Ascom phone in an Avaya Definity System.

[Montero84]

Hello,

In the organization I work for, we have an Ascom phone system and I was told it's possible to make it works in my Definity system using SIP, I have not experience doing that. Does any of you know the complete process to configure this?,

Thanks in advance

LP

 

[Midstream]

What release is your Definity? Hopefully, it's Communication Manager and is SIP compliant.

 

[Montero84]

Hello Rockspop

Thank you for replying. Please forgive my ignorance.. but, how to I check what release do I have?, as far as I know it's a Infinity S8400. And I also know it has Sip compliant.

Thanks again!

 

[Midstream]

You're in pretty good shape then.

From the command line in Communication Manager,

list config software

That will give you the exact release. I know an Ascom dealer who can help.

 

[Montero84]

Hello Rockspop

I used the command list config software and it's what I got

SOFWARE VERSION
Memory residen: R014x.00.1.731.2
Disk Resident: R014x.00.1.731.2

Thanks in advance!

 

[big70]

Im installing now 42 locations with 1500 SIP Ascom phone,s with an multiple Avaya SIP Server and works really good.

http://www.ascom.us/i75protector-ds.pdf

Only importand thing is you need a really good network and wifi network.
I also have R014x.00 and started one year ago and are ready end this year.

Greets Peter

 

[Montero84]

Hello Big70,

Thank you for replying. So, do there's a way I can get some info on how to do this?...

Thanks again!

 

[big70]

You need to work together with ASCOM else its a no go...

In ACM you program a 9640 station with alias I75 and an off station pabx station mapping with OPS and trunk (sip trunk).

In the SIP-Server you program the ASCOM Sip station.

Greets Peter

 

[Montero84]

Big70,

Thank you for the information. I know how to configure an Ascom phone and I have configured a SIP trunk in my Avaya PBX switch. I also configured a 9640 station with alias I75... example.. in my ACM I have the station 5545. how do I configure that one in the SIP-Server?, another question... how do I configure a Off PBX station (OPS). what does it exactly do?

I am so sorry to bother you with this.

Thanks again,

Lenin

 

[big70]

The Off pabx (OPS) makes the connection with the SIP Server. (Like EC500) You need to make a user and a extension in the SIP Server.

Change off-pabx station mapping 5545.

Extension Application Phone Number Trunk Configset
5545 OPS 5545 trunk SIP 1

Nice thing is you can have the 9640 registered and the SIP phone also. And if you call 5545 both phone will ring.
So no need of bridge appearance.

 

[Montero84]

Good morning Big70

thanks again for your help.. I have a question, when I set up the extension 5545.. should I configure a coverage remote or something like that to jump to the Ascom switch?. I configured the extension 5545 in my Ascom system and also in the Sip server and in the definity system, I can call from the Ascom phone to any other phone but I can't call from the Definity switch to the 5545.

I configured the off-pbx-station but I am not sure if i am missing some configuration.. maybe a coverate remote or something like that?

Thanks again!

 

[big70]

Nope no coverpath needed its the OPS and what you filled in as trunk whats making the SIP phone to ring.

If you do a list trace of the phone thats calling what do you see.

And you use the I75 (SIP) wright?

If you call from the I75 to a avaya phone you see the name and number from 5545...

Greets Peter

 

[Montero84]

Hello big70.

In my uniform-dialplan I have the:
Ext Len Del Insert Digits Net conv Node Num
5545 4 0 aar n

In off-pbx-telephoone station-mapping I have:

Station Dial config
Extension Aplication Prefix CC Phone Number Trunk set
5545 OPS 5545 30 1

When I dial from Avaya from ext 5564 to Ascom dialing 5545<-- it's also the extension number in the Ascom station.. when I do list trace, I get this:

LIST TRACE

time data

14:39:32 Calling party station 5564 cid 0x132
14:39:32 Calling Number & Name 5564 Perez, Lenin
14:39:32 dial 5545
14:39:32 term station 5545 cid 0x132
14:39:32 dial 5545
14:39:32 seize trunk-group 30 member 97 cid 0x134
14:39:32 Calling Number & Name NO-CPNumber NO-CPName
14:39:32 Proceed trunk-group 30 member 97 cid 0x134
14:39:32 denial event 1180: No user responding D1=0xe0011 D2=0x12
14:39:32 idle trunk-group 30 cid 0x134
14:39:47 idle station 5564 cid 0x132

Seem like we are almost close... please help me!

Thanks!

 

[Montero84]

Hello bit70

I made a list trace for the ext 5564, then I call from the Ascom phone ext 5545 and I see this:


LIST TRACE

time data

15:03:30 Calling party trunk-group 31 member 4 cid 0x240
15:03:30 Calling Number & Name NO-CPNumber NO-CPName
15:03:30 dial 5564
15:03:30 ring station 5564 cid 0x240
15:03:30 G711MU ssff ps:20 rn:2/1 10.111.238.113:48836 10.111.240.248:2868
15:03:30 xoip: faxT modem:Relay tty:US 10.111.240.248:2868 uid:0x919
15:03:30 G711MU ssff ps:20 rn:1/1 10.111.240.249:3016 10.111.240.248:2864
15:03:30 xoip: faxT modem:Relay tty:US 10.111.240.248:2864 uid:0x50004
15:03:38 idle trunk-group 31 member 4 cid 0x240

Thanks!

 

[big70]

How is your setup?

UDP is not used wit Ascom SIP??

Are you using the Ascom I75 SIP or do you have a Ascom system with a Q-sig setup...

Looks like we are talking about two difrent things.....

Greets Peter

 

[Montero84]

Hello Big70

My SIP Trunk type setup in my SES is "TLS", I can select TCP or TLS. I am not sure about the trunk in the Ascomo side. We use the I75 wireless phone, so I guess we have Ascom I75 SIP.

Please advise.

Thanks!

 

[big70]

(I guess) is not enough i know ascom also has its own h323 gateway were you make a q-sig trunk.

Phone,s can be SIP or H323, this can be changed in the PDM from ASCOM.

You need to work with the guys from ASCOM

So if its SIP directly use a station with OPS and the Avaya SIP-Server if its an ASCOM Gateway you need to use UDP and a Q-sig Trunk.

Greets Peter

 

[Montero84]

Hello Big70

I hope you can get this reply... and I hope you are still willing to help me with this project..

I have been working with the PDM from ASCOM and I know where to change the configuration for SIP or H323 in the PDM... I have my Avaya SIP server configured I am wondering if you would be willing to help me with the UDP and the Q-sig trunk...

Thanks again!