker

OK, background/config. running * (show version reports 0.9.0) on Mandrake 9.2 (kernel: 2.4.22-32mdk) with a dual 800mhz PIII with 256M Ram 4port FXO digium card, no IRQ sharing I can find (cat /proc/pci & cat /proc/interrupts), vmstat reports a minimum of 80+% CPU idle when problem occurs. connect to a Grandstream 101 (GS) via vpn (no nat). Link has 100ms - 150ms ROUND TRIP latency

On May 3, 2006, at 7:40 PM, Jean-Marc Valin wrote: > >> Yes. Jean-Marc has made the API more similar. >> >> Jean-Marc: Have you looked at the API we have for the >> asterisk/iaxclient jitterbuffer? > > Just did. > >> It's pretty close to what you have now -- the major difference is >> that >> your jb still assumes it can

Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that?

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