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 < [EMAIL PROTECTED]> wrote: > On 10/10/08, Juan Rodríguez <[EMAIL PROTECTED]> 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 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227
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