Help redirecting 6xxx numbers over PRI to asterisk

[Havokmon]

Hi guys,

I'm trying to create a set of routes to allow anyone who dials 6xxx 'extensions' on the Fujitsu to transfer to an asterisk. I have an asterisk connection via a PRI, and I can call Fujistu extensions from the asterisk, but I can't figure out how to go the other way.

My TG is 211, and I created an NP:
Access Expect Feature
Code Digits
6 4 457:AAR

I assume overlap sending means the '6' will not be stripped from the number.

I didn't define anything additional in the Numbering plan for that entry.

I then created an AARLC:
CODE XLC ARSTB EDL
6 11 4
And I created an ARS route:
11 0 1 211 0 0 ******** Next Disable Dial 0 0

But I get 'Denied' when I try to dial 6000 (which exists on the Asterisk) What did I do wrong?
Whoops. I fixed the COS restrictions on AAR Now I get 'invalid number'. I don't think the 6 is being stripped, if I call '000' on the asterisk, I get a 'cannot be completed as dialed'.. It doesn't seem to be leaving the Fujitsu.. And now I see it seems to want at least 7 digits for anything to happen. I'm not sure how to get it to process and connect with only 4 :/

Any suggestions would be appreciated.

Thanks
Rick

 

[ksphoneman]

Change the 6 access code to
ASS NPSYS,,,6,4,304

 

[Havokmon]

Hmm I still get 'Invalid Number'.. I have to dial 66204, the 6 to get into AAR and 6204 to trigger my route, so I assume. But it doesn't seem to want to push the call down the trunk.
Any other hints?

 

[ksphoneman]

Oops,
Also make sure SVP 2, 67 is 1
67 AAR access code 00:* Dialed
01: Unnecessary to dial
CHA SVP,2,67,1

 

[Havokmon]

Ahh. Ok, that stopped the extra '6' requirement. I think my problem is that this is a PRI, and I don't see where the TG knows how to send OUT the PRI. I see in DIS TRK where straight T1s are associated with a TG, but I don't see where to do that for the PRI.. and ass trk doesn't work for PRIs.

Also, this PRI was already configured, and we didn't have any outgoing traffic, so I'm not sure if it's setup for that... and I can't figure out what the command would be :/

Almost there

 

[ksphoneman]

The PRI trunk group should be a Trunk line type of TIE.(TLT=3). The ARS route is what passes the dialed number to the tandem system.

In this example several number ranges are routed from the PSTN to the Fujitsu F9600 to the Cisco Call Manager. At this site the F9600 supports 10,000 DID numbers and 17 Fujitsu Integrated Private Network (FIPN) sites and approximately 5000 users.
It supports Name and Number delivery between the two systems.
Incoming PSTN calls tandem through the F9600 to the CCM. CCM PSTN calls tandem back through the F9600 to the PSTN.

*** Note the Fujitsu PBX must have the following software to use the DMS100 protocol or the NI1/NI2 protocol to a Cisco Call Manager.
IPRCBS – ISDN Primary Rate Central Office Basic Service.

**** The following hardware is required on the Fujitsu F9600 system
1- BPTKCA, 1-B2DK2B plus the associated cables or 1-BPTKEA plus the associated cables.
See the Circuit Card guide for further details.

Build two trunk groups for the ISDN service, one is for B-channels and one is for the D-channel.
# TRUNK GROUP DATA LIST #

TGN TYP TID TNN SPC AKI COF TLT DGN RGN COS RSM FRL TRS HNT NAME
AKW AKR AKB RGT AOT GRD REL HKS AFT SHK RHK OPR DMF
MIN PRE MAK BRK DGT PST PBO PBF COP PGT MID
PAC MBC STG DT IAS DTS ABS DTK OOC NOC PTF TCS TCR TDT VCM OGF
CRC
NSF NSFFG PRMFF PRMFV CDNFG TON NPI
(Cisco TIE trunk B channels trunk group)
500 5 38 0 5 0 0 3 0 1 1 5 7 0 1 'VOIP'
0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0
28 0 0 0 0 0 0 0 1 21 0 0 0 0 0

0 0 0 0 0 0 0
(Cisco TIE trunk D channels trunk group)
501 5 37 0 4 0 0 0 1 1 1 1 1 0 0 'VOIP'
0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0
28 0 0 0 0 0 0 0 0 0 0 0 0 0 0

0 0 0 0 0 0 0

=>DIS ISINF,500


(PSTN is connected to the Fujitsu F9600, the Cisco CM is behind the Fujitsu. The Fujitsu RX the clocks from PSTN LD circuits and TX this clock to the Cisco CM)
*** Note if the Cisco is in front of the Fujitsu, the DMS100 protocol causes ISDN Information Element errors to the Cisco, therefore use the NI2 protocol instead, this disables Name & Number to/from the Fujitsu. The error is caused by a status message sent from the Fujitsu that the Cisco does not expect and causes calls to drop)
One other note, the names on the CCM must be all upper case to display properly on the F9600, this may be a character set problem that I have not gotten through yet.

