El Thu, Jun 28 de 2007 a las 20:18 +0530, [EMAIL PROTECTED] comentaba: > Hi, > I am trying to establish call through sip phone between two PC > connected to linux box on which asterisk server is running > > 1st PC is having IP Adress : 192.168.1.149 > 2nd PC is having IP Adress : 192.168.1.53 > > Now, I am tying to dial from 1st PC to 2nd PC > > I am trying to dial from 1st PC to 2nd PC through asterisk server > The problem is 1st PC is calling directly to 2nd PC not through > asterisk server > > I am doing the following additions in configuration files > > 1) sip.conf > > [general] > context=sip > bindport=5060 > bindaddr=0.0.0.0 > > [phone1] > type=friend > host=192.168.1.149 > port=5060 > nat=yes > dtmfmode=rfc2833 > context=sip > > [phone2] > type=friend > host=192.168.1.53 > port=5060 > nat=yes > dtmfmode=rfc2833 > context=sip > > 2) extensions.conf > exten => 11,1,Dial(SIP/phone2,20,tr) > > Now, I am calling from sip phone1 by name [EMAIL PROTECTED]
I guess thats why the phones are talking directly: [EMAIL PROTECTED] Either call extension '11' from phone1 or add a extension named 'phone2' to extensions.conf and call that extension ('phone2') without the ip address. Make sure your softphones are correctly configured: sip proxy address (* address), username, etc. Btw, both devices in sip.conf are declared as 'friend'.. thus you must specify a secret (and optionally a username): [phone2] type=friend username=phone2 secret=qwerty host=192.168.1.53 port=5060 nat=yes dtmfmode=rfc2833 context=sip > It is not being called through asterisk server running on linux m/c. It > is calling directly. As, I am running "sip debub" but no packet dumping is > taking place. Can anybody will tell me the error I am doing. > Thanx and regards > sanchal > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users