I notice that priority of queue members is not being respected.
Using mysql realtime.
These are the queue members (in table queue_members) :
Local/queuemem0@ExternalCallFromQueue Local/queuemem1@ExternalCallFromQueue Local/queuemem2@ExternalCallFromQueue Local/queuemem3@ExternalCallFromQueue Local/queuemem4@ExternalCallFromQueue Local/queuemem5@ExternalCallFromQueue Local/queuemem6@ExternalCallFromQueue
Asterisk queue show :
sipserver*CLI> queue show myqueueq myqueueq has 0 calls (max unlimited) in 'ringall' strategy (10s holdtime, 99s talktime), W:0, C:13, A:0, SL:0.0% within 0s Members: queuemem0 (Local/queuemem0@ExternalCallFromQueue) with penalty 1 (realtime) (Not in use) has taken 5 calls (last was 551 secs ago) queuemem1 (Local/queuemem1@ExternalCallFromQueue) with penalty 1 (realtime) (Not in use) has taken no calls yet queuemem2 (Local/queuemem2@ExternalCallFromQueue) with penalty 2 (realtime) (Not in use) has taken 4…
I am running a pjsip test between two servers one running pjsip and one running chan_sip
The chan_sip side is sending requests based on qualify=yes.
The pjsip side is showing notices.. Exp
?[Oct 4 18:09:02] NOTICE: res_pjsip/pjsip_distributor.c:347 log_unidentified_request: Request from '"asterisk"
I have a pjsip install that is not pulling it's realtime registrations from an ODBC database. When I have them directly pull from a MySQL database everything seems to work. I am having trouble finding a requirements document for the pjsip realtime registrations. Is there some kind of special config that has to be used to trigger the connection for realtime registrations over ODBC? My realtime connections to aors, auths,contacts, and endpoints via ODBC are working as expected. Any ideas are appreciated. Asterisk v 13.5.0 Registrations line from sorcery.conf registrations=realtime,px1_ps_registrations Line for database from extconfig.conf px1_ps_registrations => odbc,pjsipRealtime pjsip show…
Hi guys i'm using asterisk 11.18.0. I need to send the pound # sign to my SIP provider, but each time it's reencoded in %23. I try to put pedantic=yes in the sip.conf as advised here:
but nothing's changed.
Have someone already met this issue please ?
thanks a lot,
Greetings everyone, I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream. In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information. I have noticed that when I do a MULTICAST page and send data from MP3Player, I get no sound on my speakers and get the following from 'core show channel PJSIP/xxx': NativeFormats: (slin)WriteFormat: slinReadFormat: slinWriteTranscode: No ReadTranscode: No I have noticed that when I do a…
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I have asterisk with sip registered accounts (realtime). Moreover I have SIP trunk defined as type=peer in sip.conf.
When call is incoming from SIP trunk with CLID of one of sip friend defined in MySQL sippeers table asterisk refuses INVITE as not authorized.
I tried to use insecure=port,invite options under SIP trunk definition in sip.conf but this not solves the problen.
Could you point me what could be the solution?
Thanks and regards Jarek
We are using 1.8 on CentOS 6
We use asterisk servers in pairs for machine level failover. On a recent pair we pointed the astdb location of both nodes of the pair to the same location on a shared storage device. Now it would appear that if the asterisk service is restarted, and queue members added via the AMI are forgotten.
Is there any issues in trying to share a single astdb over 2 machines that we are unaware of?
Thanks in Advance