# ISDN TRUNK INFORMATION LIST #

# ONE-INTERFACE #

< D-CHANNEL >

TNN TGN EN MS PRTCL

0 501 06081800 1 DMS100
(MS=1 Fujitsu is the master of the connection to Cisco, this site uses two ISDN PRI trunks)

< B-CHANNEL >

TNN TGN EN QDID FNA SUBDID DID UNA RSS RSASG TYP NSF

0 500 06081802 - - - - - - - MUL 0
0 500 06081803 - - - - - - - MUL 0
0 500 06081804 - - - - - - - MUL 0
0 500 06081805 - - - - - - - MUL 0
0 500 06081806 - - - - - - - MUL 0
0 500 06081807 - - - - - - - MUL 0
0 500 06081808 - - - - - - - MUL 0
0 500 06081809 - - - - - - - MUL 0
0 500 06081810 - - - - - - - MUL 0
0 500 06081811 - - - - - - - MUL 0
0 500 06081812 - - - - - - - MUL 0
0 500 06081813 - - - - - - - MUL 0
0 500 06081814 - - - - - - - MUL 0
0 500 06081815 - - - - - - - MUL 0
0 500 06081816 - - - - - - - MUL 0
0 500 06081817 - - - - - - - MUL 0
0 500 06081818 - - - - - - - MUL 0
0 500 06081819 - - - - - - - MUL 0
0 500 06081820 - - - - - - - MUL 0
0 500 06081821 - - - - - - - MUL 0
0 500 06081822 - - - - - - - MUL 0
0 500 06081823 - - - - - - - MUL 0
0 500 06081824 - - - - - - - MUL 0

< CBC GROUP TGN >

PILOT MTGN - NSF MTGN - NSF MTGN - NSF MTGN - NSF

NONE

# ONE-INTERFACE #

< D-CHANNEL >

TNN TGN EN MS PRTCL

0 501 06161600 1 DMS100

< B-CHANNEL >

TNN TGN EN QDID FNA SUBDID DID UNA RSS RSASG TYP NSF

0 500 06161602 - - - - - - - MUL 0
0 500 06161603 - - - - - - - MUL 0
0 500 06161604 - - - - - - - MUL 0
0 500 06161605 - - - - - - - MUL 0
0 500 06161606 - - - - - - - MUL 0
0 500 06161607 - - - - - - - MUL 0
0 500 06161608 - - - - - - - MUL 0
0 500 06161609 - - - - - - - MUL 0
0 500 06161610 - - - - - - - MUL 0
0 500 06161611 - - - - - - - MUL 0
0 500 06161612 - - - - - - - MUL 0
0 500 06161613 - - - - - - - MUL 0
0 500 06161614 - - - - - - - MUL 0
0 500 06161615 - - - - - - - MUL 0
0 500 06161616 - - - - - - - MUL 0
0 500 06161617 - - - - - - - MUL 0
0 500 06161618 - - - - - - - MUL 0
0 500 06161619 - - - - - - - MUL 0
0 500 06161620 - - - - - - - MUL 0
0 500 06161621 - - - - - - - MUL 0
0 500 06161622 - - - - - - - MUL 0
0 500 06161623 - - - - - - - MUL 0
0 500 06161624 - - - - - - - MUL 0

< CBC GROUP TGN >

PILOT MTGN - NSF MTGN - NSF MTGN - NSF MTGN - NSF

NONE

=>DIS SVP,2,67 (This service parameter will cause the F9600 to pass the AAR access code to the next node)
# SERVICE LIST #
TYPE = 2 ( SVSDT )
ID-----DATA
67 1


=>DIS NP,,,308
(Example of a number that belongs to the Cisco, all Fujitsu numbering/dialing plans are displayed)
****Note this access code can also be created as a TIE trunk access code (FNO 512~767).
Remember to add the FNO to the Class of Service assignments).
This implementation uses AAR access codes. By use of AAR, an ARS route is used to send the calling party name.

# NUMBERING PLAN LIST #

TNN= 0

TTID= 0 (General dial plan)

DIGIT EDL FNO TGN TGX AJC RDD DOC TTN DN SVN

308 5 304 0 3 0

TTID= 1 (Hook flash dial plan)

DIGIT EDL FNO TGN TGX AJC RDD DOC TTN DN SVN

308 5 304 0 3 0

TTID= 4 (Attendant console dial plan)

DIGIT EDL FNO TGN TGX AJC RDD DOC TTN DN SVN

308 5 304 0 3 0

TTID= 7 (TIE trunk dial plan)

DIGIT EDL FNO TGN TGX AJC RDD DOC TTN DN SVN

308 5 304 0 3 0

TTID= 8 (CO trunk dial plan)

DIGIT EDL FNO TGN TGX AJC RDD DOC TTN DN SVN

308 5 304 0 3 0


=>DIS AARLC,,308
(Automatic Alternate Route location code table for 308XX)

# AAR LOCATION CODE LIST #

TNN CODE XLC ARSTB EDL CODE XLC ARSTB EDL CODE XLC ARSTB EDL

0 308 500 5



=>DIS ARSR,500
(Automatic Route Selection for access code 308XX)

# ARS ROUTE TABLE LIST #

ARSTB TNN POS TGN FRL PTNNO T0 T1 T2 T3 T4 T5 T6 T7 LARF CMPF
(NARSTB) CDNFG TON NPI NAMF

500 0 1 500 0 0 * * * * * * * * 0 0
0 0 0 1 (NAMF used to send the name on this route, and is why AAR is used.)


