Ahmed [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 09, 2005 1:14 AM
To: asterisk-users@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
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
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:[EMAIL PROTECTED]
Sent: Wednesday, March 09, 2005 1:14 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] i am missing something
Dont get caught by the same thing which had me ripping my hair out!
I had installed Fedora core 2 on a box and forgot that it had installed iptables
firewall!
Type iptables -L and see if there are any rules? iptables -F will
flush them for the time
being, then try again.
It worked for me, wow how
Mike Dent wrote:
Dont get caught by the same thing which had me ripping my hair out!
I had installed Fedora core 2 on a box and forgot that it had installed iptables
firewall!
Type iptables -L and see if there are any rules? iptables -F will
flush them for the time
being, then try again.
It worked
Did you try the iptables -L as I suggested though?
It's probably still present in Debian.
Mike
On Mon, 22 Nov 2004 02:47:53 +0500, Adnan Ahmed [EMAIL PROTECTED] wrote:
Mike Dent wrote:
Dont get caught by the same thing which had me ripping my hair out!
I had installed Fedora core 2 on a
Mike Dent wrote:
Did you try the iptables -L as I suggested though?
It's probably still present in Debian.
Mike
On Mon, 22 Nov 2004 02:47:53 +0500, Adnan Ahmed [EMAIL PROTECTED] wrote:
Mike Dent wrote:
Dont get caught by the same thing which had me ripping my hair out!
I had installed
Adnan Ahmed wrote:
hi,
I am not registered my SIP Phone with Asterisk i spend almost one
day but find no luck my configs are.
Please post console log with errormessage..
My guess is the host=192.168.10.195 definition and the use of
context=sip not matching the dialplan.
/Soren
Are you sure your phone is registered? It probably is. In sip.conf you
you have context=sip and in sip you have only 101, So all you can dial
from your phone is 101.
You might want to put under sip context include = outgoing and try
making an outgoing call
On Wed, 24 Nov 2004 13:43:58 +0100,
hi,
I am not registered my SIP Phone with Asterisk i spend almost one day
but find no luck my configs are.
sip.conf
[general]
port=5060
bindaddr=192.168.10.195
disallow=all
allow=alaw
allow=ulaw
[101]
username=101
type=friend
secret=1234
host=192.168.10.195
context=sip
callerid=101101
el Flynn wrote:
Adnan Ahmed wrote:
hi,
I am not registered my SIP Phone with Asterisk i spend almost one
day but find no luck my configs are.
snip
*clisip show peers
Name/UsernameHost
Mask Port Status
101/101
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