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

[Jonas Kellens]

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> 
> [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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