Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
On Fri, Oct 17, 2008 at 01:24:35AM -0400, Juan Rodríguez wrote: Tzafrir: Following the comments on your post, I started checking (after breaking my head 'googling') the UDP ports in use, and found out that the script that my Asterisk is running was using UDP connection too. This caused that ports from 10,000 to 20,000 could not be used by Asterisk. I change the port range from 10,000 to 40,, and now everything looks OK. Why not change it to 9000- ? Do you actually need more than 1000 sockets at a time? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
I do, I am planning to have little more than 1000. Right now I had managed little more than 700 SIP channels + 100 IAX channels. Do you think this can cause any problem?? --I mean, having this RTP ports range-- Tzafrir Cohen wrote: On Fri, Oct 17, 2008 at 01:24:35AM -0400, Juan Rodríguez wrote: Tzafrir: Following the comments on your post, I started checking (after breaking my head 'googling') the UDP ports in use, and found out that the script that my Asterisk is running was using UDP connection too. This caused that ports from 10,000 to 20,000 could not be used by Asterisk. I change the port range from 10,000 to 40,, and now everything looks OK. Why not change it to 9000- ? Do you actually need more than 1000 sockets at a time? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
On Fri, Oct 17, 2008 at 03:11:17PM -0400, Juan E. Rodríguez wrote: I do, I am planning to have little more than 1000. Right now I had managed little more than 700 SIP channels + 100 IAX channels. Do you think this can cause any problem?? --I mean, having this RTP ports range-- If you never had anything close to the order of magnitude of 1 SIP channels, the range of 1 RTP ports should have been well over enough. Unless your scripts have done very funny things (using over 5 sockets per Asterisk channel. Which is funny indeed, becuase even chan_h323 isn't that bad). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
Tzafrir: Following the comments on your post, I started checking (after breaking my head 'googling') the UDP ports in use, and found out that the script that my Asterisk is running was using UDP connection too. This caused that ports from 10,000 to 20,000 could not be used by Asterisk. I change the port range from 10,000 to 40,, and now everything looks OK. Thanks for replying, Juan On Fri, Oct 10, 2008 at 3:09 PM, Juan Rodríguez [EMAIL PROTECTED] wrote: Having 600 channels it would be like 1200 RTP ports. And on the rtp.conf I have fonfigured from 1 to 2. I do not think this is the problem. Thanks, Juan On Fri, Oct 10, 2008 at 1:38 PM, Tzafrir Cohen [EMAIL PROTECTED]wrote: On Fri, Oct 10, 2008 at 11:56:34AM -0400, Juan Rodríguez wrote: Kristian: Thanks for your reply. I am running asterisk as root, but still getting this error. I did a test while running asterisk 1.4.21 version setting ulimit -n 32768, but after restaring asterisk it stop working with less than 150 calls (less than 300 channels). Are file descriptors the problem? ls /proc/PID_OF_ASTERISK/fd | wc Maybe there are really not enough open ports? Start with: netstat -anu Or: netstat -anup -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan E. Rodríguez -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup media stream for this call. I start getting: ERROR[14844] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error) [Oct 9 22:26:45] ERROR[14832] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error). I had installed Asterisk-1.4.21, but this version stop from receiving calls after these errors occured. Then I downgrade to version 1.4.19 (because I had have tested that version), but after getting these error it stop from creating the outbound call. The configuration of the * is an incomming call from the my SIP Provider and after internal manage it makes a second call to other destination--DID--. For AGI compatibility issues I could not use Version 1.4.22 (issues whith DeadAGI for billing purpuses). This is my rtp.conf [general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 ; rtpstart=1 rtpend=2 This is my sip.conf for the TRUNK [TRUNK] type=peer nat=never host=destination.public.ip.address fromdomain=my.public.ip.address dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 Please help. -- Juan E. Rodríguez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
