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Listening to agents on multiple server scenario

Posted:
Fri Jul 20, 2007 11:36 am
by hotdog0627
Hi,
On a multi server scenario, how do I allow monitoring of agents from other asterisk servers?
From AST_timeonVDADall.php, when I click on "listen", if my X-Lite is on the same asterisk server as the agent, the call goes thru. But if we are on different asterisk servers, the call doesnot connect, saying "Call failed: Not acceptable here."

Posted:
Fri Jul 20, 2007 12:00 pm
by hotdog0627
also, why is it that after 20 seconds I get disconnected from the sessionID that I am listening to?

Posted:
Fri Jul 20, 2007 4:25 pm
by aster1
X-lite should work fine from that links . NOT ACCEPTABLE HERE means that asterisk is not accepting codec sent by phone . If your x-lite is registered then asterisk uses peer definition to select use codecs .. however when you click on monitor link x-lite makes a sip call to another asterisk server .. this is considered as simple incoming sip all by asterisk . Set
allow=ulaw
allow=alaw
allow=gsm
in general section of sip.conf in all asterisk servers then you should be able to barge any agent on any server by click on the link in page .

Posted:
Tue Jul 24, 2007 6:24 am
by gerski
make sure you have all iax trunk config in each server.. you can refer it to LOAD_BALANCE.txt, and create dialplan to dial the session ID to another server.

Posted:
Tue Jul 24, 2007 6:03 pm
by hotdog0627
im using only SIP trunks. aster's advise worked. thanks.

Posted:
Tue Aug 07, 2007 8:56 pm
by hotdog0627
weird, but ... the system works on "X-Lite" (Windows Version)
BUT... on a linux OS, with "xtensoftphone" it could not connect to conferences other than the conferences where the xtensoftphone is registered.
Why does it behave differently?
Does anybody know how to set up xtensoftphone properly?
Thanks in advance,
Noel