asterisk users - Apr 2006 - Re: new_callback_call and conf disconnect

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@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@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@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@synotek.com> wrote:
> > > >
> > > >
> > > >
> > > > On 4/19/06, VICIDIAL <vicidial@gmail.com> wrote:
> > > > > >  [root@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
>
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