[Asterisk-Users] Sample extensions & SIP Conf files

[Stuart Buchanan]

I have got extreme problems with getting any incoming calls to ring
extensions, the sip debug info shows the CLI of the call coming in, but the
extensions do not ring, and the caller receives either a engaged tone or the
line is unobtainable depending on which provider is using.
 
I have used dozens of sample files to test, and I know I am doing it right
(aren't I?? :p) could one of you fine gentlemen/ladies be prepared to send
me a working sip.conf & extensions.conf file so I can compare the actions
with my conf files.
 
I know there are no firewall issues with this as its not behind a NAT or a
firewall for the purposes of the install. But its driving me mad.
 
If any of you are using Pipemedia's Pipecal service then I would be very
interested in the format of your files.
 
I know you are all thinking I am stupid, but I swear blind I am not! I have
looked at virtually every sample file on wiki and every tutorial on the web
and it all looks ok. Is anybosy else using the latest CVS release on red hat
9? Is it working ok, as I am convinced that all is ok with my config and
there is something else afoot. But until I totally rule out my stupidity I
can't rest.
 
If anyone has got some files I can look at please mail to "please at
broadbandtap dot co dot uk" sorry for putting mail address like that, it
just will help the spam when this message gets archived 
 
Regards
 
Stuart Buchanan
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