SIP Trunking Routing

[Bibbyboy]

I have a customer who has SIP trunking at one location, and they have another IP Office at another location that just has 4 analog lines. Both IP Offices' are 406 on 4.2 software. Because the phone company at the phone system with the analog lines would not release the numbers to the carrier they are doing SIP trunking with, they could not port those numbers over to the SIP trunk, therefore they had to come in as analog lines. However, I want outgoing calls from that phones system to go through the other phone system that has the SIP trunking, I figured the way I would do this is to make the line group ID on the ARS table the same as the one on the other phone system (even though it wasn't on the dropdown) but all I got was a waiting-for-line, what can I do to make this work?

 

[Westi]

@Maxwell
bas1234 has exactly what I use, he actually gave me that hint. You need then 2 URI entries one for incoming and one for outgoing and use bogus line ID for the incoming on the fake caller ID and a bogus line ID outgoing for the real number that gets the incoming calls.
Practical jokes are great
You can also send the name you would like to show up, don't forget to put that in as well otherwise you might give yourself away when it says Kitchen, like I did the first time, LOL.

Joe W.

FHandw., ACS

If you can't be good, be good at it!

 

[Bibbyboy]

I had thought of that, the only problem is that the main site with the SIP trunking wants their DID's to show up, so USER data is in those fields...if I change that it will mess up the main site, how can I make it where only calls from the other phone system go out on another SIP URI?

 

[Bas1234]

@Bibbyboy,

Glad you've got it working... (* )

Could you explain a bit more, not sure what you mean.

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It works! Now if only I could remember what I did...
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[Bibbyboy]

alright, 201 is the ID of the URI that phone calls are now using like you had me do. What happens is I need the URI to have user data in the fields so that every users DID shows up on outgoing calls, but on the phone system that doesn't have sip trunking, I want just the main number to show, which I could input the main number to that location in the URI, except that now the calls from the SIP trunking phone system will show that number as well, I want phone calls from the the non-sip phone system to have its own URI, except that both sites are now sharing the same ARS table for outgoing calls, so what I change on one I change on both if that makes any sense.

 

[Bas1234]

How many extension do you have, i think you'll need to create a new SIP Uri for every user with the own Line group ID.
Only because you don't have the SIP licences on the site where users are. You now have created a sort of breakout.

Greetzzz...Bas

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It works! Now if only I could remember what I did...
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[Bibbyboy]

Hahahaha Bas1234, Its funny you say that, thats actually how we had it setup originally, and once we went over around 50 URI's, the phone system's 16 port digital module would not come up after a reboot, we worked with avaya and found that the programming was doing it, then we went to our current setup.

 

[Bas1234]

It must work.

Copied this from the docs.

Note that the IP Office only supports up to 150 URI entries on a SIP line.

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It works! Now if only I could remember what I did...
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[Bibbyboy]

Yes, that is what we read as well, as this customer was our first SIP install and we wanted to be completely prepared, for some reason it doesn't work with our config, and we removed shortcodes, users, exts, one at a time from the config to troubleshoot, but we never did figure out why the uri's were doing that, avaya was no help with it either, the customer is fine with the 3rd line going out as the caller ID, so Im content with our solution. Thanks again everyone.

 

[Bas1234]

What version are you running on?

Could be a bug in the 4.0 or 4.1, no problem in 4.2 as far as i'm aware of.

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It works! Now if only I could remember what I did...
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[Bibbyboy]

were on 4.2.4

 

[Bas1234]

found in a TB 4.1 had a problem;

CQ61993 Unable to add more than 101 entries within SIP URI form

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It works! Now if only I could remember what I did...
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[Maxwell1001]

@bas1234 tried what you said and makes me unable to dial out

thanks anyway will have a mess around, im in the UK this may be the issue you mentioned that your supplier wont let you send another number

ACA - IP Office
ACS - IP Office

 

[Pepp77]

To those in the UK - Voiceflex will present any number as long as you can prove its yours. For example we send out our main incoming ISDN number on our SIP calls and just needed to show them a BT bill showing our address and the phone numberwe wanted to send out.