Odd outbound problem: call progress not end to end ISDN? 1

[jermudgeon]

Hi all. Partial success story here:

I have an Option 11C that I've hooked up via a T1, TIE lines, to an Asterisk VOIP system. I've used some of the walkthroughs from this forum & other places on the web.

Using steering codes, calls work great in both directions, from Asterisk to the Nortel, and back. Love it. No problems.

Sample DCH messages:

DCH 4 IMSG SETUP blah blah
CALLING #:4460 NUM PLAN: E164
CALLED #:5411 NUM PLAN: E164

DCH 4 OMSG ALERT blah blah
DCH 4 OMSG CONNECT
DCH 4 IMSG CONN ACK

and upon hangup, the proper DISC/ NORMAL CALL CLEARING/RELEASE messages are generated.

Making a call that goes out over our T1 (COT setup) to the PSTN fails in this regard:

The call is completed, audio works, but the CONNECT message never is sent, so Asterisk thinks the call hasn't been answered, and hangs up after 5 minutes, which is the max ringtime I have it sent to.

Sample DCH messages:

DCH 4 IMSG SETUP
CALLING #:4460
CALLED #:9XXXXXXX (obscured)

DCH 4 OMSG CALLPROC
DCH 4 OMSG PROGRESS
PROGRESS: CALL IS NOT END TO END ISDN

And no connect message is generated.
Disconnect supervision works fine from either side in a call like this, a normal OMSG DISC or IMSG DISC.

Bad consequences:
1. Unless I set my call timeout to never from the Asterisk side, outbound calls get forcibly dropped
2. Call duration never gets logged to CDR in Asterisk (this is sort of OK, because I have the CDR from Nortel)
3. Any ACDs on the Option 11C that NCFW to to an "outside" line (i.e., over our PSTN trunks) just don't work at all from the Asterisk side. This is the worst and may be an unrelated problem, sadly.

I'm happy to post any configs, version numbers, etc. anyone can suggest. This has been a problem for about 6 months, and I could really use a solution.

Thanks to any and all,
Jeremy Austin
7 years Nortel Admin
5 years Asterisk
Nortel Meridian Option 11C (about 120 500's)
Asterisk VOIP (3 systems, 20 clients)

 

[Avaya911PLM]

CALL IS NOT END TO END ISDN just means the 2 phones themselves are not ISDN phones.

This is normal and not an error in itself.

--
Fletch

Nortel Emergency Services PLM

http://nortel.com/e911

Don't forget about the FREE E911 Webinar Oct. 30th!
Register online at

http://thevoicereport.com/E911

 

[jermudgeon]

Thank you, Fletch -- at least that's a partial solution to the mystery.

Still need to figure out:

1.) ACDs or DNs forwarded to external #s don't work over the TIE lines
2.) Call progress not reported over TIE to the Asterisk side -- calls never 'answered'

Thanks!
jermudgeon

 

[yyrkroon]

please provide an example of the call flow - end to end

 

[jermudgeon]

Outbound calls from asterisk still fail--I haven't fixed this problem yet. Any advice welcome. All routing working properly on the Nortel side from a native extension.

Architecture:
PSTN <-> COT lines (on a T1) <-> Nortel Meridian <-> TIE lines <-> asterisk VOIP <-> SIP clients

Calls that work:
1. Call from SIP client to Nortel extension
2. Call from Nortel extension to SIP client
3. Call from PSTN -> Nortel -> SIP client

Calls that fail:
4. Call from SIP client -> Nortel -> PSTN

Sample call progress reports from Nortel, trimmed for clarity:

1. SIP client to Nortel extension
IMSG SETUP
OMSG ALERT
OMSG CONNECT
IMSG CONN ACK
with the usual messages on call disconnect & release.

2. Nortel extension to SIP client
OMSG SETUP
IMSG CALLPROC
IMSG ALERT
IMSG CONNECT
OMSG CONN ACK

3. Call from PSTN to SIP client
IMSG SETUP
OMSG CALLPROC
IMSG ALERT
IMSG CONNECT
OMSG CONN ACK

4. And failure, call from SIP client to PSTN
IMSG SETUP
OMSG CALLPROC
OMSG PROGRESS
If I listen in on the call with a client that transmits audio immediately, I can hear the call is terminating successfully. But asterisk never receives the CONNECT message, and so the call is still in status as Dialing, and times out after 300 seconds. (I can adjust the timeout at will, but that's a poor workaround. CDR messages are incorrect, for example.)

I can post the configuration of my trunk lines & routes if it would help.

 

[GHTROUT]

Someone posted a link to this once, and I copied the file:

http://www.ghtrout.com/ghtrout/files/Asterisk_PBX_to_CS1000.pdf

You may already have it - and I have no idea if it will be on any help, but it is on topic.

The source/author is clearly listed in the file.

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[jermudgeon]

GHTROUT, thanks for posting the link. That's the model I worked from to set this up initially. Digit passing is working great; all my steering codes (and equivalent asterisk routes) work well.

The only thing I can think of is that it might be some sort of supervision issue? Except calls from PSTN <> Nortel have always worked well, connecting and disconnecting properly.

