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...: (phones 11 and 12 match 10, only different defaultip,username,secret Any idea what I miss ? sip.conf: [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes musicclass=default [10] host=dynamic defaultip=192.168.1.91 type=friend username=10 secret=secret nat=yes qualify=yes context=terrazur callgroup=2 pickupgroup=2 extensions.conf [general] static=yes writeprotect=no [macro-stdexten] include => parkedcalls exten => s,1,DBget(temp=CFIM/${ARG1}) exten => s,102,Dial(${ARG1},,Ttr) [remote] include => parkedcalls ;Incoming lines. exten => 31XXXXXXXXX,1,SetCallerID(${CALL...

...one annoying problem tho. There is an audio delay on outgoing/incoming calls. The delay is 2 to 3 seconds. Internal calls from sip to sip is no problem at all. Any idea what is wrong with our setup ? If you need more info and/or config files please ask. Met vriendelijke groet, Michiel van Baak Terrazur

I installed bristuff0.20-RC3 (which includes * 1.0.3 stable) It works fine until I disconnect the phone jack for the ISDN line. Even after plug it back in asterisk still reports that it could not create a zap channel when I try to dial out and the line gives an engaged tone when I try to dial. Re-starting asterisk doesn't solve this, I have to stop asterisk, unload the modules, reload

Hi I'm trying to deploy asterisk, but I can't seem to find documentation for the TFTP server to run the cisco 7960 ip phones' I was told before that you need it and it could run on linux. Thank You ,jm

My router (1605R) currently does not support QoS. Is there any open source software available so that I can set one up before the router? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/38b6e3f2/attachment.htm

How do I set asterisk not to answer incoming PSTN POTS calls? I want to be able to use the line for outgoing calls only. -Thanks Tim