[Asterisk-Users] SIP.CONF "Allow=All" do not work!

[Astrit]

Hi I'm also using asterisk 1.0.0
Look how I did this :


In Sip.conf 

[general]
port=5060
bindaddr=0.0.0.0
;videosupport=yes  ; Gives an error: "process_sdp: Error in codec string
'ideo 0'"
disallow=all
allow=alaw
context=lan-phones

[x.y.z.w]
context=pstn-incoming
type=friend
host=z.y.z.w ; IP address of Cisco gateway
dtmfmode=rfc2833
allow=all
;allow=alaw 

In extension.conf

[lan-phones]
include => sip

[sip]
exten => _[2]XX,1,NoOp(^Ócall for ^Ó${EXTEN})
exten => _[2]XX,2,Dial(SIP/${EXTEN},60,tr)
exten => _[2]XX,3,Congestion

[pstn-incoming]
include => lan-phones

And it works fine ...

             

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Goran Dj
Sent: Thursday, October 07, 2004 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP.CONF "Allow=All" do not work!

In sip.conf, Allow=All stopping all sounds!

When I comment out this command, everything is OK.
I can Allow all codecs one by one, but Allow=All produce same consequences
as Disallow=All.

I have Asterisk 1.0.0. Is this a bug? 


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