A Bit OT – Configure GoIP For Asterisk
I recently received a GoIP-32 for a client project — primarily outbound calling.
How should a GoIP be configured for Asterisk? No fancy shmancy Elastix or FPBX GUI — just using the configuration files.
Single Server Mode, Config By Line, and Trunk Gateway Mode all seem likely suspects.
How did you configure your GoIP and why?
What do your relevant sip.conf section(s) look like?
What does your dial command look like?
So far, all I’ve got out of it is a 503 Declined.
2 thoughts on - A Bit OT – Configure GoIP For Asterisk
Have you tried http://www.hybertone.com/en/solutionsClass.asp?Idx
Antony.
Thanks, but no joy.
When I first power up the box, ‘sip show peers’ shows:
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
goip/goip (Unspecified) D Yes Yes 0 OK (5 ms)
But then a few seconds later it shows:
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
goip/goip (Unspecified) D Yes Yes 0 UNKNOWN
Every few seconds I get ’empty’ SIP debug messages on the console like:
<--- SIP read from UDP:192.168.0.51:5087 --->
<------------->
Wireshark says they’re only 3 bytes long and contain ‘SIP’.
When I dial, I get:
— Executing [*@newline:6] Dial(“SIP/poly-e637-000000cb”, “sip/goip/xxxxxxxxxx”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
No SIP messages are displayed — I’m guessing that’s a result of the ‘status UNKNOWN’.