[asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

[Juan Rodríguez]

Kristian:
Thanks for your reply. I am running asterisk as root, but still getting this
error.

I did a test while running asterisk 1.4.21 version setting "ulimit -n
32768", but after restaring asterisk it stop working with less than 150
calls (less than 300 channels).

Any suggestion??


On Fri, Oct 10, 2008 at 11:37 AM, Kristian Kielhofner <
kkielhofner at star2star.com> wrote:

> On 10/10/08, Juan Rodríguez <jerdguez at gmail.com> wrote:
> > After getting some ERRORS like this:
> >
> > [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup
> > media stream for this call.
> >  [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup
> > media stream for this call.
> > [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup
> > media stream for this call.
> > [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup
> > media stream for this call.
> >
> > I start getting:
> >
> > ERROR[14844] chan_sip.c: Unable to build sip pvt data for
> > 'TRUNK/DESTINATION' (Out of memory or socket error)
> > [Oct 9 22:26:45] ERROR[14832] chan_sip.c: Unable to build sip pvt data
> for
> > 'TRUNK/DESTINATION' (Out of memory or socket error).
> >
> > I had installed Asterisk-1.4.21, but this version stop from receiving
> calls
> > after these errors occured.
> >
> > Then I downgrade to version 1.4.19 (because I had have tested that
> version),
> > but after getting these error it stop from creating the outbound call.
> >
> > The configuration of the * is an incomming call from the my SIP Provider
> and
> > after internal manage it makes a second call to other destination--DID--.
> >
> > For AGI compatibility issues I could not use Version 1.4.22 (issues whith
> > DeadAGI for billing purpuses).
> >
> >
> >
> > This is my rtp.conf
> >
> >
> >  [general]
> > ;
> > ; RTP start and RTP end configure start and end addresses
> > ;
> > ; Defaults are rtpstart=5000 and rtpend=31000
> > ;
> > rtpstart=10000
> > rtpend=20000
> >
> >
> > This is my sip.conf for the TRUNK
> >
> >
> >  [TRUNK]
> > type=peer
> > nat=never
> > host=destination.public.ip.address
> > fromdomain=my.public.ip.address
> > dtmfmode=rfc2833
> > canreinvite=no
> > disallow=all
> > allow=g729
> >
> >
> > Please help.
> > --
> > Juan E. Rodríguez
> >
>
> Juan,
>
>  You might need to increase the number of file descriptors available
> in Linux.  What distro are you on?  Are you using the Asterisk startup
> scripts?  In later versions this is done for you automatically if you
> are running Asterisk as root.  Have a look at this:
>
> http://www.voip-info.org/wiki/view/file+descriptors
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
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-- 
Juan E. Rodríguez
Cel. 829-886-5565
Work: 809-724-9227
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