similar to: Sounds gets choppy after 30 seconds

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 all, I have my asterisk setup as following: IP 2 x E1 x-lite <-------> Asterisk -------> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data

Has anyone had any luck getting the addon's to compile under OS X? I have been able to get 1.06 to build and run great, but I really want to get the cdr_addon_mysql.so to enable mysql writing. When I try to compile I get a bunch of Undefined symbols and I have tried adding everything to my path. I've got mysql_devel, etc. I have posted this to the Mac Asterisk mailing list about a week

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

Hi, Does anyone have suggestions hooking Zyxel Prestige 2002 to Asterisk? I have tested Zyxel Prestige with both supported codecs. Call with G.711 sounds very choppy and cracking. Almost can't understand a word. Today I installed G.729 support into Asterisk but unbearable voice quality remains. It's a little bit better though. I have tested that Zyxel ATA with some commercial SIP

Hello, everyone I'm relatively new to Asterisk (and VOIP in general), but I have a project that it will really help with. So, I setup a test system on an ancient 400MHz P3 we had lying around. It worked great. I had a test dialplan working, and had no trouble connecting to it with SIP using 3CX SoftPhone over our LAN (and over the Net through our NAT). So, we went ahead and bought a

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 All, * is running a dream now, however we have an odd problem that I am sure some guru will be able to sort out for me in no time!! When receiving or making a call about 60 seconds or so into the call we develop choppy/stutter audio problems. It then seems to clear itself only to return again, and so the pattern carries on! This has got me stumped! Our equipment is SipTone II handsets, AVM

Hi folks, I'm experiencing some problems with sound through phpAGI ... What I'm trying to do is a menu, doing some database lookups and so ... But sometimes the sound become too choppy ... just sometimes .. like 1 of 5 calls ... but is a big percentage ... And I have my current menu on the dialplan that I have no problems with it ... I'm using .gsm for both but different

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 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

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! 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

Hi all. I have this configuration: Telco <-----(E1)----->TE410P//Dual Xeon Server 2.4Ghz<-----(Ethernet)----->Switch<----->GS//BT The Server is running RedHat Linux 8.0 with kernel 2.4.18-14-smp and we are having the following 2 issues: 1.- When making calls from the GrandStream to the PSTN the audio is choopy, plus theres is a pulsing sound, but when the GS

I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider -> IAX2 over the Internet -> 20Mb fiber connection -> router -> Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running

Ok I have a question. I have Xten-lite working with our Asterisk system and I am able to make and get calls. But the main problem is the sound is very choppy and sometimes it cuts off words. I have tested it with ulaw and alaw as well as GSM. They all do the same. ulaw seems to work better. I also have an ATA-186 which works great without this problem. Here is my Sip.conf settings.

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

Hello all! If I place a call to our number, the call is routed to our Asterisk box from teliax --> IAX2 --> firewall w/ port forwarding --> * If that caller dials an extension that rings an outside line, where our * box makes an outbound connection to teliax to terminate the call, we get choppy audio. Internal extensions have been dialing outbound calls no problem for over a week. What

Hi All, I've just setup a test Asterisk system that allows incoming/outgoing calls via an ISDN card (l4i) and incoming/outgoing calls via SIP (iconnecthere). I have two SIP Softphones (Xten X-Lite) for making and receiving calls. When receiving an incoming call via the ISDN interface the sound quality is fine for the Softphone user (i can hear the caller perfectly), but the person