[Asterisk-Users] Grandstream 101

[Chris Albertson]

--- dkwok <dkwok at iware.com.au> wrote:
> Just got GS 101 phone and plugged into the network.

Peoplehere complain about these phones but I don't seem
to have a problem, well not after getting them set up
correctly.  I'm running with
Software Version:  	  Program--1.0.4.39    Bootloader--1.0.0.13   
HTML--1.0.0.20

> 
> Got ip setup however, the following problems arise:
> 
> 1. when dialing an extension, I cannot further send any key tone to 
> Asterisk.

I'm using "SIP info" also with "payload type" set to "101"

> 2. there is no sound coming from the other end.

For some reason I found I had to place the disallow=...allow=...
stuff under [gs] putting it in [General] didn't seem to do the trick.
I also put "reinvite=no" in [gs] 

I once had sound going only one way due to t stupid error in
my firewall config.  I was purposfully droping packets and logging
each one of them.  Are you running firewall software on your
* server?

"ethereal" or other ethernet sniffing software is usfull to debug
this kind of stuff

> 
> I have a sip.conf setup for GS:
> [General]
> disallow=all
> allow=ulaw
> allow=alaw
> 
> [gs]
> canreinvite=no
> dtmfmode=info
> 
> In the GS101 setting
> rtp port = 5004
> sip port = 5060
> dtmf = sip info
> codec = pcmu
> codec = pcma
> 
> Any pointer of a sample of config file would be most appreciate.
> 
> -- 
> David Kwok
> 
> Iaxtel/FWD # 17001813482 ext 1002
> 

> ATTACHMENT part 2 application/x-pkcs7-signature name=smime.p7s



=====
Chris Albertson
  Home:   310-376-1029  chrisalbertson90278 at yahoo.com
  Cell:   310-990-7550
  Office: 310-336-5189  Christopher.J.Albertson at aero.org
  KG6OMK

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