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

[Danny Nicholas]

You have set an insufficient range in rtp.conf.  Asterisk uses 2 ports per
call, but allocates 4 for transferring, etc, so when you set up a range of
10001-10040 (for example) you are basically setting a range of 10 concurrent
calls.  Check rtp.conf and make the end range larger by 8 or 12 or whatever
number of extra calls you'd like to see before you get this message again.

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, November 02, 2011 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Unable to build sip pvt data - Switching equipment
congestion

 

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.

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