"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:*[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 -- _____________________________________________________________________ -- 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
