Re: [Asterisk-Users] Grandstream Setup

Mon, 13 Oct 2003 07:28:02 -0700

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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