[asterisk-users] One thread per peer
Is this normal to create one thread per peer in Asterisk 12, chan_sip regular, not pjsip? What happens is I have 659 peers, and I get 682 tasks on ls /proc/15373/task | wc -l If this is normal then of course I can only get a few instances before my box collapses. Is it any different in pjsip? -- _ -- 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] Asterisk 12 is broken
The amount of threads went through the roof ls /proc/15373/task | wc -l 682 in version SVN-branch-12-r427618M it used to be 18 in Asterisk SVN-branch-11-r412226M How can I trace this? There are no calls open, on a disconnected system -- _ -- 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] taskprocessor fails to allocate memory
I keep getting this error [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:614 __allocate_taskprocessor: Unable to start taskprocessor listener for taskprocessor bbe08c34-9d1c-4e5f-8ae0-0cc75289caca [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:245 default_listener_shutdown: pthread_join(): Cannot allocate memory [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:614 __allocate_taskprocessor: Unable to start taskprocessor listener for taskprocessor f30fcb95-d290-4bb1-8008-290b79342c01 [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:245 default_listener_shutdown: pthread_join(): Cannot allocate memory [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:614 __allocate_taskprocessor: Unable to start taskprocessor listener for taskprocessor 38660bf7-eec2-4ce6-a9d7-63c8178a0556 [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:245 default_listener_shutdown: pthread_join(): Cannot allocate memory After 18 instances of Asterisk using parameter -C /etc/asterisk1/asterisk.conf -C /etc/asterisk2/asterisk.conf -C /etc/asterisk3/asterisk.conf etc. The machine has 180 GB of RAM and 16 cores. ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 1048576 max locked memory (kbytes, -l) unlimited max memory size (kbytes, -m) unlimited open files (-n) 1048576 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 8192 cpu time (seconds, -t) unlimited max user processes (-u) unlimited virtual memory (kbytes, -v) unlimited resources is plenty free -h totalusedfree shared buff/cache available Mem: 177G 60G111G508K5.4G 116G Swap: 269G 0B269G and nothing else runs in the box I am using regular chan_sip Where do I go from here? Your help is appreciated. -- _ -- 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] How to find RTP address of ongoing call?
Not sure if this helps but I've used the following in my dialplan in the
past:
;Get MTA IP from SIP header
;same => n,Verbose(2,rtpdest = ${CHANNEL(rtpdest)})
you'll see something like the following in the logs:
[Nov 8 13:29:05] == rtpdest = 192.168.1.75:7078
not sure how to do it via CLI though.
Bryan Burroughs
On 11/08/2014 07:57 AM, Markus wrote:
Hi list,
probably this is a FAQ but I can't seem to find it. How to find the
RTP IP address of an ongoing SIP call?
"sip show channels" seems to list the RTP address under the very left
column called "Peer". And it also lists the associated "Call ID" which
I could associate with a call by executing sip show channel
and before figuring out the Channel by running core show channels
concise, but the issue is that the Call ID output from sip show
channels is cut off and limited to 16 characters.
Thanks!
Markus
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[asterisk-users] How to find RTP address of ongoing call?
Hi list, probably this is a FAQ but I can't seem to find it. How to find the RTP IP address of an ongoing SIP call? "sip show channels" seems to list the RTP address under the very left column called "Peer". And it also lists the associated "Call ID" which I could associate with a call by executing sip show channel and before figuring out the Channel by running core show channels concise, but the issue is that the Call ID output from sip show channels is cut off and limited to 16 characters. Thanks! Markus -- _ -- 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
