Passing DID and CID when linking two Trixbox servers

matttux
Posts: 4
Member Since:
2009-05-06

I have a Trixbox server with a PRI connected and i am forwarding all calls from this PRI to another Trixbox server via a SIP Trunk.

The problem i am having is that i am not reciving any Caller ID or the DID dialled. See below:
---------------------------------------------------------------------------------------------------
-- Executing [s@from-sip-external:1] GotoIf("SIP/xxxxxxxxx-094a0f50", "1?from-trunk||1") in new stack
-- Goto (from-trunk,s,1)
-- Executing [s@from-trunk:1] NoOp("SIP/xxxxxxxx-094a0f50", "No DID or CID Match") in new stack
-- Executing [s@from-trunk:2] Answer("SIP/xxxxxxxxx-094a0f50", "") in new stack
-- Executing [s@from-trunk:3] Wait("SIP/xxxxxxxxx-094a0f50", "2") in new stack
== Spawn extension (from-trunk, s, 3) exited non-zero on 'SIP/xxxxxxxx-094a0f50'
-- Executing [h@from-trunk:1] Hangup("SIP/xxxxxxxx-094a0f50", "") in new stack
== Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/xxxxxxxxx-094a0f50'
---------------------------------------------------------------------------------------------------

I am getting all the information on the box with the actual PRI connected but how do i pass these along the SIP connection to the other box?

The way i have the current setup is as follows:
- On the box with the PRI i setup a SIP extension (1000) and all incoming calls are forwarded to this extension
- On the 2nd box i setup a SIP trunk with the following details:

Peer details:
allow=ulaw&gsm
canreinvite=no
host=xxxxxxxxx&dynamic
insecure=very
qualify=500
secret=xxxxxxxxx
type=peer
username=xxxxxxx

USER Details:
allow=ulaw&gsm
canreinvite=no
dtmfmode=rfc2833
insecure=very
qualify=500
secret=xxxxxxxx
type=user
username=xxxxxxx

Registration String:
'my_username':'my_password'@'myserver'

Many thanks for any help.



IcelandDreams
Posts: 415
Member Since:
2007-09-11
I'll give an uninformed

I'll give an uninformed guess, why pass the call to a local extension 1st? That's where you loose CID. Perhaps pass it directly to the other server based on inbound routes or CID like you would for local extensions? My second box is only for tests but I get CID to the second box via a simple IAX trunk. Is your extension numbering unique and logical between boxes? Someone else might have a better solution based on your setup.



matttux
Posts: 4
Member Since:
2009-05-06
Thanks for the reply. I set

Thanks for the reply. I set up a SIP extension on the system with the PRI (extension 1000) and i register the second box using these extension details. So Box2 is registering using user: 1000, pasword, IP of PRI box.

This is the only way i knew of linking the 2. If i set up an IAX trunk i have no option in the inbound routes to pass this down the trunk. How do you do this, is it a custom app?



IcelandDreams
Posts: 415
Member Since:
2007-09-11
There seems to be several

There seems to be several ways to link boxes. I do a very simple IAX trunk without extensions other than actual phones. Mine looks similar to this:

Outgoing Settings-----
trunk name : (name)
secret=(password, make it long and difficult)
type=friend
qualify=yes
host=(ip/hostname of other side)
context=from-internal
username=(authentication name)
nat=(yes or no)

no Incoming Settings or Registration.
The 1st box has inbound routes normally and if the destination is an extension on the remote box it gets there over an outbound route. Simple and effective.



matttux
Posts: 4
Member Since:
2009-05-06
Ok i see what you mean. My

Ok i see what you mean. My probem is that we need a PRI but because of a few problems we needed to put this in a remote location. I have a number range for testing but will be porting a few hundred DIDs onto the PRI.

The second box has all of these Inbound Routes set up so i was hoping to do one forward for all calls which will contain all cid and did details once forwarded. At no point will i have any local extensions on the box with the PRI, all calls in will just be sent to the second box.

I have tried playing around with a custom destination to forward calls this way. Is that likely to work?

And thanks again for the help.



matttux
Posts: 4
Member Since:
2009-05-06
Got it sorted pretty easy in

Got it sorted pretty easy in the end. Just put exten => _X.,1,Dial(SIP/trunk/${EXTEN}) in the extensions.conf



Comment viewing options

Select your preferred way to display the comments and click "Save settings" to activate your changes.