150/4 = 37.5. maybe your sip peer has a conflicting range?
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 youd like to see before you get this message again. From: [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. -- _____________________________________________________________________ -- 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
-- _____________________________________________________________________ -- 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
