1) If you are using phone alias why on server 1 these lines are commented out?
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; 100-350 phone extensions now auto-generated
; extensions for other SIP and IAX call center phones
; cc100-cc150 SIP Phones
;exten => _1[0-5]X,1,Dial(sip/cc${EXTEN},20,to)
; cc300-cc350 IAX Phones
;exten => _3[0-5]X,1,Dial(IAX2/cc${EXTEN},20,to)
If you created the phones via the admin interface- why on server 2 they are not?
Compare extensions-vicidial.conf, iax-vicidial.conf, sip-vicidial.conf in /etc/asterisk.
2) Again on server 1- these lines are indeed duplicated:
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exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
just a few lines apart from each other. We already been on this problem...
3) Again in server 1- why these lines are commented out?
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;exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)
;exten => _91NXXNXXXXXX,3,Hangup
Did you create a trunk via the admin interface? If so- why they are NOT commented out on server 2?
Is server 1 going to dial out only through one account, the one on server 2?
If using 2 providers- why the second trunk is not defined in the global section?
4) On both servers- you are missing quite a lot entries compared to extensions.conf.sample.
5) Is your asterisk version properly set up for each server in the admin page-->servers?
6) I'd start from a scratch with both extensions.conf, using the proper extensions.conf.sample from docs directory as a base and amend it as per your needs. They have to be the same on both servers, just amended as per load balance document. Do not introduce changes unless you know what they are about.