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Old PC to run Open H323 Gatekeeper. How to configure it and where to get it

Dr-IP provided me with this solution to my problem to terminate voip calls to a voip partner; Can he or anyone else tell me what would be the dial-peer config on both cases and how to implemente the last option of Openh323 ?
Here is Dr-IP comments on one of my previous questions:
" You can also use a crossover cable between the E1 ports to do effectively the same thing, but that ties up two E1 ports on the gateway which isn’t an optimal solution most of the time. Another solution would be to get an old PC and run OpenH323 Gatekeeper on it which is free, and would give you more flexibility, especially if you continue to expand your network with additional gateways "

Please help
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jeanpoman
Asked:
jeanpoman
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1 Solution
 
Joel_SiskoCommented:
Here is a link to get you started:

http://www.gnugk.org/gnugk-cisco-as5300.html

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Joel_SiskoCommented:
Few things needed if you need more specifics than the link above may provide:

1. Your current configuration of gateway(s) both Cisco and non cisco
2. Will you be using this on a Windows system or Linux? I would go towards the Linux for a few two main reasons: stability and more available tools to work with.

Kindest regards

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Joel_SiskoCommented:
Jeanpo,

How are things coming along? Do you neda ny additional infomration?

Kindest regards
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jeanpomanAuthor Commented:
if you use a crossover cable betwween E1 on you gateway .the call id for the call will be same as a result the gateway will know that the call is looped  and the GW will drop the call.

I know  OpenH323 Gatekeeper it works but not as good as a proper vendor gatekeeper although you could use it do not expect high quality call termination
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Joel_SiskoCommented:
I will double check to see if there is a way to turn off the logical loop of the Cisco Gateway, never knew that exsisted, learned something new today, its a good day indeed.

Would it be possible to add a rule to the outbound of the E1 loop to insert digits, so this way when recieved by the inbound loop it does not appear to be the same? Then you could remove the digits when recieved?

In regards to OH323 , how many calls per hour are you trying to terminate? Also how many calls per hour do you process thru your entire system?
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jeanpomanAuthor Commented:
I am trying to terminate 600 calls a day. This matter is turning my head and I am about to adandon the project.
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Joel_SiskoCommented:
Were you able to configure the OH323? I think by your previos response that you were having problems with the quality of it? 600 calls is really not alot and the system should be able to handle it. What has actually been happening with the calls? Have you been able to route the calls from call shop to voip partner? If this was solved, then it might just require a bit of tweaking to get you to handle your call volume.

Kindest regards
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Dr-IPCommented:
If you are gong to loop through a crossover cable you would have the sending gateway add a prefix to its calls, IE 999, and create a dial peer on the Cisco with the destination pattern off 999T. Which would cause the gateway to strip off the 999 so that when the call was looped back it would be in the original format, at that point the gateway would thing it’s a different call and send it out.  

Gateway 1 subscriber dials 541234567

Gateway 1 sends number and prefix 999, IE 999541234567

Reweaving gateway 2 gets call 999541234567 and sends it to the pots dial peer with the destination 999T.

Call is returned to gateway 2 through loop back as 541234567, matches VOIP dial peer, and is sent to vender.

As for configuring the open gatekeeper app, that depends on the OS, and your network. Cisco gateways are configured the same as if you were using a Cisco gatekeeper, but the gatekeeper it’s self is completely different, but its documentation isn’t that bad. I haven’t used it a lot myself, but I have tested it in a limited production environment where it saw over 50K calls a day. It handled the volume pretty well, but I didn’t go much further than that since it was clear that even though it had good routing capabilities that it was no where close to being able to handle the 100K+ routes I have to deal with. So even though it gained my respect, I realized it couldn’t do what I needed, so I moved on to getting a soft switch, which wasn’t cheep, but did what I needed it to do.    



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Joel_SiskoCommented:
Jeanpoman,

DrIP provided the detailed version of my post above, I think with the current call volume, using the E1 loop with add/delete would be a good way to go. Once you have this implemented I would seriously look at moving to SIP. The open Sip Express Router can handle Thousands of calls per  hour and SIP enables a much more flexable enviroment.

Kindest regards
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Dr-IPCommented:
One thing I should note, most switches by default if you send a call to them on the same E1 that it normally terminates it on, will send it back out on that same E1. Normally that is considered an undesirable trait because it causes loops, but in your case it could work for you. So maybe you should try sending the call to the switch using a prefix that is striped by the pots dial peer and see of the switch sends it back out.    
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Dr-IPCommented:
" I would seriously look at moving to SIP. The open Sip Express Router can handle Thousands of calls per  hour and SIP enables a much more flexable enviroment."

I have played both sides of the fence, and more when it comes to VOIP. I use H233, SIP, and MGCP protocols for VOIP networking, so you can say I am pretty open minded when it comes to VOIP protocols, but I a far from believing SIP is the cure all many have portrayed it to be. I have been listening for 5 years how SIP was going to revolutionaries VOIP, and was an early adopter, but haven’t seen it come even close to its promise.

After over 3 years H323 still accounts for more than 80% of my VOIP traffic, and even MGCP accounts for more traffic than SIP. The way I see it, the noise about SIP is unwarranted since the only place SIP seems to be going anywhere is for little SIP phones, while the big boys continue to run their VOIP networks on H323.
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Joel_SiskoCommented:
Dr-IP,

You have beed drinking Cisco Kool-aid to long (LOL). Actually glad to have some one in the VoIP area whose knowledge goes beyond the wiki at http://www.voip-info.org (pop shot at number VoIPs :) )

I will have to respectivley diasagree on the SIP (not saying it is the end all, but has the best change for winning the VoIP war). I have to head to a clients , but will add more later to this post and my reasoning.

I would like to see if you could answer why H.323 took hold in the past 3 years? (I have been listening to VoIP/CTI for 15 years) and have the answer why it did (at least my humble opionion).

Look forward to future posts with you Dr-IP

Kindest regards

Joel_Sisko

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