[asterisk-users] Unable to build sip pvt data - Switching equipment congestion

[Jonas Kellens]

Hello list,

can anyone tell me what the following means (found in messages log) :


/[Nov  2 11:16:21] ERROR[18407] rtp.c: No RTP ports remaining. Can't 
setup media stream for this call.
[Nov  2 11:16:21] WARNING[18407] chan_sip.c: Unable to create RTP audio 
session: Address already in use
[Nov  2 11:16:21] ERROR[18407] chan_sip.c: Unable to build sip pvt data 
for 'sipaccount7' (Out of memory or socket error)
[Nov  2 11:16:21] WARNING[18407] app_dial.c: Unable to create channel of 
type 'SIP' (cause 42 - Switching equipment congestion)/


Thank your for explaining the problems and a possible solution !


Greetingz,
Jonas.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111102/0b72292b/attachment.htm>