[Asterisk-Users] howto (Also SIP VIA header)

[Tjardick van der Kraan]

Well can't help you with everything but at least i noticed your extension is
incorrect


----- Original Message -----
From: "Jörg Bauer/Denic" <bauer at denic.de>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, April 22, 2003 5:19 PM
Subject: [Asterisk-Users] howto


> my config:
> exten => .,1,Dial(SIP/${EXTEN})       // whatever you get....try to
> establish a SIP-session ......
>
> but this expands to Dial(SIP/test)    !!!  Domain is gone !!!!

This is because your extension is missing the _ which tells Asterisk to look
for a numer pattern.

so replace it with this and it should work:

exten => _.,1,Dial(SIP/${EXTEN})

I also asked (but haven't had any answer so far) if it would be possiible to
use the VIA SIP-header.

SIPphone <=> Asterisk <=> Firewall <=> SIPphone (or another SIPservice for
that matter)

It is very easy to map 5060 UPD (and TCP) to the Asterisk box, together with
a range off for example 100 ports of RTP (8000-8100).

After that the sip.conf might have something like:

rtprange=800-8100
outsidehost=firewall.ourdomain.com

That way Asterisk can add the outsidehost in the VIA header and the other
side knows where to send the packets.

I have send an email about this before to the list but i have know idea how
can help me develop these settings into a patch for asterisk. If anyone
would like to help be, please contact me (off list) and i'll get you some
example SIP packets that support this option (Xlite phone, PingTel etc).

Greetings,

Tjardick