Peer gateway

Hi there!

I've got problems making a GXW4108 gateway with asterisk. I followed all the existing tutorials but still, no success. Need your help, I'm completely stuck!

Here is what I want to do:

Link my GXW4108 gateway with Asterisk, so that all my external calls (incoming and outgoing) are going through my gateway. My FXO lines being connected to it. That's it. So for example, when I dial 9 9613 3214 on my IP phone, Asterisk takes the call and make it through my GXW4108 FXO lines! Easy?

Here is what I do so far:

Under sip.conf:

[101]
type=peer
context=outbound_calls
host=192.168.10.27 (GXW4108 address)
insecure=port

Under extensions.conf:

[outbound_calls]
exten=_9NXXXXXXX,1,Dial(SIP/${EXTEN}@101) (I've also tried with (SIP/101/${EXTEN})

GXW4108 config

Everything is default, except:
-Profile 1 :
SIP server to 192.168.10.24 (Asterisk srvr address)
SIP Registration to NO
-FXO lines :
Wait for dial tone to YES
Stage method to 1

On my phone
When I dial 9 9645 2145, I get:
"Call failed
Reason code: 503"

Thanks a lot for your help! My hair is turning gray..

No ideas at all?

Hi.

I just send the calls straight to the IP address of the gateway

exten=_9NXXXXXXX,1,Dial(SIP/${EXTEN}@192.168.10.27)

Works fine and im sure its what the notes say as well

Ian

Thx for your answer!

I've made the change, but it still gives me the same error.. 503
I don't get it.. Why isn't it working? Is there anything else I need to configure?

Hi

You need to follow the notes in full on from the GS site.

I dont think you have made all the changes yet.

I do have a set of screen shots but am out today so wont be able to post them till later

Ian

Hi,

Ok thanks, if you can send my your screen shots, it'd be great!
The GS tutorial is very short and if you want to use the gateway as a peer. It just says to change a few things, and I've done that..

Thx for you help!

Hi Here are the screen shots as promissed

http://cyber-cottage.co.uk/site/images/ ... an4108.pdf

Ian

Thx a lot for your screenshot!
My configuration is still not working and I've got some questions regarding yours.

Why are you registering SIP numbers (under channels)? I though a peer gateway wasn't needing any of them..

What's that Unconditional Call Forward setting for?

How did you choose your caller ID scheme? And the Vocoders?

Thx a lot for your help!

Ok, I get the unconditional call forward.. What about the rest?

Why are you registering SIP numbers (under channels)? I though a peer gateway wasn't needing any of them..

They are for sending calls from the gateway to asterisk, Im sure it was described this way in the notes I first used.


How did you choose your caller ID scheme? And the Vocoders?

Im in the UK so I selected the callerID that works here and as for codecs these are the ones I wanted to allow.


This config works as I have a number of these gateways on customer sites all setup this way.

Ian

Thanks a lot for all your help, I found my error.

The "Prefix to Specify Port(1 stage dialing method)" was set to 99. In my dial plan I use 9 as a prefix for outbound calls, and Hong-Kong mobile phone numbers start with 9 most of the time. Thus, I was dialing numbers starting with 99 and the gateway was using this "Specify port" function.

Thanks again