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

[Danny Nicholas]

As I understand it,  the scenario you describe would only use 2 channels (I
don’t think the RTP channel gets established until connection;  I could be
wrong about this as Asterisk might pre-reserve the channels for early media,
etc.) – do keep in mind however that although you mention 2 and 2 (1 each
for incoming channel and ringing/answered), you have actually “blocked out”
8 channels as the range of 4 per call is used.  

For example

SIP/100 calls SIP/101

Call from SIP/100 uses 11501 and 11502 and 11503 and 11504 are reserved for
transfer, etc.

SIP/101 answer uses 11505 and 11506 and reserves 11508/11509 for its next
steps.

 

I would suggest doing a netstat –anp while you are ringing your 10 peers to
see which RTP ports are in use.

 

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 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unable to build sip pvt data - Switching
equipment congestion

 

On 11/02/2011 04:13 PM, Danny Nicholas wrote: 

150/4 = 37.5.  maybe your sip peer has a conflicting range?


Where do I set this range in my peer definition ? I don't think there is
such a parameter in sip.conf


To be perfectly clear, how many RTP-ports are needed in the below situation
:

- an incoming call to a group of SIP-peers (10 in total)
- 1 peer answers this incoming call

My thought : 2 RTP for incoming channel, 2 RTP for channel to SIP peer
(and the other peers don't matter)

Am I correct ?

Or is there a need for a channel to every peer that is "ringing" ?







 

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 10:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unable to build sip pvt data - Switching
equipment congestion

 

Hello,

thank you for your answer...

Current range (rtp.conf) : 11500 - 11650

Current calls : 20 à 25

Is this not sufficient ??




Jonas.



On 11/02/2011 04:00 PM, Danny Nicholas wrote: 

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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