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How can i control the codec for the calls. For example I have 3 SIP phones registered to asterisk The firs two are in the local area network (behind nat)- I want to use g711 between them and to connect directly (canreinvite=yes) and the third is in internet - want all calls to it and from it to use g729 and media to go through asterisk. So if Phone 1 calls Phone 2 the codec to be g711, but when

Hi, If I add an entire disk to a new pool by doing "zpool create", is this reversible? I.e. if there was data on that disk (e.g. it was the sole disk in a zpool in another system) can I get this back or is zpool create destructive? Joubert This message posted from opensolaris.org

...oo. Here's my sip.conf entry for Nikotel (left out the register stuff 'cause it's working): [nikotel] type=friend host=calamar0.nikotel.com username=user secret=pass fromuser=user fromdomain=nikotel.com qualify=yes context=nikotel-incoming insecure=very canreinvite=no promiscredir=yes diallow=all allow=alaw allow=ulaw allow=gsm extension.conf: [nikotel-incoming] exten => 3740525,1,NoOp(Invoming call via nikotel-us) exten => 3740525,2,Dial(IAX2/christoph&SIP/30&${CONSOLE},30) exten => 3740525,3,VoiceMail(u30) exten => 3740525,4,Hangup Thanks for any help, Christoph

Hi, I got my * working fine with FWD at office with 2 extensions, i receive calls and i can make calls thru FWD. I got also my * at home, and i connected it using auth=rsa. From my home, i can make calls using my office iax, but if i try to redirect incomming calls from FWD to my * at home, it rejects the call. I created the pub/key pairs for rsa and its working ok and i just pasted the