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

[Stephen R. Besch]

Sorry about the post to the wrong level of the thread, but something was 
wrong with the first copy of the message (i.e., my mail reader wouldn't 
display it). Comments are inline.

>I tried Stephen advice and it did not work. I stil got the 404 error

>>[general]
>>dtmfmode=rfc2833

This does not match the selection used in your phone, and ironically, is 
the only choice that does not seem to work on the GS phones. Use inband 
or info and make sure that you set the phone the same way.

>;[snomsip]
>;type=friend
>;secret=blah

....

>>;[pingtel]
>>;[cisco]
>>;[cisco1]

....

You might consider deleting all of these unused bits from your file, or 
at least from the email before you send it. If you need them later, you 
can always copy and paste them back from a reference copy of the file.

>>[1001]
>>type = friend
>>context = default
>>secret = gol
>>host = dynamic

Unless you have a good reason for using the dynamic option, I would not 
use it.  In your case, the phone's IP is "Hardwired", and private to 
boot. Just put the IP in after the host=. You also avoid the (possibly 
still present) grandstream bug which loses registrations from time to time.

>>callerid = "STREAM-1001" <1001>
>>;dtfmmode=inband

Ironically, this is what you used on the phons. Why is it commented here?

>>canreinvite=no
>>defaultip=192.168.0.105
>>
>>
>>[1002]

Same for phone 2


>>
>>This is the configuration of my SIP-phones:

....

>>outboundproxy=null
>>outboundproxy_port=null

If all else fails, put your server IP in here! Use default port

....

>>registration_expiration=10

You may find registration to be a problem with the GS. See comments above.

....

>>send_dtmf=in-audio

!!!!This must match the entry in sip.conf (In the GS world, in-audio = 
inband)

Sincerely,

Stephen R. Besch