asterisk users - Jan 2004 - SIP register/auth with Grandstream BudgeTone-100

Hello,

I have a problem with asterisk and Grandstream BudgeTone-100.
With default configuration everything works (in anonymous mode and fixed
IP), but if Im trying to enable registering, it dos not work.
I used 'sip debug' and verbose level 10, nothing happens if I switch
telephone on (no messages about bad auth etc). As I understood, after
switching phone on at first it will try to register in asterisk if Im
trying to call somewhere.

I searched in list-archive and I didnt found that anybody else has this
kind of problem. I read also:
http://lists.digium.com/pipermail/asterisk-users/2003-June/013288.html
and I did so.

sip.conf
---------
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = default               ; Default for incoming calls
disallow=all                    ; Disallow all codecs
allow=g729

[cisco]
context=in
type=friend
insecure=yes
host=<removed>
dtmfmode=rfc2833

[grandstream1]
type=friend
secret=grandstream1
host=dynamic
context=class1
dtmfmode=rfc2833

[grandstream2]
type=friend
secret=grandstream2
nat=yes
host=dynamic
context=class1
dtmfmode=rfc2833

Asterisk ver: Asterisk CVS-01/22/04-18:13:23

Grandstream ver: Program--1.0.3.81    Bootloader--1.0.0.7    HTML--1.0.0.18

* And as I mentioned before, without registration and with static IP
everything works, it seems, that something is misconfigured in my setup
for authentication or this phone firmware is buggy? (but its latest, I
checked www.grandstream.com)



____________________________
Best Regards:
               Key Aavoja




/* Never argue with an idiot. They drag you down to their level, then beat
you with experience.*/

Key, 

I've been playing with the Grandstreams for some weeks; one good way to see
the registration messages it to monitor the network with Ethereal. (packet
sniffer).  You'll see the SIP messages coming and going, with complete
decoding. This works pretty much as predicted when using VOCAL. (another SIP
server) 

That said, I recently asked about registration with Asterisk from the
Grandstreams, and received a reply that if the phones have a fixed IP
address they do not "register" as they are already accounted for in
the
sip.conf by the user name/number assigned to the phone, and indeed the
phones work fine (at least inside the firewall).  
Other soureces have described Asterisk as having "limited or not fully
implemented SIP support"... but I have never been able to determine to what
extent SIP support is lacking. 

I was able to get the Grandstreams running only recently with Asterisk... My
sip.conf for the Grandstreams has several differences from yours: For a
grandstream phone with an assigned number of 1000 and a fixed IP address of
192.168.0.160 I have the following: 


[1000]
type=friend
username=1000
host=dynamic
reinvite=no
canreinvite=no
qualify=300
callerid="Larry's Desk" <1000>
mailbox=1000
nat=no
dtmfmode=info
disallow=all
allow=ulaw
allow=mlaw

See the message I sent to Tom Scott on 1/21 for a little more information.  

I would very much like to put together a set of instructions for the
Grandstream phones+asterisk that could expand on the scattered information
found on the wiki/web/draft manual and this list. 

And let us know what you find!

     
-- L

Key Aavoja wrote:
> Hello,
> 
> I have a problem with asterisk and Grandstream BudgeTone-100.
> With default configuration everything works (in anonymous mode and fixed
> IP), but if Im trying to enable registering, it dos not work.
> I used 'sip debug' and verbose level 10, nothing happens if I
switch
> telephone on (no messages about bad auth etc). As I understood, after
> switching phone on at first it will try to register in asterisk
Yes
  if Im
> trying to call somewhere.
Registers before any calls are made.

Probably your extension name and registration data don't match.  Here is 
my SIP config and a list of the GS phone settings:

[exten106]
type=friend
context=administrator
callerid=<829-3289 106>
username=sbesch
host=dynamic
dtmfmode=info		;or inband if you prefer
secret=yourpassword
qualify=5000
mailbox=106
canreinvite=no		;As long as the phones are NAT'ed

The caller ID only means something to our internal extensions, since the 
phone company will not let me set callerid data. I don't think that the 
username is needed. I use it because it shows up in the CLI response to 
SIP SHOW PEERS and helps me identify the phone. The important bits are 
that the extension name (the part in "[]") and the secret match the
data
in the phone setup:

Sip User ID: exten106
Authenticate ID: exten106
Aithentication Password: yourpassword
Sip Registration: Yes
Send DTMF: via SIP Info


Don't make the mistake of thinking that the username entry in SIP.conf 
has anything to do with the Authenticate ID. IT doesn't. The only thing 
that works is to set the User ID and the Authenticate ID to the same thing.

You may find that the GS phones will dissappear after a while if you use 
dynamic registration. Alas, this is a bug in the GS firmware(1.0.3.81). 
I don't know if it has been fixed in later releases. I am not willing to 
update my phones until the firmware gets much more stable - they are all 
working and my philosophy is that if it ain't broke, don't fix it - so I
haven't been able to test this. If your phones have fixed addresses, you 
might as well specify the IP addresses in SIP.conf and preemptively 
avoid the problem of the GS registrations dissappearing.

Hi,Key Aavoja,
    Have you successfully registed to * with secret specificated?
Regards.

bfrac

----- Original Message ----- 
From: "Key Aavoja" <key@neoon.com>
To: <asterisk-users@lists.digium.com>
Sent: Friday, January 23, 2004 2:00 AM
Subject: [Asterisk-Users] SIP register/auth with Grandstream BudgeTone-100

> Hello,
>
> I have a problem with asterisk and Grandstream BudgeTone-100.
> With default configuration everything works (in anonymous mode and fixed
> IP), but if Im trying to enable registering, it dos not work.
> I used 'sip debug' and verbose level 10, nothing happens if I
switch
> telephone on (no messages about bad auth etc). As I understood, after
> switching phone on at first it will try to register in asterisk if Im
> trying to call somewhere.
>
> I searched in list-archive and I didnt found that anybody else has this
> kind of problem. I read also:
> http://lists.digium.com/pipermail/asterisk-users/2003-June/013288.html
> and I did so.
>
> sip.conf
> ---------
> [general]
> port = 5060                     ; Port to bind to
> bindaddr = 0.0.0.0              ; Address to bind to
> context = default               ; Default for incoming calls
> disallow=all                    ; Disallow all codecs
> allow=g729
>
> [cisco]
> context=in
> type=friend
> insecure=yes
> host=<removed>
> dtmfmode=rfc2833
>
> [grandstream1]
> type=friend
> secret=grandstream1
> host=dynamic
> context=class1
> dtmfmode=rfc2833
>
> [grandstream2]
> type=friend
> secret=grandstream2
> nat=yes
> host=dynamic
> context=class1
> dtmfmode=rfc2833
>
> Asterisk ver: Asterisk CVS-01/22/04-18:13:23
>
> Grandstream ver: Program--1.0.3.81    Bootloader--1.0.0.7
HTML--1.0.0.18
>
> * And as I mentioned before, without registration and with static IP
> everything works, it seems, that something is misconfigured in my setup
> for authentication or this phone firmware is buggy? (but its latest, I
> checked www.grandstream.com)
>
>
>
> ____________________________
> Best Regards:
>                Key Aavoja
>
>
>
>
> /* Never argue with an idiot. They drag you down to their level, then beat
> you with experience.*/
>
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