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:*[email protected]
[mailto:[email protected]] *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:*[email protected]
<mailto:[email protected]>
[mailto:[email protected]] *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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