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#113559 03/17/11 11:01 AM
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teleco Offline OP
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I picked up a new customer recently with 3 hotels and thought I could ease my way into any hardware and programming issues but that changed today. I have a SX200 w/ 6 bothway loop start lines when you dial 9 it will grab 1stcall, trunk 1, 2nd call trunk 2 then 3 but on the 4 call after you dial 9 and the number it takes about 90secs. for the switch to grab trunk 4 and complete the call.So if you don't wait the 90secs for the call to complete the next call trys to access the 4th trunk so nobody can complete a call untill the switch completes a call on trunk 4. Hope that wasen't too long an explanation. Also the LS/GS card # 9109-011-001-SA handle CLID? Thanks for any help you can provide.

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#113560 03/17/11 11:06 AM
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normally the 90 sec or whatever timeout is due to a bad trunk or trunk port. The system will try to seize the line and will wait for the timeout before trying the next line available in the Trunk Group. Only a CLASS trunk card (8 Port) or a SPine Trunk Module will handle CLID unless it's an ICP and the trunks are LS and connected to the the onboard ASU.


Jon


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#113561 03/18/11 12:36 AM
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I agree that you've got a bad trunk! Best way to troubleshoot is to do a direct trunk access on each trunk, from the console. If you can't grab the trunks manually, the ARS will never be able to grab them automatically!


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#113562 03/18/11 02:13 PM
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Or simply busy out trunk 4 and see if the problem goes away.

#113563 04/03/11 11:47 AM
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Hi. I have two hotels with sx-200s. I have read some forums regarding setting up sx200 pbx and asterisk. What I would like to do is use our existing pbx and its features and connect asterisk and use a voip provider and cancel our current provider. DO i need to connect these using a t1 or an fxs / fxo? Will it be as simple as making these connections and configuring asterisk, or will I need to make some configurations in the sx200?

Thank you


ej
#113564 04/03/11 12:41 PM
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teleco Offline OP
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Sorry for the delay responding. What I've found out is 3 lines were dropped at the cutover so I busied them out at the console. I need to remove them from the dial 9 trunk group. Where can I get a good PDF manual? Also would that manual apply to the SX200MX for what I'm trying to do? Thanks

#113565 04/03/11 05:29 PM
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Quote
Originally posted by ejamnadas:
Hi. I have two hotels with sx-200s. I have read some forums regarding setting up sx200 pbx and asterisk. What I would like to do is use our existing pbx and its features and connect asterisk and use a voip provider and cancel our current provider. DO i need to connect these using a t1 or an fxs / fxo? Will it be as simple as making these connections and configuring asterisk, or will I need to make some configurations in the sx200?

Thank you
Is this a homework project? You replace whatever the proerties had with the same equipment. The asterisk box is overkill. If you don't know the difference between FXS and FXO, you probably shouldn't even attempt to do this at all.

Just a thought.

Carl

So as not to be mean, a CO trunk connects to an FXS on the serving facility and you only have to touch the PBX if you need to convert it to loop start if you insist on the stupid asterisk box without ground start capabilities.

#113566 04/04/11 01:56 AM
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DO i need to connect these using a t1 or an fxs / fxo?
------------------------------------------------------------------------------------------------------
Not to step on Carl's toes, but there are several questions that your question raises. First what SX200 are you referring to. Analog, Digital, Light, ML/EL, ICP?
Why do you think inserting an Asterisk will help? (Voip provides CHEAP communications right now, but the GOVERNMENT WILL find a way to TAX it, eventually)
You will need a high level of QOS in order to have decent communications. And at some point, you will have to get back on the legacy phone network to communicate with the end point.
As Carl asked, 'is this a homework project?'. This is NOT a 'do it yourself' project. I've been involved in the turn-up of a company using Asterisk to extend the lifetime of older Mitel systems and provide some of the bells and whistles that weren't around when they were current supported systems, and I'm here to tell you, it hasn't been fun! I'm certified on a wide range of Mitel from SX10 through Lite19 and partners are Asterisk pros. The only reason I agreed to get involved is that I'm retired and needed a challenge. We are now in discussions with a 'COLO' regarding a merger that will greatly enhance our ability to do what we are doing.

To quote an assortment of TV shows, 'Don't try this at home!'.

And please, from now on, start a new topic unless you are adding to a topic on the same subject. No extra charge (still free smile ) and you'll get better results. By the same token, don't drag up an old topic that is on your subject. Rule of thumb, if no one has posted on a topic in 2 weeks, it's probably time to start a fresh one, if you can't find the answer doing a search of the subject here.


When I was young, I was Liberal. As I aged and wised up, I became Conservative. Now that I'm old, I have settled on Curmudgeon.
#113567 04/04/11 10:00 AM
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Originally posted by teleco:
Sorry for the delay responding. What I've found out is 3 lines were dropped at the cutover so I busied them out at the console. I need to remove them from the dial 9 trunk group. Where can I get a good PDF manual? Also would that manual apply to the SX200MX for what I'm trying to do? Thanks
For a manual, try the "manuals" button at the top of this page.

For the trunks, delete them from form 16. Make sure you delete the right trunks. If you aren't sure, it's worth the money to have a qualified tech go onsite.

#113568 04/07/11 07:14 AM
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It is a learning project but it is also for an actual business.

{You replace whatever the proerties had with the same equipment}

I dont know what you mean.

{Analog, Digital, Light, ML/EL, ICP?}

It is not an the ICP; I dont know what it is (we have regular analog lines). I want to switch service from our current phone company to voip. The reason I mentioned the asterisk box was because I reasoned I can use this to connect to a sip provider.


{And please, from now on, start a new topic unless you are adding to a topic on the same subject. No extra charge (still free [Smile] ) and you'll get better results. By the same token, don't drag up an old topic that is on your subject.}

Noted.


ej
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