Hello,
This is my first time using asterisk/freepbx. I will try and set this out as clearly as I can if you need anymore clarification please ask
I have just done a fresh install of AsteriskNow 2.0.2 64bit.
I am using 2 Yealink SIP-T20P’s.
I have been following this guide to instal/configure it:
https://wiki.asterisk.org/wiki/display/AST/Installing+AsteriskNOW
I’ve set up SIP accounts and extensions.
The phones have successfully registered.
I tried to make a test call from one phone to the other and vice versa and no matter which way I try to make the call it is declined.
When I go onto the asterisk cli and try to make this call the output I get is:
== Using SIP RTP TOS bit 184
== Using SIP RTP CoS mark 5
== Spawn extension (users, 6001, 1) exited non-zero on ‘SIP/demo-alice-00000011’
Does anyone have any idea why this would be happening?
Thank you in advanced,
Jethro