[Asterisk-Users] simple question on sip.conf

[David J Carter]

Hi all

Disregard my last post I replied to the wrong e-mail, I should have replied
to an off list e-mail.

That will teach me not to put my glasses on.

Dave

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Olle E. Johansson
Sent: 12 December 2003 08:01
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] simple question on sip.conf

SW wrote:

> Hi folks,
>
> I want to fix hole in my asterisk set up.
>
> I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN,
> Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go
> 'other' places. This senario works fine.
>
> Now the issue is someone else running a vocal or another SIP proxy can
> redirect his calls to my * as well. Those calls two will come through
> general section of the sip.conf and could land on the PSTN as well. (do
not
> ask my *'s IP address folks, I am not going to run a free PSTN g/w :)
>
> So, how do I prevent other than my own proxy to use the general section of
> the sip.conf file ? As a mater of fact all calls from fwd and iconnect two
> land on the general section.
>
> Hope some one can shead some light.
>
> Cheers
>
> SW
>
> Here is my sip.conf
>
> [general]
> ;calls arrive from sip lands here
> port=5060
> context=default-in
Change this to
   context=forbidden
And define "forbidden" context in extensions conf to whatever you like,
example

exten => s,1,playback(tt-monkeys)
exten => s,2,hangup

> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> maxexpirey=180
> defaultexpirey=160
> ;Connect to Free World Dialup (no NAT)
> register=61358:xxxxxxx at fwd.pulver.com/61358
This one does not have a defined context.
[fwd.pulver.com] section missing

> ;Connect to iconnect
> register=15108688610:xxxx at sipauth.deltathree.com/15108688610
> canreinvite=no
Same here.

>
>
>
> [iconnect]
> ;incoming does not land here, why ? outgoing is fine
> type=friend
> secret=xxxx
> username=yyyyyyyy
> host=sipauth.deltathree.com
> dtmfmode=inband ; required by iconnect
> context=iconnect-in
> canreinvite=no
> allow=alaw
> allow=ulaw
> allow=g729
>
>
> [fwd]
> ;incoming does not land here, why ?  outgoing is fine
> type=friend
> secret=xxxxxxx
> username=61358
> host=fwd.pulver.com
> context=fwd-in
> allow=alaw
> allow=ulaw
>
> [vocal]
> ;used when dialed in from vocal **** not working  ****
> type=friend
> host=ip of vocal server
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> port=5060
> canreinvite=no
> context=vocal-in
>
>
> [vocal-out]
> ;used to dial out to vocal
> type=friend
> host=ip of vocal server
> allow=g729
> allow=ulaw
> allow=alaw
> port=5065
> canreinvite=no
>
> [6300]
> type=friend
> username=6300
> context=intern
> ;secret=blah
> host=dynamic
> ;defaultip=192.168.254.4
> dtmfmode=info
> nat=1
>
> [6301]
> type=friend
> username=6301
> host=dynamic
> dtmfmode=inband
> context=intern
> nat=1
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>


--
*** Olle E. Johansson, oej at edvina.net

Mobile +46 70 593 68 51, Edvina AB, http://www.edvina.net
Runbovägen 10, 192 48 Sollentuna, Sweden
Phone: +46 8 594 78 810, Fax: +46 8 594 78 820


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