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

[Stephen R. Besch]

--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