=>DIS SVP,2,180 (Service parameters that effect Name & Number on tandem calls)

# SERVICE LIST #

TYPE = 2 ( SVSDT )

ID-----DATA ID-----DATA ID-----DATA ID-----DATA ID-----DATA

180 1 (Enable DMS100 Name display)

(Other service parameters in Group 2)

217 Calling party number to PRI-CO when 00:* Station calling line ID
TLTYP is TIE 01: Station DN (ISSLC + DN)
218 Calling party number when connecting 00:* Not provided
from PRI-TIE or PRI-CO to PRI-CO 01: Provided
222 Waiting trunk information display for call 00:* ‘OUTSIDE CALL’
waiting 01: Trunk information or calling
party information
(SVP 2,217= 0 to use the stations CLIST assignment SVP,2,218= 1 to pass through the CLID on a tandem call)
(SVP,2,222=1 to send CLID during call waiting.)

 

[Havokmon]

I've started over with a freshly installed PRI card, and an existing TG that hasn't been used in years. It used to be a local PRI TG, with calls flowing bothways. I've assigned this TG to the new PRI with ISTRK.

When I change my ARSR to TG 150 instead of 211, I now get 'Number Restricted'.

It's been a long time since I've installed a T1 card, but I don't remember it being this difficult to get calls to route out it

 

[ksphoneman]

After assigning the trunks did you SET PTMI for all of the trunk channel equipment numbers?

 

[Havokmon]

Nope, but I was able to receive calls from the asterisk over this PRI. For giggles I've done the ptmi now. No difference.

I've now totally started over, and followed the above instructions to create my TGs new - after taking it too literally, and setting up national instead of 5ess, I've got it back up.

But now I've lost DNIS
Oh wait, I had it as a CO, and not a TIE. All good now. I can dial my Fujitsu extension from the asterisk again.
(heh funny - I'm using kphone as my asterisk sip client, but I have no mic. While I'm wondering why my phone isn't ringing, the operator - who happens to be my wife - is wondering why there is no one on the other end with this weird 6204 #


I still get 'Number Restricted' when I try and dial 6204 from the Fujitsu. It seems like a FRL or COS issue, but I have the COS set right. My DT200 flashes AAR twice, then says 'Number Restricted'.

I've got the manuals, and I even have my old EMML I course book (nothing on PRIs or ARS though :/ ) - I still have no idea what the restriction is.

 

[Havokmon]

It's definitely a restriction of some sort, if I put a working outgoing trunk as step 1 in my route 11, I still get restricted.

I don't know where else to look.

 

[Havokmon]

A little further - I deleted my aarlc and re-created it.
I seem to have set ID=1. Now that It's 0 (other), I can go out my 'valid' TG PRI in my route 11.

I also discovered my TG had the wrong settings. I was comparing with my last test TG, which didn't work either. Setting it to be the same as my working primary PRI TG seems to have worked, but now I get 'Busy Camp=On?'

Just a thought - Since this is connected to another PBX, should it be verifying there is a dial tone?

 

[ksphoneman]

SEND ME YOUR EMAIL
moneil at allegiantusa dot com

 

[ksphoneman]

=> DIS COS,,301,306 (Add FNO 304 or 305 or 306 to the existing station COS. The FNO used will depend on your existing PBX design. If AAR has never been used then use FNO 304. ASS COS,,304,SCOS,ECOS

# COS CHECK TABLE LIST #
Cos Matrix (* = allowed)
1 2 3 4 5 6
FEATURE 0123456789012345678901234567890123456789012345678901234567890123
.
ARS Unrestricted
301: ****************************************----------*-------------
ARS Defined
302: ----------------------------------------------------------------
ARS Restricted
303: ----------------------------------------------------------------
AAR Unrestricted
304: ****************************************----------*-------------
AAR Defined
305: ----------------------------------------------------------------
AAR Restricted
306: ----------------------------------------------------------------

 

[Havokmon]

Yep, I've already done 0-6, or giggles I just opened up 0-60
ASS COS,,304,0,60

Still get "Busy Camp On" on the DT200. It seems the PBX thinks the trunk is busy. Oh duh. It's busy because I didn't SET PTMI after re-doing everything.

Now I get Invalid Number. Ugh. And I see nothing in my SMDR log that it went out the TG.

It's time for the weekend and a stiff drink.

 

[Havokmon]

Oh! Asterisk is rejecting the call - so it IS going down the trunk! I think I may be good from the Fujitsu side!

Thanks!