Listening to agents on multiple server scenario

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Listening to agents on multiple server scenario

Postby hotdog0627 » Fri Jul 20, 2007 11:36 am

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."
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Postby hotdog0627 » Fri Jul 20, 2007 12:00 pm

also, why is it that after 20 seconds I get disconnected from the sessionID that I am listening to?
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Postby aster1 » Fri Jul 20, 2007 4:25 pm

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 .
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Postby gerski » Tue Jul 24, 2007 6:24 am

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.
http://www.perfectnetworkcorp.com
Perfect Network Corporation
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Postby hotdog0627 » Tue Jul 24, 2007 6:03 pm

im using only SIP trunks. aster's advise worked. thanks.
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Postby hotdog0627 » Tue Aug 07, 2007 8:56 pm

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
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