[Asterisk-Users] Re: new_callback_call and conf disconnect
Abhimanyu Rapria
Abhimanyu at Synotek.com
Wed Apr 19 05:02:22 MST 2006
We are using G711 for phones to talk to Asterisk and G729 licenses at
asterisk to talk to ITSP
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 <vicidial at gmail.com> 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 <Abhimanyu at synotek.com> 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 <vicidial at gmail.com> wrote:
> > > Are you using an SMP Linux kernel?
> > >
> > > What kind of hard drives are you using?
> > >
> > > MATT---
> > >
> > > On 4/19/06, Abhimanyu Rapria <Abhimanyu at synotek.com> wrote:
> > > >
> > > >
> > > >
> > > > On 4/19/06, VICIDIAL <vicidial at gmail.com> wrote:
> > > > > > [root at vicidial2 ~]# 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 * 1.5times
> > 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060419/3c38084a/attachment.htm
More information about the asterisk-users
mailing list