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

[Ruben Rögels]

"Number of wished concurrent calls" times 4 = "Number of ports you'll
need to setup in rtp.conf" ;-)

regards,
Ruben

Am 02.11.2011 16:05, schrieb 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.
>>
>>
>> --
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> 
> 
> --
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> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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