H323 Trunk, how to configure


i got a trial account of a small voip provider which is only the information that it has to be configured as a H323 trunk, and ip address and usernam and password. i know it can be done as a custom trunk.

H323/[email protected]

the x representing the ip address. but i cant see where to put the username and password.

any advice ?



Yes, I too have a provider that only runs H323. I am currently running FreePBX 2.4.0 and need to know what can be done so I can receive H323 calls on my system.

Hope there is somebody out there with a solution.



I did google, today, last friday and last thursday.
But I´m new to Elastix, and each answer I find brings more questions.

I need to connect an Avaya Communication Manager to an Elastix 1.3-2 using an H323 trunk.
Can somebody tell me how to do this step by step?

as_bat, I take it you didn’t try and google today?

asterisk connect to an Avaya using H323 trunk

will get you 4 good posts in the first 10 displayed.

You just can’t edit extensions.conf, sip.conf Put the stuff in the *_custom.conf files instead.

h323 is not a generalized interface that is exposed as such so you’ll have to hand edit several files.

Double post, sorry

I did everything you told me in your step by step guide (BTW thank you!!!) but I still can´t call from one elastix extension to an avaya extension or viceversa.
I suspect that the channeltypes has something to do with it.

Type Description Devicestate Indications Transfer

MGCP Media Gateway Control Protocol (MGCP) yes yes no
IAX2 Inter Asterisk eXchange Driver (Ver 2) yes yes yes
OOH323 Objective Systems H323 Channel Driver no yes no
Zap DAHDI Telephony Driver w/PRI no yes no
Agent Call Agent Proxy Channel yes yes no
SIP Session Initiation Protocol (SIP) yes yes yes
Phone Standard Linux Telephony API Driver no yes no
Local Local Proxy Channel Driver yes yes no

8 channel drivers registered.

Should I have a h323 device or the ooh323 is enough?
Should it be up (yes) the same way the SIP or IAX2 are now?

I’ve got a bunch of Avaya 5610 phones that I’d like to use with a working Asterisk box. I compiled with h323, got all my sip trunks, routes and phones working and then I hit a brick will with h323.

I created a custom trunk H323/[email protected]:1720 and a basic h323.conf but the phone can’t “Discover” my Asterisk box.

I have used these phones on an Avaya IPOffice so I know they work and how to point them to the PBX but they don’t see it or vice versa.

This seems pretty basic compared to some of the stuff you guys are doing. Can anyone give me a step by step or at least tell me how to verify that h323 is working on asterisk??



Do you have iptables running (or another firewall inbetween) and if so did you open up all the ports needed for the h323 protocol?

  • iptables is not running
  • Firewall is off…Selinux is disabled.
  • The h323 phone is on the same subnet.
  • The phone can access tftp to get it’s updates and config
  • I can contact the pbx with Netmeeting so I think h323 is alive
  • The phone (Avaya 5610) stops at discover (asterisk box IP)
  • When I look at “channels” using freePBX gui I do not see an h323 channel
  • I have a “custom” trunk definition in FreePBX gui, it has this in the “custom dial string” H323/[email protected]:1720
  • I have a h323.conf based on someone else’s working config

I am not trying to connect asterisk to an Avaya PBX. I am only trying to use some donated Avaya h323 phones to help out a non-profit.

Any other thoughts?



What versions of FreePBX and asterisk are you using? I’m no h323 expert but in quickly searching there seems to be severla different versions that have come out over time so knowing the versions here will be important.

Asterisk is
FreePBX is 2.5.1
PWLib is 1.8??
H323 NOT OH323 is the 2nd to last stable release not sure on the number

I went digging for a while last night on finding support for those phones, h323 and asterisk. While it looks like it should just work based on just configuring the protocol. I could not find any real solid info to point you to on how to go about it or what values are required versus optional.

I’d recommend trying the #asterisk IRC channel and/or the Asterisk forums for support. I don’t know the h323 protocol and what it all needs to know what details are really needed for setting it up properly.

If you go to these sources try and not mention FreePBX as they frown on automated systems where the admin does not really know all the details of each line of the dial plan. It should not be a problem currently as you really need help in configuring protocol and the phone not the dial plan.

If you figure it out I’d ask that you post it back here and any notes to help the next person out.

Good luck.

I am a newbie on Asterisk, thus the FreePBX interface. The other reason was a dead-line that has long since passed. I had to go with a used digital phone system for my non-profit, foreclosure assistance friends.

I am not a Linux newb and got my 1st taste around 1994 when I downloaded about 96 Slackware floppy images so I thought I would give it (asterisk) a shot.

Either way I am going to keep the Asterisk system I created here at the house and drop my home phone analog service. It’s a kick to have extensions, voicemail and all the features of a good VOIP system at home.

I will take your advice and pose as a hardcore, asterisk hound on some of these other channels.

AND,…I will definitely post a “step by step” solution here if I figure it out or get some answers,

Thanks Again,


I had everything working but h323 before I started fighting that battle and I broke something. Now I can’t answer incoming calls or get them to go to voice mail. If I call the asterisk box from my cell the phones ring, but when I pick them up it keeps ringing.

My gut says this is something simple. I’m ready to give up on the GUI and learn the .conf text language.

My SIP trunk provider is “Callcentric” if that helps.



3 Soft Phones from X-lite

1 hard phone from Grandstream

Outbound calling works beautifully

internal ext to ext works perfectly

Inbound calling is a mess.

I have 4 phones
2 ring groups
hopefully an IVR
I want a “parking” lot and queues someday

Do you know where I might find a good Sample config to learn and build from??

sounds like you have a messed up inbound route someplace. Try deleting all the inbounds to start and re-create jsut what is needed.

Also post a log as it’s hard to guess without seeing what is actually happening.

believe it or not my SIP trunk provider, Callcentric gave me the fix.

sip_general_custom.conf add


They really rock. Most of my experience with vendors in 19 years of Info Tech has been about proving I didn’t do something wrong before they would help me. These guys assumed nothing and looked at almost every one of my .conf files + debug data from the CLI.

I was tempted to ask them about my h323 issues.

Thank You! fskrotzki for your input too. I will figure out this h323 thing someday and send you the details.


I’m new using Elastix platform. Somebody can guide me how to configure h323 trunks in Elastix. I need to link an IPO406 via H323 against Elastix. If you post a link to investigate it, I’ll apreciate it. thanks.