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</o:shapelayout></xml><![endif]--></head><body bgcolor=white lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>As I understand it, the scenario you describe would only use 2 channels (I don’t think the RTP channel gets established until connection; I could be wrong about this as Asterisk might pre-reserve the channels for early media, etc.) – do keep in mind however that although you mention 2 and 2 (1 each for incoming channel and ringing/answered), you have actually “blocked out” 8 channels as the range of 4 per call is used. <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>For example<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>SIP/100 calls SIP/101<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Call from SIP/100 uses 11501 and 11502 and 11503 and 11504 are reserved for transfer, etc.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>SIP/101 answer uses 11505 and 11506 and reserves 11508/11509 for its next steps.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I would suggest doing a netstat –anp while you are ringing your 10 peers to see which RTP ports are in use.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Jonas Kellens<br><b>Sent:</b> Wednesday, November 02, 2011 10:20 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Unable to build sip pvt data - Switching equipment congestion<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>On 11/02/2011 04:13 PM, Danny Nicholas wrote: <o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>150/4 = 37.5. maybe your sip peer has a conflicting range?</span><o:p></o:p></p><p class=MsoNormal><br>Where do I set this range in my peer definition ? I don't think there is such a parameter in sip.conf<br><br><br>To be perfectly clear, how many RTP-ports are needed in the below situation :<br><br>- an incoming call to a group of SIP-peers (10 in total)<br>- 1 peer answers this incoming call<br><br>My thought : 2 RTP for incoming channel, 2 RTP for channel to SIP peer<br>(and the other peers don't matter)<br><br>Am I correct ?<br><br>Or is there a need for a channel to every peer that is "ringing" ?<br><br><br><br><br><br><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> </span><o:p></o:p></p><div><div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0in 0in 0in;border-color:-moz-use-text-color -moz-use-text-color'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [<a href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Jonas Kellens<br><b>Sent:</b> Wednesday, November 02, 2011 10:06 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Unable to build sip pvt data - Switching equipment congestion</span><o:p></o:p></p></div></div><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><span style='font-family:"Helvetica","sans-serif"'>Hello,<br><br>thank you for your answer...<br><br>Current range (rtp.conf) : 11500 - 11650<br><br>Current calls : 20 à 25<br><br>Is this not sufficient ??<br><br><br><br><br>Jonas.<br><br><br></span><br>On 11/02/2011 04:00 PM, Danny Nicholas wrote: <o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>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.</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> </span><o:p></o:p></p><div><div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0in 0in 0in;border-color:-moz-use-text-color'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [<a href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Jonas Kellens<br><b>Sent:</b> Wednesday, November 02, 2011 9:57 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> [asterisk-users] Unable to build sip pvt data - Switching equipment congestion</span><o:p></o:p></p></div></div><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><span style='font-family:"Helvetica","sans-serif"'>Hello list,<br><br>can anyone tell me what the following means (found in messages log) :<br><br><br><i>[Nov 2 11:16:21] ERROR[18407] rtp.c: No RTP ports remaining. Can't setup media stream for this call.<br>[Nov 2 11:16:21] WARNING[18407] chan_sip.c: Unable to create RTP audio session: Address already in use<br>[Nov 2 11:16:21] ERROR[18407] chan_sip.c: Unable to build sip pvt data for 'sipaccount7' (Out of memory or socket error)<br>[Nov 2 11:16:21] WARNING[18407] app_dial.c: Unable to create channel of type 'SIP' (cause 42 - Switching equipment congestion)</i><br><br><br>Thank your for explaining the problems and a possible solution !<br><br><br>Greetingz,<br>Jonas.</span><o:p></o:p></p><pre> <o:p></o:p></pre><pre> <o:p></o:p></pre><pre>--<o:p></o:p></pre><pre>_____________________________________________________________________<o:p></o:p></pre><pre>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<o:p></o:p></pre><pre>New to Asterisk? 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