I followed the instruction http://wiki.linuxmce.org/index.php/How_to_properly_add_support_for_SIP_providers
to add support for my Swedish SIP provider (www.affinity.se
). The tech support claims they have customers running their service with asterisk. When opening up the phone line in the web admin it says "Registered <date> <hour>" under status. My guess is that this means it successfully connected to the host, and the host accepted the credentials.
Now the problem is that neither outgoing nor incoming calls work. I don't know anything about asterisk so I'm quite lost what I have messed up. The instruction was very simple and straightforward though... The only magic number I found was $DECLARED_PREFIX = "9". It seems to be set to 9 for most, but not all, available providers. I don't know what it is, or if it is important. I tried "9" and "", same result.
When I placed a test call from my cell phone I noted the following entry in /var/log/asterisk/cdr-csv/Master.csv
"","NNNNNNNNNN","s","from-pstn","""NNNNNNNNNN"" <NNNNNNNNNN>","SIP/XXXXXXXXX-b54972b0","","SayAlpha","","2010-12-07 21:46:34","2010-12-07 21:46:34","2010-12-07 21:4
Where NNN is my cell phone number and XXX is my phone number assigned by SIP provider.
Does this say anything meaningful? When I place the call a voice says "The number you have dialed is not in service. Please check the number and try again". Are there any other relevant logs that could give me a hint about whats going on?
When I try to place an outgoing call, I get the message "Call dropped. Reason: Normal clearing".
Any help is appreciated!