We are using G711 for phones to talk to Asterisk and G729 licenses at asterisk to talk to ITSP<br>
<br>
Could you please suggest transcoder to use from G711 and G729 and which
is comptible with Asterisk. We will like to avoid using TDM if possible<br>
<br>
Also i remember that initially we didn't have G729 and were using only
711 for with vicidial but then also we had same problems. at that time
it was only 2 or 3 agents. Again has anyone used it with SIP on Both
sides of asterisk ?<br>
<br>
Also what could be causing conference mixing, 1 agent can listen upto 3
customers, actually customers talk very clearly to each other.<br>
<br>
our pacing ratio is very less only 1:1.2 - 1:1.5<br>
<br><br><div><span class="gmail_quote">On 4/19/06, <b class="gmail_sendername">VICIDIAL</b> <<a href="mailto:vicidial@gmail.com">vicidial@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
What codec are you using for your SIP phones?<br><br>The next step I might suggest is having a separate transcoding<br>Asterisk server that would just act as a gateway between your provider<br>at G729 and ULAW so that you don't have to do your transcoding on the
<br>VICIDIAL server through meetme.<br><br>MATT---<br><br>On 4/19/06, Abhimanyu Rapria <<a href="mailto:Abhimanyu@synotek.com">Abhimanyu@synotek.com</a>> wrote:<br>> Hi Mat<br>><br>> linux kernel is SMP<br>
> Linux <a href="http://vicidial2.esselshyam.net">vicidial2.esselshyam.net</a> 2.6.11-1.1369_FC4smp #1 SMP Thu Jun 2<br>> 23:08:39 EDT 2005 i686 i686 i386 GNU/Linux<br>><br>> harddisk is WD sata 80 GB.<br>>
<br>> Has anyone use vicidial with SIP phones and SIP ITSP trunks before ?<br>><br>> We had full call recording enabled, Tomorrow we are going to call disabling<br>> it. but i am still doubting meetme how it mixes to sip calls.
<br>><br>> Regards<br>> Abhimanyu<br>><br>><br>> On 4/19/06, VICIDIAL <<a href="mailto:vicidial@gmail.com">vicidial@gmail.com</a>> wrote:<br>> > Are you using an SMP Linux kernel?<br>> >
<br>> > What kind of hard drives are you using?<br>> ><br>> > MATT---<br>> ><br>> > On 4/19/06, Abhimanyu Rapria <<a href="mailto:Abhimanyu@synotek.com">Abhimanyu@synotek.com</a>> wrote:
<br>> > ><br>> > ><br>> > ><br>> > > On 4/19/06, VICIDIAL <<a href="mailto:vicidial@gmail.com">vicidial@gmail.com</a>> wrote:<br>> > > > > [root@vicidial2 ~]# cat /proc/loadavg
<br>> > > > > 1.52 1.19 1.07 1/168 25019<br>> > > ><br>> > > > Is this on a single processor machine? If so you are running at over<br>> > > > 100% which is probably causing your problems with choppy audio.
<br>> > ><br>> > ><br>> > > Its a single CPU Pentium IV Hyper threading machine with one GB RAM<br>> with<br>> > > most of services stopped in init 3 mode on linux 1.2.xx kernel. totol
<br>> number<br>> > > of zap/pseudo channels + sip channels + sip itsp channels * 1.5 times<br>> pacing<br>> > > ratio will be not more than 50<br>> > ><br>> > > We need to run is only 12 agents using sip and 12 outgoing sip calls.
<br>> > > Database and Webserver are on seperate machine. If i place direct call<br>> from<br>> > > 12 agents, all calls go through fine. system also doesn't behave<br>> overload if<br>> > > i do top it shows only 0-15 % load.
