IP Trunking with IP Office and S8500

[macpat]

We have two locations, one is an IP Office 403, the other is an S8500. when calls are placed over the IP Trunks from the IPO 403 everything works fine. When you call from the S8500 side it will disconnect at 15 seconds. If the coverpath is set shorter than 15 seconds it will cover to VMPro fine. If the person being called on the IPO 403 side picks up before 15 seconds than the call is fine. This only happens if the phone rings for 15 seconds and no one picks up and it does not cover. Any thoughts? We went to avaya and still can't figure out what is happening.

Any ideas will be greatly appreciated.

 

[bwilch]

There is an Avaya white paper on H.323 which details how to configure an H.323 IP trunk between an IP600 and an IP Office. I'm not sure the exact location of it, because I have a hard copy instead of a soft copy. Just coming out of an IP Office class, a fellow student gave it to me. I'll do a little digging and see if I can find where it's at on the Avaya site. The title of the document is: "How to configure an H.323 IP trunk between an Avaya IP600 Internet Protocol Communication Server and Avaya IP Office Server - Issue 1.1"

Hope that helps a little at least. I'll let you know if I find it online.

 

[petran]

I believe this is the document that bwilch mentions...

http://www1.avaya.com/enterprise/applicationnotes/deftoipoiptrunk.pdf

Petran.

 

[bwilch]

Yep, that's it. Just used it to setup the IP Office to our S8500. The only things you don't have to do in that document is setup the Cajun really. Otherwise, it's pretty straight forward. If you have any specific questions, let us know, I literally just set ours up earlier this week.

 

[digital12188]

Bwilch,

You are able to make calls in both directions with no problems.. No time out going from 8500 to IP office?

 

[bwilch]

Correct, I can dial from an IP Office station (203) to my deskphone (5525) without any problems and continue a conversation. Even the call information (station # and name) are passed across. If you need, I can send you ss's of how we have our trunk group configured.

 

[bwilch]

Here's our sig group:
SIGNALING GROUP

Group Number: 5 Group Type: h.323
Remote Office? n Max number of NCA TSC: 0
SBS? n Max number of CA TSC: 0
Trunk Group for NCA TSC:
Trunk Group for Channel Selection: 5 X-Mobility/Wireless Type: NONE
Supplementary Service Protocol: a


Near-end Node Name: clan-1a02 Far-end Node Name: IPOffice
Near-end Listen Port: 1720 Far-end Listen Port: 1720
Far-end Network Region:
LRQ Required? n Calls Share IP Signaling Connection? n
RRQ Required? n H245 Control Addr On FACility? n
Media Encryption? n Bypass If IP Threshold Exceeded? n

DTMF over IP: out-of-band Direct IP-IP Audio Connections? n
IP Audio Hairpinning? n
Interworking Message: PROGress

Here's the trunk group

TRUNK GROUP

Group Number: 5 Group Type: isdn CDR Reports: y
Group Name: IP Office COR: 11 TN: 1 TAC: 815
Direction: two-way Outgoing Display? n Carrier Medium: IP
Dial Access? y Busy Threshold: 99 Night Service:
Queue Length: 0
Service Type: tie Auth Code? n TestCall ITC: rest
Far End Test Line No:
TestCall BCC: 4
TRUNK PARAMETERS Auto Page Line Retrieval? n
Codeset to Send Display: 0 Codeset to Send National IEs: 6
Max Message Size to Send: 260 Charge Advice: none
Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc

Trunk Hunt: cyclical QSIG Value-Added? n
Digital Loss Group: 13
Incoming Calling Number - Delete: Insert: Format:
Bit Rate: 1200 Synchronization: async Duplex: full
Disconnect Supervision - In? y Out? y
Answer Supervision Timeout: 0

TRUNK FEATURES
ACA Assignment? n Measured: none Wideband Support? n
Internal Alert? n Maintenance Tests? y
Data Restriction? n NCA-TSC Trunk Member:
Send Name: y Send Calling Number: y
Used for DCS? n
Suppress # Outpulsing? n Format: public
Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider

Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Send Connected Number: y
Modify Tandem Calling Number? n
Send UUI IE? y
Send UCID? n
Send Codeset 6/7 LAI IE? y Ds1 Echo Cancellation? n



SBS? n Network (Japan) Needs Connect Before Disconnect? n
DSN Term? n

The last page is just the trunk members. When then setup the AAR via "ch aar ana 2" and put in this line
Dialed Total Route Call Node ANI
String Min Max Pattern Type Num Reqd
20 3 3 5 aar n
This allows any 3 digit call starting with 20 to go through the correct trunk group.

Let me know what other questions you have.

 

[macpat]

I have confirmed that my S8500 side looks exactly like yours. How about the settings on the IP Office 403. Also, our solution works fine when you dial the S8500 from the IP Office. When you dial from the S8500 to the IP Office the problem occurs. It seems to time out at 15 seconds, but again only when you call from the S8500 to the IP Office. Any ideas?

 

[bwilch]

Do you have a shortcode created which would be something like

short code: 5xxx
Tele #: 5NJ
Line Group: (whichever line group you created)
Feat: Dial

The 5xxx is our extenion numbers on my switch, thus why the 5NJ is a 5 instead of another number.

On the IP Line VOIP tap, do you have "enable faststart, out of band DTMF, and Allow direct media path" selected?

 

[macpat]

ours is setup exactly like that. I am at a loss at this point. works fine in one direction but not in the other.

 

[bwilch]

Toss me an email at
bwilch at sbcglobal dot net
and I'll get you some contact information so we can get you setup.

 

[ecscomms]

Hi,

I seem to be having the same kind of problem with my clients network. They have 5 sites which compose of 2 Definitys and 2 IP Office 403/6 and finally we are putting a Small office edition in the UK.

The sites are all linked by a 2Mbs VPN Link, and currently it all works fine with all sites except the UK.

From the UK to definity 1 we can call them and get the call established and have a 15 Sec conversation before the call drops. If a user on Definity 1 calls the UK the Phone rings but there seems to be no speech path once answered. (We get name and number info passed both ways)

From the UK to definity 2 we can call them and get the call established and have no problems. If a user on Definity 2 calls the UK the Phone rings but there seems to be no speech path once answered. (We get name and number info passed both ways)

From the UK to both IP Offices the calls are established and work fine both ways with no problems.

A seperate bit of info is that i can configure a IP4620 to connect to the definty 1 as a IP station and have no issues with bandwidth and call quality.

I am guessing that the issue lies some where on the definity side?

Any help would be appreciated.