[asterisk-users] Query

[Victor Toofic]

El Thu, Jun 28 de 2007 a las 20:18 +0530, sanchal.singh at alliance-infotech.com 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 phone2 at 192.168.1.53

I guess thats why the phones are talking directly: phone2 at 192.168.1.53

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
>