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