Kristian: Thanks for your reply. I am running asterisk as root, but still getting this error. I did a test while running asterisk 1.4.21 version setting ulimit -n 32768, but after restaring asterisk it stop working with less than 150 calls (less than 300 channels). Any suggestion?? On Fri, Oct 10, 2008 at 11:37 AM, Kristian Kielhofner [EMAIL PROTECTED] wrote: On 10/10/08, Juan Rodríguez [EMAIL PROTECTED] wrote: After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup media stream for this call. I start getting: ERROR[14844] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error) [Oct 9 22:26:45] ERROR[14832] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error). I had installed Asterisk-1.4.21, but this version stop from receiving calls after these errors occured. Then I downgrade to version 1.4.19 (because I had have tested that version), but after getting these error it stop from creating the outbound call. The configuration of the * is an incomming call from the my SIP Provider and after internal manage it makes a second call to other destination--DID--. For AGI compatibility issues I could not use Version 1.4.22 (issues whith DeadAGI for billing purpuses). This is my rtp.conf [general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 ; rtpstart=1 rtpend=2 This is my sip.conf for the TRUNK [TRUNK] type=peer nat=never host=destination.public.ip.address fromdomain=my.public.ip.address dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 Please help. -- Juan E. Rodríguez Juan, You might need to increase the number of file descriptors available in Linux. What distro are you on? Are you using the Asterisk startup scripts? In later versions this is done for you automatically if you are running Asterisk as root. Have a look at this: http://www.voip-info.org/wiki/view/file+descriptors -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
On 10/10/08, Juan Rodríguez [EMAIL PROTECTED] wrote: After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup media stream for this call. I start getting: ERROR[14844] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error) [Oct 9 22:26:45] ERROR[14832] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error). I had installed Asterisk-1.4.21, but this version stop from receiving calls after these errors occured. Then I downgrade to version 1.4.19 (because I had have tested that version), but after getting these error it stop from creating the outbound call. The configuration of the * is an incomming call from the my SIP Provider and after internal manage it makes a second call to other destination--DID--. For AGI compatibility issues I could not use Version 1.4.22 (issues whith DeadAGI for billing purpuses). This is my rtp.conf [general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 ; rtpstart=1 rtpend=2 This is my sip.conf for the TRUNK [TRUNK] type=peer nat=never host=destination.public.ip.address fromdomain=my.public.ip.address dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 Please help. -- Juan E. Rodríguez Juan, You might need to increase the number of file descriptors available in Linux. What distro are you on? Are you using the Asterisk startup scripts? In later versions this is done for you automatically if you are running Asterisk as root. Have a look at this: http://www.voip-info.org/wiki/view/file+descriptors -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
On 10/10/08, Juan Rodríguez [EMAIL PROTECTED] wrote: Kristian: Thanks for your reply. I am running asterisk as root, but still getting this error. I did a test while running asterisk 1.4.21 version setting ulimit -n 32768, but after restaring asterisk it stop working with less than 150 calls (less than 300 channels). Any suggestion?? Here's another (fuller) list, shamelessly lifted from another mailing list: ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 99 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 244 ulimit -l unlimited Make sure these are in your Asterisk startup scripts before Asterisk starts. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
On Fri, Oct 10, 2008 at 11:56:34AM -0400, Juan Rodríguez wrote: Kristian: Thanks for your reply. I am running asterisk as root, but still getting this error. I did a test while running asterisk 1.4.21 version setting ulimit -n 32768, but after restaring asterisk it stop working with less than 150 calls (less than 300 channels). Are file descriptors the problem? ls /proc/PID_OF_ASTERISK/fd | wc Maybe there are really not enough open ports? Start with: netstat -anu Or: netstat -anup -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
Having 600 channels it would be like 1200 RTP ports. And on the rtp.conf I have fonfigured from 1 to 2. I do not think this is the problem. Thanks, Juan On Fri, Oct 10, 2008 at 1:38 PM, Tzafrir Cohen [EMAIL PROTECTED]wrote: On Fri, Oct 10, 2008 at 11:56:34AM -0400, Juan Rodríguez wrote: Kristian: Thanks for your reply. I am running asterisk as root, but still getting this error. I did a test while running asterisk 1.4.21 version setting ulimit -n 32768, but after restaring asterisk it stop working with less than 150 calls (less than 300 channels). Are file descriptors the problem? ls /proc/PID_OF_ASTERISK/fd | wc Maybe there are really not enough open ports? Start with: netstat -anu Or: netstat -anup -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan E. Rodríguez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
