similar to: Choppy Sound when using linux router or asterisk

hello, I got those extensions: exten => 555,1,MeetMeCount(500|count) exten => 555,2,Gotoif,$[${count} = 1]?6 exten => 555,3,Meetme,500|pMs|1234 exten => 555,4,Playback,goodbye exten => 555,5,Hangup exten => 555,6,Goto(from-internal-custom,556,1) exten => 555,7,hangup exten => 556,1,System(/bin/cp /etc/asterisk/1-test /var/spool/asterisk/outgoing/) exten =>

Hi, Matt: What?s your server specifications that did you use? Best Regards, Cleviton. -----Mensagem original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]Em nome de Matt Florell Enviada em: sexta-feira, 23 de junho de 2006 11:38 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] Meetme max users

Hello, I have two broadvoice accounts. Lately, very often my broadvoice accounts are in unregistered state. When I log into asterisk I see: voip*CLI> sip show registry Host Username Refresh State sip.broadvoice.com:5060 yyy@s 120 Request Sent sip.broadvoice.com:5060 xxx@s 120 Request Sent xxx and yyy are my ohone numbers, when I do tcpdump on my router, I see only packets comming out my

Suppose I have one agent login into two different queue and there are calls waiting in both queues. If the calls in one queue has higher call prority (set QUEUE_PRO to higher value) than the calls in other queue, will the agent get the higher prority call first or the QUEUE_PRO has no effect? We have an Asterisk server( 1.2.17 with CentOS) running as ACD. We are having problem using weight option

Does anyone knows what is the max of users that meefme can handle. I am using Iax2 clients to connect to the conference. Thanks

Hi, I''m trying to figure out how to use Linux QoS. Default setting has three queues (bands) and should prioretise outgoing tarffic based on TOS field. I try to test that by flooding Ethernet interface by netperf or iptraf and running ping -f with -Q and without Q. -Q doesn''t affect ping results, it suffers anyway. It seems that I don''t understand something. I verified

Hey! I am trying to configure the voice onference with MeetMe application for my internal users. I have my server and 4 clients on same LAN and following is my extensions.conf file: [globals] Ahsen=SIP/222 Tahami=SIP/444 Uzair=SIP/333 Wasif=SIP/555 [internal] exten => 1234,1,Macro(voicemail,${Ahsen}) exten => 4321,1,Macro(voicemail,${Uzair}) exten => 5678,1,Macro(voicemail,${Tahami})

Hi, I have two accounts with broadvoice. Now, I want to be able to distinguish between them. I though that this would be simple by adding "/EXTEN" at the end of the register statement. For example: register => num1:pass@sip.broadvoice.com/1000 Unfortunately, this is not working. When I call into my box I hear busy tone. My config looks like this: [root@voip asterisk]# cat sip.conf

I have wondershaper to limit my upload at 400kilobits (my line is 600kbps). I do a lot of torrent seeding and I dont want my pings killed when I''m uploading so I set low prority source ports as follows (by the way, I have bittornet to only use ports 6881-6910): NOPRIOPORTSRC="6881 6882 6883 6884 6885 6886 6887 6888 6889 6890 6891 6892 6893 6894 6895 6896 6897 6898 6899 6900 6901

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-

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

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

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

Andrew, Please find links below to sound clips from different combination. You will see that the 2.6.9.34 kernel on trixbox 2beta still does not sound ok. What did you do on the trixbox 1.2.2 vmware load to make it work better? I say better because it is still not perfect as you will notice that it accelerates near the end. I really appreciate your help and the great work you are doing. Also,

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

1.) I'm getting messages in my log: Unable to open pseudo channel for timing... Sound may be choppy. Unable to open IAX timing interface: No such file or directory I'm using kernel 2.6, I don't think I timing, do I? 2.) I'm losing IAX registration with provider nor IAX protocol will go through. Though, I can ping both providers just fine. When I reboot the firewall, the