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

[Juan Rodríguez]

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081010/252dbe81/attachment.htm