Hey,
To save money, we have decided to go with an asterisk server and a SIP Trunk for our phone system. I used AsteriskNow 1.7.0 32bit.
Details:
FreePBX 2.7.0.5
CentOS 5
Asterisk 1.6
I have created a SIP trunk for use with video conferencing equipment and I have them registered to each other. My problem is I cannot dial between the Asterisk and the Video Conferencing equipment (VCSC).
I get ‘call denied’ when trying to call devices registered to Asterisk from a device registered to my VCSC.
This is the log from Asterisk:
[Aug 10 09:24:18] VERBOSE[2615] netsock.c: == Using SIP RTP TOS bits 184
[Aug 10 09:24:18] VERBOSE[2615] netsock.c: == Using SIP RTP CoS mark 5
[Aug 10 09:24:18] NOTICE[2615] chan_sip.c: Failed to authenticate device “My Name” sip:[email protected];tag=41a6d436b2270105
It confuses me why it is trying to authenticate a client registered to my VCSC. Shouldn’t it just accept anything from my VCSC, since it is registered?
I am new to Asterisk, so any help would be greatly appreciated!
Thanks,
Robert
PS
if you need more information I am glad to post it.