Avaya IP Office 500 v2 and Asterix 2

[xotaPT]

Hello all,

First of all i would like to thank all the user of Tek Tips, found lots of usefull itens here, that make my life easyer.

Second i've started working with avaya and asterix recently and i'm having some issues i dont seem to be able to solve.

My current setup

Avaya as Principal
Asterix as Secondary.

Avaya as all the BRI connections as well as GSM and everything is handle trought Avaya.

Asterix is used to increase the lines with as a low cost solution.

So far i managed to configured calls between both system, wich work very well.

I can call, transfer calls between both systems without any issue, i have a sip trunk in avaya that connects to Asterix, and i've configured a Call Router that accepts "*" in that sip line, wich makes that asterix can call any avaya extension without a problem.

So far so good.

The problem is when i try to use any of the GSM or BRI lines, i get a 404 Not Found when calling from asterix.

I've been searching all around, Tek Tips, Google, Avaya, Asterix etc, but i have not found any solution for this problem.

Can you guys help me out?

Please remember i'm kinda noob on this thing of Avaya and Asterix.

thx

 

[amriddle01]

The problem is the system has no match for what you are dialling hence the 404, you need to add an incoming call route to match what they are dialling. For example if you used line group 10 in your * * * URI then add an incoming call route for line group 10 with incoming number left blank and make the destination . (just a dot), this will cause whatever the users dial in the asterisk to be matched against the IP Office system shortcodes and so routed accordingly

 

[xotaPT]

hello amriddle01, thank you for your reply.

But that was already configured, and that makes the "internal" calls work perfectly between asterisk and Avaya! but the "external" calls simply doesnt work!

That is what makes all of this strange, if i call an avaya extension everything works like a charm but when i try to call a number that is not an extension, wich avaya should take it trought the ARS the call doesnt even get to Avaya, and returns the following error in Asterix!

-- Executing [999999999@local_processing:8] Dial("SIP/phone04-000001be", "SIP/999999999@trunk_AVAYA|120|M(CDRData^SIP/phone04-000001be^1338998892.477)") in new stack

    -- Called 999999999@trunk_AVAYA

    -- SIP/trunk_AVAYA-000001bf is circuit-busy

  == Everyone is busy/congested at this time (1:0/1/0)

    -- Executing [999999999@local_processing:9] Hangup("SIP/phone04-000001be", "") in new stack

Take the "999999999" as a number, that was not the actualy called number!

 

[tlpeter]

You better make a trace on the ipoffice.


BAZINGA!

I'm not insane, my mother had me tested!

 

[xotaPT]

tlpeter, sry for asking a perhaps "noob" question, but if i dont even get the call on the ipoffice, how should i trace something that doesnt get here?

 

[xotaPT]

it was indeed a noob question!!!!

06-06-2012 17:53:46-349ms  Line = 18, Channel = 1, SIP Message = Response, Direction = From Switch, From = 404@firewall.jtm.pt, To = 999999999@10.10.0.250, Response = 100 Trying

06-06-2012 17:53:46-351ms  Line = 18, Channel = 1, SIP Message = Response, Direction = From Switch, From = 404@firewall.jtm.pt, To = 999999999@10.10.0.250, Response = 404 Not Found

06-06-2012 17:53:46-354ms  Line = 18, Channel = 1, SIP Message = Ack, Direction = To Switch, From = 404@firewall.jtm.pt, To = 999999999@10.10.0.250

there u go with the trace, and now i understand the error, but i dont understand how can i solve it.

as far as i understand, it its trying to redirect a number to a SIP trunk avaliable at the ipoffice(10.10.0.250) being the asterisk(firewall.jtm.pt), however it should de redirecting to the ARS, but that isnt happening

 

[tlpeter]

It should send IP asterix and not firewall.something.
I guess that is not was is configured on the ipo.
What should happen is that you dial a prefix and then it should go to the ipoffice.


BAZINGA!

I'm not insane, my mother had me tested!

 

[xotaPT]

Hello,
Whats actualy configured in the IPO is "firewall.jtm.pt" its an internal domain, i have tryed both with an IP or with a Domain, and no use, still doesnt work. altought with any of the solutions the Externsions works without an issue, but still cant call outside! states the same not found!


