[Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP

[David J Carter]

I use GS 101 & 102, have a look at my configs at
http://www.codepipe.com/id25.htm .

Hope they help.


Dave

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Stephen R.
Besch
Sent: 23 March 2004 20:22
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf
phone HELP


--snip--

> I am having trouble setting the /etc/asterisk/sip.conf file.
> This is my file:
>

1) Add in the [general] section:

disallow=all
allow=ulaw
allow=alaw
allow=any other codec that you want to (or can) support.

While some have found that this must be specified for each and every
phone, I have found that it works fine specified just once in the
general section.

> [243075]
> type = friend
> context = default
> secret = gol
> host = dynamic
> callerid = fono75 <243075>
>
2) Include dtfmmode=info or inband and match to phone's setting

3) I may have been too tired at the time, but once I tried using long
extensions (more than 5 digits) and could not make them work either -
same error you are getting.  I would limit your extensions to 4 digits
and see if it helps.

4) You may also need to add

canreinvite=no

to each phone definition.

>
> and our SIP phones configuration are the following:
>
>  SIP Server: 192.168.0.102
>
>  Outbound Proxy:  <Empty>
>

5) I would set this to be the same as the server if you want to make
outbound calls.

Hope this helps

Stephen R. Besch
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