<br>> > ><br>> > ><br>> > > > ><br>> > > > ><br>> > > > > There are 2 types of non hung ups<br>> > > > > 1) when there is a channel shown in output of "show channels" for
<br>> non<br>> > > > > hungup call in output of sip show channels (it has status ACK)<br>> > > > > 2) when there is no channel shown in output of "show channels" for<br>> non
<br>> > > > > hungup call in output of sip show channels (it has status d)<br>> > > > ><br>> > > > > usually you will find these calls between agent conferences and<br>> slowly
<br>> > > they<br>> > > > > keep on coming down and then stay at the end. Also if we login<br>> agents<br>> > > and<br>> > > > > conference is set between asterisk and agent but we dont resume
<br>> agent in<br>> > > > > autodial, then that conference remains set for full shift of 8 hrs<br>> for<br>> > > all<br>> > > > > 12 agents. but if we resume, then at start all is fine voice is
<br>> fine,<br>> > > etc<br>> > > > > but after 15 mins voice start detrioiting, chopping, low voice, etc<br>> and<br>> > > > > after some time calls get disconnected.<br>> > > > >
<br>> > > > > Also when we used the new vicidial.php file and everything is fine<br>> and<br>> > > > > agent is talking, suddenly the screen will come that the client hang<br>> up<br>> > > with
<br>> > > > > option go back or dispose call.<br>> > > > > agent keeps on talking for long time after that. This started after<br>> > > using<br>> > > > > new vicidial.php
which didn't have new_callback_call function<br>> > > > > implementation. Further many times the call is just hungup for now<br>> > > reason.<br>> > > > ><br>> > > > > Also we changed file recording name to include epoch and used new
<br>> > > > > vicidial.php but for the first time we lost a sale recording (we are<br>> > > using<br>> > > > > All call rec)<br>> > > ><br>> > > > Did you alter the code of
vicidial.php in any way from the release?<br>> > > > did you copy all files in the agc folder to your web server.<br>> > > ><br>> > > ><br>> > > > MATT---<br>> > > >
<br>> > > ><br>> > > ><br>> > > > ><br>> > > > > Example:<br>> > > > ><br>>
> > > > <a href="http://220.227.174.2">220.227.174.2</a>
agent15 018a383f451 00102/00000<br>> > > ulaw No Tx:<br>> > > > > ACK<br>>
> > >
> <a href="http://220.227.174.2">220.227.174.2</a> agent12
049848c527c 00102/00000<br>> > > ulaw No Tx:<br>> > > > > ACK<br>> > > > > <a href="http://220.227.174.2">220.227.174.2</a> agent1 3a18628547c<br>> 00102/00000<br>
> > > ulaw No Tx:<br>> > > > > ACK<br>> > > > > <a href="http://220.227.174.2">220.227.174.2</a> agent3 4ed0716a2a7<br>> 00102/00002<br>> > > ulaw No Rx:
<br>> > > > > ACK<br>> > > > > <a href="http://203.196.128.56">203.196.128.56</a> 6139467510 59a500d9076<br>> 00102/00101<br>> > > unkn No (d) Rx:<br>> > > > > BYE
<br>> > > > > <a href="http://203.196.128.56">203.196.128.56</a> 6139467507 07d286e5742<br>> 00102/00000<br>> > > g729 No Tx:<br>> > > > > ACK<br>> > > > >
<a href="http://220.227.174.2">220.227.174.2</a> agent2 5b98321f087<br>> 00102/00004<br>> > > ulaw No Rx:<br>> > > > > ACK<br>> > > > > <a href="http://220.227.174.2">
220.227.174.2</a> agent8 4df9d1be253<br>> 00102/00004<br>> > > ulaw No Rx:<br>> > > > > ACK<br>> > > > > <a href="http://220.227.174.2">220.227.174.2</a> agent5 36a5e01a0a0
<br>> 00102/00004<br>> > > ulaw No Rx:<br>> > > > > ACK<br>> > > > ><br>> > > > ><br>> > > > ><br>> > > > ><br>> > > > > > What version of Asterisk are you using?
<br>> > > > ><br>> > > > ><br>> > > > > Asterisk 1.2.5-netsec currently<br>> > > > ><br>> > > > ><br>> > > > > > have you applied the
<br>> cli_chan_concise_delimiter.patch<br>> > > to<br>> > > > > Asterisk if<br>> > > > > > you are using 1.2.X?<br>> > > > ><br>> > > > ><br>> > > > > I have to check this.
<br>> > > > ><br>> > > > > > MATT---<br>> > > > > ><br>> > > > > ><br>> > > > > ><br>> > > > > > ><br>> > > > > > >
<br>> > > > > ><br>> > > > > ><br>> > > > > > --<br>> > > > > > MATT---<br>> > > > > ><br>> > > > > > The astGUIclient/VICIDIAL project is sponsored by
<br>> > > > > > Binfone Telecom - <a href="http://www.binfone.com">http://www.binfone.com</a><br>> > > > > ><br>> > > > ><br>> > > > ><br>> > > >
<br>> > > ><br>> > > > --<br>> > > > MATT---<br>> > > ><br>> > > > The astGUIclient/VICIDIAL project is sponsored by<br>> > > > Binfone Telecom - <a href="http://www.binfone.com">
http://www.binfone.com</a><br>> > > ><br>> > ><br>> > ><br>> ><br>> ><br>> > --<br>> > MATT---<br>> ><br>> > The astGUIclient/VICIDIAL project is sponsored by
<br>> > Binfone Telecom - <a href="http://www.binfone.com">http://www.binfone.com</a><br>> ><br>><br>><br><br><br>--<br>MATT---<br><br>The astGUIclient/VICIDIAL project is sponsored by<br>Binfone Telecom -
<a href="http://www.binfone.com">http://www.binfone.com</a><br></blockquote></div><br>