its the log for an internal and external call


07-06-2012 13:49:37-967ms Line = 18, Channel = 1, SIP Message = Response, Direction = From Switch, From = 403@firewall.jtm.pt, To = 10@10.10.0.250, Response = 100 Trying
07-06-2012 13:49:37-968ms Line = 18, Channel = 1, SIP Message = Invite, Direction = To Switch, From = 403@firewall.jtm.pt, To = 10@10.10.0.250
07-06-2012 13:49:37-977ms Call Ref = 3, Alerting, Extension = 10, Button = 1
07-06-2012 13:49:37-981ms Call Ref = 3, Originator State = Incoming Alerting, Type = Trunk, Destination State = Alerting, Type = Target List
07-06-2012 13:49:37-982ms Line = 18, Channel = 1, SIP Message = Response, Call Ref = 3, Direction = From Switch, From = 403@firewall.jtm.pt, To = 10@10.10.0.250, Response = 180 Ringing
07-06-2012 13:49:38-791ms Line = 18, Channel = 1, SIP Message = Cancel, Call Ref = 3, Direction = To Switch, From = 403@firewall.jtm.pt, To = 10@10.10.0.250
07-06-2012 13:49:38-793ms Line = 18, Channel = 1, SIP Message = Response, Call Ref = 3, Direction = From Switch, From = 403@firewall.jtm.pt, To = 10@10.10.0.250, Response = 487 Request Terminated
07-06-2012 13:49:38-794ms Call Ref = 3, Originator State = Clearing, Type = Trunk, Destination State = Alerting, Type = Target List
07-06-2012 13:49:38-794ms Call Ref = 3, Disconnect from Originator End
07-06-2012 13:49:48-011ms Line = 18, Channel = 1, SIP Message = Response, Direction = From Switch, From = 403@firewall.jtm.pt, To = 999999999@10.10.0.250, Response = 100 Trying
07-06-2012 13:49:48-013ms Line = 18, Channel = 1, SIP Message = Response, Direction = From Switch, From = 403@firewall.jtm.pt, To = 999999999@10.10.0.250, Response = 404 Not Found
07-06-2012 13:49:48-016ms Line = 18, Channel = 1, SIP Message = Ack, Direction = To Switch, From = 403@firewall.jtm.pt, To = 999999999@10.10.0.250

 

[tlpeter]

You are getting a 403.
It looks like it is registered as an extension and not as a trunk.


BAZINGA!

I'm not insane, my mother had me tested!

 

[xotaPT]

403 its the extension of asterisk not the error

 

[tlpeter]

funny
I guess i need some sleep.
OK, i see a 404 not found.
Try adding the actual ISDN numbers in to the SIP URI.
Perhaps this works.
I have no idea why it does not use the trunks on the IPO.
I should see it with my own eyes i guess but that is not an option.


BAZINGA!

I'm not insane, my mother had me tested!

 

[xotaPT]

about you seeing it with your own eyes, if you care to elaborate, perhaps it will become possible


About the sip URI, adding them in the SIP uri, didnt work, the error was exactly the same.

As far as i understood avaya system, it works like this.

[ol 1]Number come in from asterisk.[/ol]
[ol 2]IPO try's to check for an extension with that number.[/ol]
[ol 3]If exists the call is passed to that extension.[/ol]
[ol 4]If doesnt exist it should pass to the ARS to find out what to do.[/ol]

The error is in the point 4, if the number is not an extension, it simple doesnt work at all, the call that was suposed to go trought the ARS to leave the IPO trought one of the trunks avaliable, ISDN, GSM, or Voip.

 

[amriddle01]

Are you sure this part is already configured (this is what makes it route out the Avaya via its trunks)

then add an incoming call route for line group 10 with incoming number left blank and make the destination . (just a dot)

Also make sure any prefix the Avaya is expecting is present in their dialled number from the asterisk

 

[xotaPT]

It's already like that, altought i have just found out a solution and it works like a charm

i will leave it here for the next guys that try to make the same configuration

First of all you need to configure the IPO like you said

Incoming Call Route with the number of the SIP line, Dialed Number as an "*" and Destination its a "."(dot).

Now in the sip URI you have to place the following

Local URI -> "*"
Contact -> "*"
Exibicion number -> "*"

With this details, everything is now working like a charm and without any issue so far, the call if matched to an extension places an internal call, if not goes trought the ARS and leaves de IPO following the General Rules!


Anyway thank you very mutch for your help, and there goes a star for you thx

 

[amriddle01]

In my first post I eluded to the * * * URI, but I had assumed that you had that in place already from this...

and i've configured a Call Router that accepts "*" in that sip line

I guess you meant the asterisk not the IPO, at least it works now anyway

 

[clarkstyx18]

Hi xotaPT , Ive been looking for the same solution for a long time.. When I started interconnecting IPO and Asterisk, I was able to make the extension to extension calling through each pabx vice versa, but when I try to use the analog trunk from Asterisk extension, It wont let me go through.

Here's my IPO incomming call route configuration
Incoming Call Route with the number of the SIP line, Dialed Number as an "*" and Destination its a "."(dot).( Are you referring Dialed number as Incomming number? I only added * on "incomming number field" and the rest is the same.

In SIP uri, Its already, "*" on all the fields..


could it there be something wrong with my asterisk config

can you post your asterisk Outbound route dial plan, thanks..

PS: I cant trace using IPO Sysmon since calls are not landing through the sip trunk.

 

[amriddle01]

If the call isn't hitting the IPO and not showing in monitor then yes, it's an issue in the asterisk. Monitor would show it coming in, regardless of how you had IPO programmed

 

[clarkstyx18]

@amriddle01, yeah its asterisks fault, its just the dial plan, now I can call from any asterisk extension using IPO analog trunks, this will save us from buying 3rd party IP endppoint licenses