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
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
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
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
--
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?
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