Soft phones not conencting after fresh vicidial install

[davesdatasystems]

OK for the stats

Opensuse
Vicibox redux 3.1.10
Asterisk 1.4.39
Office side server
AMD 4 core 4gig processer/4 gigs of ram desktop computer

Ok, I have ran into a huge issue, and i hope i can see a answer by tomorrow morning, but here is what happened, the dialer crashed, and completely corrupted, i attempt to fix, but could not fix it, so i had to do a fresh install, i first thought the error was in Asterisk, but i do not feel it is anymore, i think i messed up the network configuration on it. SO i am going to post as good as i can what i have in YAST LAN, and see if one of you guys that are much smarter then me can spot the issue with it

Network card setup>staticlly assigned ip address ticked> 172.16.0.100 /24
firewall is disabled
Hostname is listed as simply dialer

the name servers as instructed by brighthouyse are
65.32.1.65
65.32.1.70

And the place where i feel i may of made the error, Netowrk settings>routing Default IPv4 Gateway
172.16.0.1 is what i have set in there. enbale IP forwarding is not ticked.

ALso just to show the error that i am getting, i will post it here, but i am sure it is a small part of a more major issue


[May 26 23:45:17] NOTICE[2960]: chan_sip.c:16711 handle_request_subscribe: Received SIP subscribe for peer without mailbox: (null)
[May 26 23:45:17] NOTICE[2960]: chan_sip.c:16835 handle_request_register: Registration from '<sip:cc500@XXX.XXX.XXX.XXX;transport=UDP>' failed for '173.171.124.173' - No matching peer found

I will post any other info i need to post, however, i think it is in the network confiration, so please help, or please ask for any other info that can be used to help with this

[boybawang]

check the passwords of your softphone

[davesdatasystems]

I did check that, however i did have a resolve to the issue. I dont know why, but several files had to be rewritten to work right, I dont know what is different between now and 6 months ago, but now it dont seem to work the same. Before it was install, edit sip file, plug and the agents would log in and go. When i get the info, i will pass it on, but it was a lot of editing to make it work the way it was suppose to

[rrb555]

try upgrading to latest svn? or backup database and install a new one then upload ur database on it. this is far more easy

[davesdatasystems]

thanks rrb555, we are building a primary dialer using this as the backup, but we learned many lessons from this install and i will be using the new build and the new svn on the dialer when we do it.

[rrb555]

be sure to note that latest SVN trunk has some issues check this one viewtopic.php?f=2&t=25262

[davesdatasystems]

I did notice that, i went to a older build, to resolve some of the issues, but i did run across that, but that post will help me is resolving that field issue