search for: 2145551212

...snip from my sip.conf: sip.conf [maxtnt] type=friend host=10.10.14.131 insecure=very dtmfmode=inband callerid="MaxTNT" <maxtnt> context=trunktntin qualify=yes reinvite=no canreinvite=no disallow=all allow=ulaw debug lab1*CLI> <-- SIP read from 10.10.14.131:5060: INVITE sip:2145551212@10.10.14.121:5060;user=phone SIP/2.0 t: <sip:2145551212@10.10.14.121:5060;user=phone> f: "NO CID NAME" <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a Remote-Party-Id: "NO CID NAME" <sip:1239@10.10.14.131:5060;user=phone>;screen=no;id-t...

I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @<proxy-addr>... So the callerID on the phone looks like: 2145551212@10.10.10.10 which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? 7.5 and 7.4 don't do this. I just rolled back and 7.5 works as expect...

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

...d. I've searched all over the Wiki and can't find out why I am missing this. When I do a playback(demo-congrats) instead of the Dial command, demo audio plays back just fine. What am I missing and where can I look more to find the problem? Extensions.conf: [Provider_T14] exten => 2145551212,1,Dial(SIP/User1,30,r) CLI debug: *CLI> -- Starting simple switch on 'Zap/73-1' -- Executing Dial("Zap/73-1", "SIP/User1|30|r") in new stack -- Called User1 == Spawn extension (Provider_T14, 2145551212, 1) exited non-zero on 'Zap/73-1' --...