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SIP/H323 GW or H323 operator

Posted on 2005-04-18
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Last Modified: 2012-05-05
I have an internal VoIP network running H.323 with OpenH323 GK, netmeeting clients/ ATA 186 client and some propriatary H.323 equipment.

I would like to be able to call to and from PSTN/Mobile phones from my H.323 network but my problem is that I can only find operators selling SIP access in the country I am in (Denmark) - noone seems to use H323 anymore :-(

1) Does anyone know an H.323 operator in northen europe?
2) Is there any opensource projects from which I can get a SIP/H323 Gateway? I dont want to spend too much money and time on bying a hardware box as this is only a test project, so I need a solution that relies either on OpenSource or evaluation software.
3) Does anyone see another solution for me?


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Question by:TeeRoles
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4 Comments
 
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Expert Comment

by:grblades
ID: 13814002
Asterisk supports H323 so you could configure it to talk to the SIP voice provider and your H323 system to pass voice calls onto Asterisk.
Asterisk will then convert between SIP and H323.
See http://www.voip-info.org/wiki-H.323
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Author Comment

by:TeeRoles
ID: 13814126
Thanks grblades.
But when you look at Asterix it just seems like quite a complex system with a lot of features.
I am therefore afraid that installing Asterix just for the purpose described above might be a bit of work.
Have you any idea how difficult Asterix is to configure for this if I have zero experience in Asterix?
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Accepted Solution

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grblades earned 2000 total points
ID: 13814439
It is fairly easy to configure just for the basic stuff you want. I am configuring it myself for the company I work for.

The h323 config file would look something like that shown which would permit h323 connections from the IP specified

[general]
port = 1720
bindaddr = 0.0.0.0
[incoming]
type=user
host=192.168.1.1
context=voip

Then in sip.conf you have your SIP provider configuration which is something like :-

[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
register=username:passwordhere@your.isp.hostname
[voipprovider]
secret=passwordhere
username= username
fromuser= username
type=peer
context=incoming
host=your.isp.hostname
canreinvite=yes
nat=0

Then you have entried in extensions.conf to route calls through to the VOIP provider:-

[general]
static=yes       ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[voip]
exten => _9.,1,Dial(SIP/${EXTEN:1}@voipprovider,60,Tr)


This should be all you need for a basic configuration. You would configure your H323 system to route all calls starting with the telephone number '9'  to Asterisk. Asterisk will then remove the leading '9' and pass the rest of the number onto the Voip provider.
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Author Comment

by:TeeRoles
ID: 13814755
Thanks a lot grblades.

I will try your suggestion and install the Asterix as a H323/SIP GW.
If I cant get it to work i'll be back with a new question :-)
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