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

[Jonas Kellens]

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.
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                 http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111102/750f49bd/attachment-0001.htm>