[Asterisk-Users] Grandstream Setup

[Stephen R. Besch]

This config is fine, with the exception of one thing.  While I realize 
that I may be the lone dissenter regarding the SEND DTMF option, I have 
found that all three options work fine as long as * is the only 
listener.  However, as soon as you start talking to the outside world, 
the only thing that works for me, in all cases, is the inband option 
(inband to *, in-audio to the Grandstream).  As soon as I make any calls 
to another IVR, the SIP Info and RFC2833 options fail - that is, they do 
nothing!  I don't know whose problem this is, but since * works so well 
with the inband option, I can see no rational reason to worry about it.  
I just use the inband option on all my phones.

One other note. The most critical thing as far as I can tell, for 
getting Grandstream to register, is to have the extension name (the 
thing in the []'s in sip.conf) exactly match both the SIP UserID and the 
Authenticate ID in the Grandstream setup.  The username option in * 
makes no difference whatsoever with the Grandstream.  If it still 
doesn't work, make sure that you can ping the phone from the * box. If 
you can't, then you have to fix your network setup first.

Stephen R. Besch

rnc Info Lists wrote:

>My config that works for number 1 is below.   Everything works including
>the voice mail waiting light. All of this for * was copied from or based
>on:
>http://www.automated.it/guidetoasterisk.htm.  This is an EXCELLENT getting
>started site.   Can't help you with #2 but am sure others can.
>
>sip.conf for extension 2000
>[2000]
>
>type=friend           ; This device takes and makes calls
>username=2000         ; Username on device
>secret=9overthruster7 ; Password for device
>host=dynamic          ; This host is not on the same IP addr every time
>context=from-sip      ; Inbound calls from this host go here
>mailbox=2000           ; Activate the message waiting light if this
>                      ; voicemailbox has messages in it
>
>
>extensions.conf
>
>exten => 2000,1,Dial(SIP/2000,20)
>exten => 2000,2,Voicemail(u2000)
>
>
>Budge Tone config:
>
>SIP Server:  192.168.0.110  (my * box)
>SIP Userid:  2000 (userid is same as extension
>Authenticate ID: 2000
>Authenticate password:  9overthruster7
>Send DTMF:  Via SIP info   (in order for the dtmf to be recognized by
>voicemail)
>
>  
>
>>Hi People,
>>
>>Ok i've tried everything I can think of but cant get this to work.
>>
>>Can someone please give me an example of their sip.conf settings and also
>>the
>>details of the settings in their grandstream phone to allow:
>>1. Grandstream phone to register with asterisk when on same lan.
>>2. Grandstream phone to register with asterisk when phone is behind a nat.
>>
>>Regards,
>>Aaron.
>>
>>
>>
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