mallgood

Wow, its been a long time since I've posted! (almost 2 years!)

I was just curious if anybody had tried setting up an IPO with a VCM to talk to an Asterisk based PBX for SIP, etc.
I heard that Avaya was going to be pushing SIP functionality to the IPO in a short while, but this is more of just a learning process.

What I would ultimately like to be able to do is setup an Asterisk server that was outward (internet) facing that was multi-homed (dual nics) and be able to have clients with SIP phones connect and make calls out over our IP Office system's trunk lines.

Anyone have any experience with something along these lines?

My main problem is getting the H323 setup correctly.

TIA!

Phone dude

I am not familiar with asterisk. Can you set up voip trunks in the Asterisk?
Below is a cut and paste from another sites FAQ. It is designed for 2 ipos on a network, but you will get the jist of it...



I hope this FAQ will help with enquiries about setting two sites for VoIP. The link could be xDSL, lease lines or even Frame Relay controlled by third party routers. It does not matter, as long as you could ping from one network to another, and that the third party routers support the protocols that you are using. Other configuration with the WAN ports are possible.

I do not intend to rewrite the manual, but if my information is wrong, please do correct me. Please note I have several fully functional sites with this similar setup.

Required hardware:
IP Office or ArgentBranch at both sites
VCM at both sites.
Static IP Address for ADSL modems.

You will have to use your imaginations:


Site A Site B

ADSL Modem ADSL Modem
192.168.42.254 /24 172.16.0.254 /24

___ ___
| | xDSL | |
--| |---/-------------------/---| |---
| I__I I__I |
| |
| |
| |
| ____ ___ |
| | | | | |
| | I | | I | |
I_| P | | P |_|
| O | | O |
I___I I___I

IP Office IP Office
192.168.42.253 /24 172.16.0.253 /24

Extn range = 200 Extn range = 300

Line;
Line Group ID = 20 15
No of channesl = 5 5
IP Add = 172.16.0.253 192.168.42.253
Compression = G.729 G.729

Short Codes;
Short Code = 3N 2N
Telephone No. = . .
Line Group ID = 20 15
Feature = Dial Dial

IP Route;
IP Address = 172.16.0.0 192.168.42.0
IP Mask = 255.255.255.0 255.255.255.0
Gateway = 192.168.42.254 172.16.0.254
Destination = LAN1 LAN1

PBXtech.info

 

dhaggar

A colleague of mine almost had this working on our IP Office about 18 months ago. You will have to use h323 until sip is available on the IP Office and he found h323 on Asterisk very flaky. I know he eventually gave up trying because he couldn't get the call supervision to work. At one stage he could establish a call but it was nailed up until the Asterisk server was rebooted!!. He also had trouble with calls in one direction but I don't remember if it was Asterisk to IP Office or IP Office to Asterisk...
Good luck in you quest.

Phone dude

SIP is compatible with the new 4.0 released monday.

mallgood

Yeah. The older versions of the H323 stuff is *very* flaky. They have made a lot of improvements and it seems to work now.

I have got H323 & SIP working on the asterisk box. I can connect a SIP phone and a MS Netmeeting client (H323) to it and make calls back and forth.
I currently don't have a VCM, so I can't try the H323 trunking to the IPO, but I think I have the setup right.

I would just skip the whole exercise and use SIP, but I have an IPO403! We were an early adopter and now we are paying for it.

Speaking of which, can you run 3.2 software on a 403? or just 3.1? anyone know?

randyc

It depends on the phones. If you have 20XX phones you can only use 3.0DT. If you take the DT out of it you can run 3.2 fine.

PBXtech.info

 

dhaggar

I'm running 3.2(54) OK but its a 403 DS. I'm thinking about going to 4.0 (to get SIP support). Anyone any thoughts / experience about 4.0 on a 403 DS?

Phone dude

4.0 is not supported on IP403. That said, it may or may not work. Sure it may work, but no support for it.

tlpeter

it won't work because you won't get it on the 403
the 403 does not have enough memory to load the 4.0 software