URI Dialing Inbound


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.

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!




if you need more information I am glad to post it.