Trixbox trunk to a digital gateway

Hi all, I've been asked to try and set up and test some basic VoIP systems that might be used on my site, and now it is the turn of trixbox-CE. I've got it installed and running ok on the server, and can make calls SIP-to-SIp no problem. What I cant do is call the outside world as I cant get it to talk to my Samsung or Audiocodes gateways correctly. I cant get offsite unless I go through the Samsung gateway which has 16 channels into my non-IP capable Siemens switch.

All the tutorials I can find on Trixbox concentrate on setting up SIP trunks via built-in cards or to voip suppliers directly, and not how to set up to talk to a gateway device. I had a test setup of Sip-X working previously, and it was comparitively easy to get it working with the Samsung, but trixbox has me somewhat stumped as I dont know which settings to put in. Anyone using trixbox through a gateway that can give me a clue what to enter?

Hi

I would look at Piaf or elastix before trix. both of these also use freepbx but in a more vanilla form.

As to conecting to a gateway it depends on the gateway most just register to a peer on the * for incoming and * just sends the invite to the gateway direct but gateways differ

Ian

I'm ok sticking with Trix for now, the only part I need to get working is the trunk link and I'll be able to complete my task.

Are you saying I just give the gateway the registration details of a user (real or made for the purpose) and have it log in like any other SIP device? Worth a try i suppose, I haven't tried that yet.

The Samsung appears to be very similar in design to Cisco kit, the IOS is almost identical. We only bought it as Samsung give us lots of free support as they're trying to move into the Enterprise voice market now. Thankfully voice quality was way better than the Audiocodes kit, but their implementation of the SIP protocol appears to be slightly different from everyone elses. It makes it incompatible with Sip-X-ecs, but they say it should work just fine with Asterisk based systems.

** UPDATE **

I tried registering the gateway with Trixbox with no success. However, I left the gateway with a basic config and played around with the trunk settings in Trixbox with partial success. I can dial out but not in, although the called party cannot hear me even though I can hear them. Not a stunning success by any measure, but still heartening as it means I am on the right track. I suspect I just need to put the right commands in the trunk and inbound route configs to get this working.

The link you provided was excellent, lots of info to go through there. However it would help if I new exactly which coommands I need to enter and which I dont need. Never having seen this configured means I'm groping around in the dark, trying commands and then testing to see if I have 2-way comms.

Currently, a debug run on the gateway during inbound calls shows SIP invites being sent to the trixbox server from the gateway, but not being acknowledged in any way. Previous fumblings had it returning a 404 error, which so far has been the best I've had on inbound calls. Here's what I currently have in my trunk config boxes...

OUTGOING
host=129.230.68.XXX
nat=no
type=peer
canreinvite=yes
DTMFMode=rfc2833

INCOMING
type=peer
nat=no
canreinvite=yes
DTMFMode=rfc2833


That's it so far. Can anyone clue me in on what I'm missing? VoIP is new to me, and I need all the help I can get here. Thanks.