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

[Kristian Kielhofner]

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