[Asterisk-Users] Asterisk + GrandStream SIP phones

[pesb]

Hi,
Thanks for the help. You were correct. There was some data missing in the 
extension.conf file
I was able to call one SIP phone from the other. I was even able to call an 
H323 IP phone registered to the gnugk GK (It has Asterisk registered to him 
as a GW).
But, I have another problem rigth now.
All the RTP Data Flow is passing through the Asterisk Proxy, which is a bad 
thing if I want to have many SIP phones in my system.
How can I configure the SIP phone in order to make all RTP data flow directly 
from one SIP phone to the other? 
And, how can I configure it in order to make all RTP data flow directly from 
one SIP phone to the H323 IP phone (the one registered to my gnugk GK)?
I would also like to be able to make calls from a SIP phone to the other SIP 
phone, but instead of having the ASTERISK PBX authorizing the calls, it would 
be the H323 GK the one that would authorize calls. How can I do this?

Thanks again

On Monday 29 March 2004 15:58, David J Carter wrote:
> Try this small extensions.conf
>
> Don't think I have missed owt.
>
> My config files are here, you just need to add your own extension numbers.
>
> http://www.codepipe.com/id25.htm
>
> Dave
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of pesb
> Sent: 29 March 2004 19:26
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Asterisk + GrandStream SIP phones
>
>
> -This is my 'sip.conf' file:
>
> ;*************************************************************
> ;
> ; 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
> tos=184
> maxexpirey=3600         ; Max length of incoming registration we allow
> defaultexpirey=120              ; Default length of incoming/outoing
> registration
> disallow=all                    ; Disallow all codecs
> allow=ulaw                      ; Allow codecs in order of preference
> allow=alaw
>
>
> [1004]
> type=friend
> username=1004
> secret=
> reinvite=no
> canreinvite=no
> host=dynamic
> dtmfmode=inband
> mailbox=1004
> nat=1
> disallow=all
> allow=ulaw
> allow=alaw
>
> [1005]
> type=friend
> username=1005
> secret=
> reinvite=no
> canreinvite=no
> host=dynamic
> dtmfmode=inband
> mailbox=1005
> nat=1
> disallow=all
> allow=ulaw
> allow=alaw
>
> ;*******************************************
>
>
> -And this is the basic seting of my two GrandStream SIP phones:
>
> ***************[1005]****************
> IP Address:192.168.0.105
> Subnet Mask:255.255.255.0
> SIP Server: 192.168.0.103
> Outbound Proxy:<empty>
> SIP User ID:1005
> Authenticate ID:1005
> Authenticate Password:123
> Name:1005
>
> Preferred Vocoder:
> choice 1: PCMU
> choice 2: PCMA
> choice 3: G723
> choice 4: G729
> choice 5: G726-32
> choice 6: G728
>
> G723 rate:  6.3kbps
> Silence Suppression:No
> Send DTMF:in-audio
>
> ***************[1004]****************
> IP Address:192.168.0.104
> Subnet Mask:255.255.255.0
> SIP Server: 192.168.0.103
> Outbound Proxy:<empty>
> SIP User ID:1004
> Authenticate ID:1004
> Authenticate Password:123
> Name:1004
>
> Preferred Vocoder:
> choice 1: PCMU
> choice 2: PCMA
> choice 3: G723
> choice 4: G729
> choice 5: G726-32
> choice 6: G728
>
> G723 rate:  6.3kbps
> Silence Suppression:No
> Send DTMF:in-audio
>
> ******************************
>
> I have 2 SIP GrandStream phones, both phones are correctly registered to
> the Asterisk server. But, when I try to make a call from registered phone
> '1005' to registered phone '1004', dialing 1004, Asterisk responds with the
> 'Status:
> 404 Not Found' message.
> How do I have to dial? What else do I need to set?
> Find attached my traffic captured on ethereal.