Using SIP lines as extra trunks? 2

[plug1]

Afternoon everyone, Im currently setting up a site that only has 2 analogue lines attached to an IP 500 v1 box running 5.022 software. Im looking to add 2 SIP trunks as extra lines so that when the 2 analogues are in use they automatically fall over onto the sip trunks. Ive got the SIP trunks installed and they are registered and I can dial into them but when the 2 analogue lines are in use the third call simply waits for a line, it doesnt go to the SIp trunks.. what am I doing wrong here? Both the SIps and the analogues are on line group 0.

 

[Bas1234]

Change the Line group ID of the SIP trunk to a free number say 10 (so in the ARS and SIP the 0 needs to 10), then create an extra ARS, and at the bottom you can select a ARS overflow (this needs to selected in the Main ARS50)

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
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[plug1]

Sounds easy enough but where do I select ARS overflow? Is that the alternate route part of the ARS?

 

[Darby123]

Thats correct. But I had the same sinario on our in house system. I cannot get thye overflow working correctly at the moment as ther system is only send 6 digits to the SIP provider.

Im trying to work through this today, but I will post as soon as I get more progress

 

[Bas1234]

So how does your ARS looks like now?

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

 

[plug1]

Like this.. its still not working at the moment.

 

[plug1]

Try again..

 

[hairlessupportmonkey]

I trust this is your lab or test setup? if not, get rid of RemoteManager or rename it to something less known, like a convenient dialup name of your choice.

I know you only have SIP and Alog lines, but its just good practice. stops potential hacks.

I also recommend some IPO training. If you are asking these sort of questions and are installing IPOs for customers, then in fact you shouldn't be installing them. And its a bit of a cliché round here, "this aint a place for free support."

Just some friendly advise and not trying to put you off.

ACSS - SME

 

[plug1]

Thanks HSP, Ive completed the IPO training 3 years ago and we are an AVAYA business partner but its the first time Ive had to configure this, Ive set up around 30 sites with IPO's including 4 SIP sites but I havent had to do this up until now, I just thought I could get a bit of a hand here.... dont suppose you want to help me out do you?

 

[hairlessupportmonkey]

well, ARS is what used to be called LCR. Its now a lot more flexible and you can just "cascade" ARS routes for as long as you want. Just create a new ARS entry and then use the Additional Route drop down to route to that when your "Main" route is out of service.

play around with the priority and timers until you get it working the way you want it to.

Press F1 when in the ARS for more info!



ACSS - SME

 

[plug1]

Cheers, from the pics I thought thats what I had done but it doesnt seem to be kicking in? Is there something not right about the short code or is this something to do with having to put a # after the number to dial over SIP?

 

[hairlessupportmonkey]

your pic doesnt show any additional route used....

ACSS - SME

 

[hairlessupportmonkey]

wait, sorry. pic 1 does but you have it set for 30 seconds!

ACSS - SME

 

[Bas1234]

Just add the code you have there in the SIP ARS in the Main ars behind de default code.

So in the Main ARS you get;

0N | 0N | Dial | 0
0N; | 0N"@x.x.x.x" | Dial | 10

If that gives you an error try;

0N; | ."@x.x.x.x" | Dial | 10

Now you can remove the SIP ARS if you like, so now when the analog lines are busy or out of services they will over flow.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

 

[Bas1234]

Also this way you won't ave to wait for 5 up to 30 seconds to geet out from the SIP lines, and you won't get the beep,beep,beep tone.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

 

[hairlessupportmonkey]

nice idea.



ACSS - SME

 

[plug1]

Brilliant Bas1234, only thing is it seems to grab the IP lines first?

 

[Bas1234]

Swap the 2 around.

Delete them and add them in the right order.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

 

[plug1]

LOL, that sounds sensible..

 

[plug1]

Cheers Bas1234, although I couldn't get your dual ARS to work I think because the codes both were too alike, it would never pick up the 2nd route. What Ive done is create a 9N code for the SIP lines and if they need to use the backup lines they dial a 9 first.. not ideal but I just couldnt get the fall back to work... unless someone wants to point me out the obvious fault here..