[asterisk-users] Query

[Keshav K.]

For this you have to make entry in sip.conf.
it will be better if you use host=dynamic in both the phones in sip.conf

and what is  the IP you are putting   in phones which are on your PC.
Also check that your both sip phones which are on PC, are sending requestr to asterisk server or not.

Kesh.

sanchal.singh at alliance-infotech.com wrote: 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

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 11 at 192.168.1.53
It is not being called through asterisk server running on linux m/c. as no packet dumping us taking palce. As, I am running "sip debub"  no messages are seen on screen.
    What additional routing informations are to be added to sip.conf, inorder to make it work .
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


       
---------------------------------
Luggage? GPS? Comic books? 
Check out fitting  gifts for grads at Yahoo! Search.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070630/4339d402/attachment-0001.htm