[Asterisk-Users] i am missing something!

Jay Milk jay at skimmilk.net
Wed Mar 9 00:44:41 MST 2005


You'll need canreinvite=no to each sip section in sip.conf, if you want
* to stay in the loop.

> -----Original Message-----
> From: Adnan Ahmed [mailto:asteriskster at gmail.com] 
> Sent: Wednesday, March 09, 2005 1:14 AM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] i am missing something!
> 
> 
> Hello ppl,
> At initial level i configure asterisk woth only soft phones 
> ,in which one at windows machine and other is linux i am 
> using windows messenger and linphone respectively both phones 
> registered with asterisk respectively problem is that they 
> bypass asterisk on call when i send request from linphone to 
> messenger request shown on messenger but on asterisk console 
> nothing to and also if i send request from messenger to 
> linphone it doesn't recognized at all my config are:




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