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marrandy

2003-Nov-17 19:56 UTC

[Asterisk-Users] Struggling with grandstream sip to asterisk

Hello.

I had grandstream working fine to FWD through my firewall.

Now I want it to talk to the asterisk server.

Did lots of reading, attempts but I keep getting registration errors even 
though I can call to/from the sip phone from an analog phone on a tdm400 
card.

Basically.

grandstream = 192.168.1.70
asterisk = 192.168.1.1

The error I see is ;-

    -- Executing Dial("Zap/2-1", "SIP/206|20") in new stack
    -- Called 206
    -- SIP/206-582e is ringing
NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from 
'<sip:206@192.168.1.1>' failed for '192.168.1.70'

Can anyone give me a hint/clue about what is going on.

A good sip.conf for grandstream with a static address would help.

Regards...Martin

Joe Dennick

2003-Nov-17 20:05 UTC

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[Asterisk-Users] Struggling with grandstream sip to asterisk

I just got my GrandStream working.  The SIP.CONF file probably isn't
your problem except for the 'secret' entry.  Your phone doesn't have
the
proper auth/password combination to successfully register with the
Asterisk Server.  Use the web browser for your GrandStream phone to set
the 'Authenticate ID' and 'Authenticate Password' to match what
you've
configured in your sip.conf file.  My sip.conf looks like this:

	[7007]
	type=friend
	secret=blah
	host=dynamic
	mailbox=7002
	canreinvite=no

The fact that you are using a Static IP Address doesn't matter to the *,
so don't worry about it.

Once configured, I've found my GrandStream to provide decent voice
quality.  It doesn't have all of the features of a more expensive phone,
but then that isn't what you paid for.

Joe Dennick

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of marrandy
Sent: Monday, November 17, 2003 8:56 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Struggling with grandstream sip to asterisk



Hello.

I had grandstream working fine to FWD through my firewall.

Now I want it to talk to the asterisk server.

Did lots of reading, attempts but I keep getting registration errors
even 
though I can call to/from the sip phone from an analog phone on a tdm400

card.

Basically.

grandstream = 192.168.1.70
asterisk = 192.168.1.1

The error I see is ;-

    -- Executing Dial("Zap/2-1", "SIP/206|20") in new stack
    -- Called 206
    -- SIP/206-582e is ringing
NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration
from 
'<sip:206@192.168.1.1>' failed for '192.168.1.70'

Can anyone give me a hint/clue about what is going on.

A good sip.conf for grandstream with a static address would help.

Regards...Martin

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users

Walker Haddock

2003-Nov-17 20:12 UTC

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[Asterisk-Users] Struggling with grandstream sip to asterisk

> Basically.
> 
> grandstream = 192.168.1.70
> asterisk = 192.168.1.1
> 
> The error I see is ;-
> 
>     -- Executing Dial("Zap/2-1", "SIP/206|20") in new
stack
>     -- Called 206
>     -- SIP/206-582e is ringing
> NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration
from
> '<sip:206@192.168.1.1>' failed for '192.168.1.70'
> 
> Can anyone give me a hint/clue about what is going on.
> 
> A good sip.conf for grandstream with a static address would help.; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = default               ; Default for incoming calls
;
[205] ; Conference 2, Grandstream Phone
callerid="Converence 2" <205>
username=205
context=intern
qualify=yes
incominglimit=1
type=friend
insecure=yes
host=192.168.1.70
permit=192.168.0.0/255.255.255.0
dtmfmode=info
canreinvite=no
reinvite=no
callgroup=1
pickupgroup=1
disallow=all
allow=alaw
allow=ulaw

Brian West

2003-Nov-17 20:31 UTC

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[Asterisk-Users] Struggling with grandstream sip to asterisk

Show us your sip.conf entries.. and i'm sure I can point out the error.

bkw

On Mon, 17 Nov 2003, marrandy wrote:
>
> Hello.
>
> I had grandstream working fine to FWD through my firewall.
>
> Now I want it to talk to the asterisk server.
>
> Did lots of reading, attempts but I keep getting registration errors even
> though I can call to/from the sip phone from an analog phone on a tdm400
> card.
>
> Basically.
>
> grandstream = 192.168.1.70
> asterisk = 192.168.1.1
>
> The error I see is ;-
>
>     -- Executing Dial("Zap/2-1", "SIP/206|20") in new
stack
>     -- Called 206
>     -- SIP/206-582e is ringing
> NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration
from
> '<sip:206@192.168.1.1>' failed for '192.168.1.70'
>
> Can anyone give me a hint/clue about what is going on.
>
> A good sip.conf for grandstream with a static address would help.
>
> Regards...Martin
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

Robert Hajime Lanning

2003-Nov-17 23:15 UTC

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[Asterisk-Users] Struggling with grandstream sip to asterisk

<quote who="Walker Haddock">>
> ; SIP Configuration for Asterisk
> ;
> [general]
> port = 5060                     ; Port to bind to
> bindaddr = 0.0.0.0              ; Address to bind to
> context = default               ; Default for incoming calls
> ;
> [205] ; Conference 2, Grandstream Phone
> callerid="Converence 2" <205>
> username=205
> context=intern
> qualify=yes
> incominglimit=1
> type=friend
> insecure=yes
> host=192.168.1.70
> permit=192.168.0.0/255.255.255.0
                 ^
wrong subnet.

> dtmfmode=info
> canreinvite=no
> reinvite=no
> callgroup=1
> pickupgroup=1
> disallow=all
> allow=alaw
> allow=ulaw
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

-- 
END OF LINE

Tom Weeks

2003-Nov-21 11:33 UTC

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[Asterisk-Users] Struggling with grandstream sip to asterisk

Good day,

I have a new Grandstream and am having trouble connecting to *

my software version is the same as below...

I can get it to connect, but am getting  "RTP Read error: Resource
temporarily unavailable" errors whenever I dial...

Tom

----- Original Message ----- 
From: "Dave Cotton" <dcotton@linuxautrement.com>
To: "Asterisk List" <asterisk-users@lists.digium.com>
Sent: Friday, November 21, 2003 10:15 AM
Subject: Re: [Asterisk-Users] Struggling with grandstream sip to asterisk

> On Fri, 2003-11-21 at 17:57, TC wrote:
>
> > I am using the following settings
> >
> > Software Version:    Program--1.0.4.20    Bootloader--1.0.0.12
> > HTML--1.0.0.19
>
> Where is this firmware from? The GS site is still at
> Program--1.0.3.81    Bootloader--1.0.0.7    HTML--1.0.0.18
>
> or is Sipphone ahead of GS?
>
> -- 
> Dave Cotton <dcotton@linuxautrement.com>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

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