Quoting mark navarro <[EMAIL PROTECTED]>:
regarding my simple problem of not being able access the asterisk
gui, the answer lies within the hosts file. since i editted the
network file (as described below) i should have reflected the
modification in the hosts file as well. that's it.
now my problem is, i have successfully created and tested extention
lines w/c can call each other but i cannot make outside voip calls
using sip. im using x-lite sip softphone. what other files should i
look into aside from sip.conf, extension.conf? can anybody post their
sip.conf and extension.conf here?
btw, i tested my voip account directly from x-lite and it works so i
know the problem is within asterisk itself.
help please?
Is that a new issue? or the original issue of being unable to access at
the asterisk gui?
How did you test your extensions? did you try calling from one
extension to other extension? and if the problem is calling outside
(via your voip provider termination) the problem probably lies on the
outbound routes configuration on sending traffic from your extension to
your voip provider or could be the dial rules?, also try to examine at
the asterisk cli if your extensions did register on your trixbox, and
if your ext have succesfully reached the voip provider and accepted
traffic
Registered SIP ext on trixbox looks like this on cli:
-- Registered SIP '101' at xxx.xxx.xxx.xxx port 6938 expires 3600
-- Saved useragent "eyeBeam release 3007n stamp 17816" for peer 101
Accepted traffic from the extension, sending traffic to your voip provider:
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/101-b730d128", "OUTNUM=13055085328") in new stack
-- Executing Set("SIP/101-b730d128", "custom=SIP/voip_provider") in
new stack
-- Executing GotoIf("SIP/101-b730d128", "0?customtrunk") in new stack
-- Executing Dial("SIP/101-b730d128",
"SIP/voip_provider/13055085328|120|r") in new stack
-- Called voip_provider/13055085328
-- SIP/voip_provider-b6a032f0 is making progress passing it to
SIP/101-b730d128
-- SIP/voip_provider-b6a032f0 answered SIP/101-b730d128
-- Attempting native bridge of SIP/101-b730d128 and
SIP/voip_provider-b6a032f0
Hope it helps and please post all necessary logs related to these
issues. Thank you.
Best Regards,
Jorge T. Monzor III
Telecom Analyst
Arriba Telecontact Inc.
Rosario Arcade, Limketkai Center
Cagayan de Oro City,Philippines 9000
1-305-508-5328 ext. 4
URL: http://www.arribatel.com
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