Re: [asterisk-users] RTP ports used by Asterisk in dialplan
You can use tcpdump portrange 1-2 udp 2011/10/20 Andrew Higgs andrew.m.hi...@gmail.com Hi Isabel, Could you not just filter out after the fact using something like Wireshark? Regards On Thu, Oct 20, 2011 at 3:28 PM, ISABEL ORDAS ARNAL i...@tid.es wrote: Dear all, ** ** Do you know if there is a way to know the 2 RTP ports that Asterisk is using for audio flow in a call in the dialplan? I would like to launch a Linux shell command “tcpdump” to capture audio flow in those 2 RTP ports before call starts and stop capturing at the end of the call. ** ** Regards, Isabel -- Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- 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
[asterisk-users] RTP ports used by Asterisk in dialplan
Dear all, Do you know if there is a way to know the 2 RTP ports that Asterisk is using for audio flow in a call in the dialplan? I would like to launch a Linux shell command tcpdump to capture audio flow in those 2 RTP ports before call starts and stop capturing at the end of the call. Regards, Isabel Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- 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
Re: [asterisk-users] RTP ports used by Asterisk in dialplan
You could temporarily change rtp.conf to use just 4 ports (say 10001-10004) and monitor 10001 and 10002. On a production system you would have to do something with a tool like netstat to try and predict which ports in the range would be used. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ISABEL ORDAS ARNAL Sent: Thursday, October 20, 2011 8:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] RTP ports used by Asterisk in dialplan Dear all, Do you know if there is a way to know the 2 RTP ports that Asterisk is using for audio flow in a call in the dialplan? I would like to launch a Linux shell command tcpdump to capture audio flow in those 2 RTP ports before call starts and stop capturing at the end of the call. Regards, Isabel _ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- 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
Re: [asterisk-users] RTP ports used by Asterisk in dialplan
Hi Isabel, Could you not just filter out after the fact using something like Wireshark? Regards On Thu, Oct 20, 2011 at 3:28 PM, ISABEL ORDAS ARNAL i...@tid.es wrote: Dear all, ** ** Do you know if there is a way to know the 2 RTP ports that Asterisk is using for audio flow in a call in the dialplan? I would like to launch a Linux shell command “tcpdump” to capture audio flow in those 2 RTP ports before call starts and stop capturing at the end of the call. ** ** Regards, Isabel -- Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- 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
[asterisk-users] RTP ports
Hello, I need to limit the RTP ports used by an asterisk in a client, Actualy the range defined is from 1 to 2 udp ports. If I only have 10 local sip extension ¿how many ports/range should I set up in /etc/asterisk/rtp.conf? Which is the way to calculate the rtp ports needed on an instalation? Thanks in advance, Voipcrazy. -- _ -- 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
Re: [asterisk-users] RTP ports
In my installation, netstat usually indicates 4 ports per extension, so my assumption is that you would need 40 ports or a range of 1-10039. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of voip crazy Sent: Monday, May 03, 2010 6:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTP ports Hello, I need to limit the RTP ports used by an asterisk in a client, Actualy the range defined is from 1 to 2 udp ports. If I only have 10 local sip extension ¿how many ports/range should I set up in /etc/asterisk/rtp.conf? Which is the way to calculate the rtp ports needed on an instalation? Thanks in advance, Voipcrazy. -- _ -- 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
Re: [asterisk-users] RTP ports
On Mon, May 3, 2010 at 3:04 PM, Danny Nicholas da...@debsinc.com wrote: In my installation, netstat usually indicates 4 ports per extension, so my assumption is that you would need 40 ports or a range of 1-10039. Sounds reasonable, I was going to suggest 100 would easily do, but an actual measured value is even better :) r -- _ -- 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
Re: [asterisk-users] RTP ports
Hi! In my installation, netstat usually indicates 4 ports per extension, so my assumption is that you would need 40 ports or a range of 1-10039. Sounds reasonable, I was going to suggest 100 would easily do, but an actual measured value is even better :) Be a bit careful: At least in the past Asterisk did not clear unused RTP/RTCP ports as expected. Also take transfers into account, i.e. phones that run more than 1 call in parallel. Philipp -- _ -- 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
Re: [asterisk-users] RTP ports
On Mon, May 3, 2010 at 4:25 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: so my assumption is that you would need 40 ports or a range of 1-10039. Sounds reasonable, I was going to suggest 100 would easily do, but an actual measured value is even better :) In my installation, netstat usually indicates 4 ports per extension, Be a bit careful: At least in the past Asterisk did not clear unused RTP/RTCP ports as expected. Also take transfers into account, i.e. phones that run more than 1 call in parallel. Incidentally, my router was limited to 20 ports to forward and I used 1-10020. For a three person company we never had problems. (Anecdotal, I know) r -- _ -- 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
[Asterisk-Users] RTP ports in use grows and grows...
We've been seeing a pattern over the last couple of weeks with our Asterisk servers (1.0.9). The number of ports in use (RTP) seems to grow by two every minute or so. Eventually the server will run out of allowable files open and crash. We are resetting the server once per day to prevent this from occurring. Running lsof shows the end of the list like this: asterisk 26733 astx 1794u IPv4 183341654 UDP 192.168.1.123:22790 asterisk 26733 astx 1795u IPv4 183341655 UDP *:22791 asterisk 26733 astx 1796u IPv4 183312190 UDP 192.168.1.123:24170 asterisk 26733 astx 1797u IPv4 183312191 UDP *:24171 asterisk 26733 astx 1798u IPv4 183335059 UDP 192.168.1.123:21132 asterisk 26733 astx 1799u IPv4 183335060 UDP *:21133 asterisk 26733 astx 1802u IPv4 183342252 UDP 192.168.1.123:20358 asterisk 26733 astx 1803u IPv4 183342253 UDP *:20359 and the list grows and grows (slowly but steadily). We have only around 600 SIP clients on this particular server so something is clearly amiss. My questions are: 1. Are there any known file handle leaks in rtp.c or sip.c? 2. If the number of ports open eventually reaches the port boundaries I have configured in rtp.c (e.g. start at 2, end at 25000), will Asterisk crash, or it will it start to recycle those ports? 3. I have set ulimit -n to a higher number but this doesn't seem like a solution, only a temporary band-aid. Thank you, Bryan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP Ports mismatch Astwind modem support
Hello, I have been rattling my brain and searching the mailing list for 1 week before posting. Iam faced with the followng challenge. When using Xlite, all works well. When using * there is no voice either way upon off-hookor even when redirected to voice mail(problem occurs with any VOIP provider). Ethereal trace indicates, uponincoming call, arequest from the VOIP provider with all codecs imaginable then a request from * to the registered x-ten lite (configured as the receiving extension with silence set to transmit) with only the G -723 codec. Then a request from * to Xlite for audio over RTP 8552 and then an ok from X-lite to * for audio over RTP 8000. Configuration follows: ADSL direct connection to Win XP-SP2 box running Astwind, all correct ports are opened in the ICS/ICF firewalll and forwarded to the * IP. X lite registers correctly and internal calls to Voice mail are fine. sip.conf is very simple and follows X-lite sample config file. Nat is set to yes, canreinvite to no. rtp.confstates that rtp traffic starts at 8000 ends at 65000. Any clue why I cannot receive/transmit voice at all via * while the Xlite works fine directly to the VOIP provider? Regarding Astwind, does anybody know if it supports a chipset modem 537 from intel as a card? Thanks. An * evengelist ! Find the music you love on MSN Music. Start downloading now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