Here's a sample PSTN COT line:

TN 001 01
TYPE COT
CDEN SD
CUST 0
NCOS 0
RTMB 5 1
A/B BIT SIGNALING
ATDN 7999
SIGL GRD
SUPN YES
AST YES
IAPG 0
CLS UNR DTN WTA LPR APN THFD
P10 NTC

-----

And the route to which the COT belongs:

TYPE RDB
CUST 00
DMOD
ROUT 5
DES 4938
TKTP COT
PRIV NO
SAT NO
IDEF NET
RCLS EXT
DTRK YES
DGTP DTI
ISDN NO
DSEL VOD
PTYP DCO
AUTO YES
ICOG IAO
RANX NO
SRCH LIN
TRMB YES
STEP
ACOD 9
CPP NO
TARG
CLEN 1
BILN NO
OABS
TIMR ICF 512
OGF 512
EOD 13952
DSI 34944
NRD 10112
DDL 70
ODT 4096
RGV 640
FLH 510
GRD 896
SFB 3
CRD 512
TFD 0
LEXT 100
SST 3 0
NEDC ETH
FEDC ORG
CPDC NO
SPCT IMM
HOLD 02 02 40
SEIZ 02 02
RGFL 02 02
RVSD 08 31
ILLR 02 02
DRNG NO
CDR YES
INC YES
LAST YES
TTA YES
ABAN YES
CDRB YES
QREC YES
OAL YES
AIA YES
OAN YES
OPD YES
NATL YES
MUS YES
MRT 3
MR NO
MANO NO
EQAR NO
FRL 0 0
FRL 1 0
FRL 2 0
FRL 3 0
FRL 4 0
FRL 5 0
FRL 6 0
FRL 7 0
TTBL 0
OHTD NO
PLEV 2
MCTS YES
MCCD 8
MCDT 0
ALRM YES
ART 0
SGRP 0
AACR NO

---------

I'm at my wits' end, or at least I can see it from here.

Thanks to all.

 

[GHTROUT]

I *think* we need to see the TIE route/sample trunk between the Asterisk and the Nortel. This because I think it's a call from the Asterisk > Tie > Nortel > COT that fail? Right?

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[tnphoneman]

Post a tie trunk and RDB from the Nortel.

Signature===========================================

http://www.greatwhitetechnologies.com

Aastra Authorized Reseller

 

[GHTROUT]

Alternatively, posting a TIE route and trunk would be helpful.

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[jermudgeon]

tnphoneman and ghtrout,

Thanks for looking at this.

I've been watching the DCH for the TIE lines, but since I have only 'analog' (COT) trunks from my PSTN provider, there's no D-channel to debug. I have an Option 11C, which has no digital tracing! Aargh!

Here's the TIE route:

TYPE RDB
CUST 00
DMOD
ROUT 12
DES VOIP
TKTP TIE
ESN NO
CNVT NO
SAT NO
IDEF LOC
RCLS INT
DTRK YES
DGTP PRI
ISDN YES
MODE PRA
IFC ESS5
SBN NO
SRVC NNSF
NCNA YES
NCRD YES
CHTY BCH
CTYP UKWN
INAC NO
ISAR NO
TGAR 0
BCOT 0
DSEL VOD
PTYP PRI
AUTO NO
DNIS NO
DCDR NO
ICOG IAO
SRCH LIN
TRMB YES
STEP
ACOD 89
TCPP NO
PII NO
TARG
CLEN 1
BILN NO
OABS
INST
IDC NO
DCNO 0 *
NDNO 0
DEXT NO
ANTK
SIGO STD
ICIS YES
TIMR ICF 512
OGF 512
EOD 13952
NRD 10112
DDL 70
ODT 4096
RGV 640
GRD 896
SFB 3
NBS 2048
NBL 4096
TFD 0
DRNG NO
CDR NO
MUS NO
RACD NO
FRL 0 0
FRL 1 0
FRL 2 0
FRL 3 0
FRL 4 0
FRL 5 0
FRL 6 0
FRL 7 0
AUTH NO
TTBL 0
PLEV 2
ALRM NO
ART 0
SGRP 0
AACR NO


------
and here's a TIE member:

TN 002 01
TYPE TIE
CDEN SD
CUST 0
NCOS 0
RTMB 12 1
B-CHANNEL SIGNALING
TGAR 0
AST NO
IAPG 0
CLS UNR DTN WTA LPR APN THFD HKD
P10 VNL
TKID

 

[jermudgeon]

I was wrong about digital tracing.

Here's a test call from a digital set:

.09:13:30 01/04/2009
ACTIVE TN 001 12
ORIG 003 0 00 15 0 SCR MARP 0 xxxx (obscured number)
TERM 001 12 COT RMBR 5 12
DIAL DN 9
MAIN_PM DIAL AUX_PM DDL
TALKSLOT ORIG 8 TERM 8
QUEU 128
CALL ID 0 18

.09:13:30 01/04/2009
ACTIVE TN 001 12
ORIG 003 0 00 15 0 SCR MARP 0 xxxx (obscured number)
TERM 001 12 COT RMBR 5 12
DIAL DN 9xxx
MAIN_PM DIAL AUX_PM TEMPPATH
TALKSLOT ORIG 8 TERM 8
QUEU 128
CALL ID 0 18

.09:13:32 01/04/2009
ACTIVE TN 001 12
ORIG 003 0 00 15 0 SCR MARP 0 xxxx (obscured number)
TERM 001 12 COT RMBR 5 12
TDTN 0 SLOT 13 PTY SLOT 8
DIAL DN 9xxxxxxx
MAIN_PM DIAL AUX_PM NOOUTPULSE
TALKSLOT ORIG 8 TERM 8
QUEU 128
CALL ID 0 18

.09:13:36 01/04/2009
ACTIVE TN 001 12
ORIG 003 0 00 15 0 SCR MARP 0 xxxx (obscured number)
TERM 001 12 COT RMBR 5 12
TDTN 0 SLOT 13 PTY SLOT 8
DIAL DN 9xxxxxxx
MAIN_PM DIAL AUX_PM OUTPULSE
TALKSLOT ORIG 8 TERM 8
QUEU NONE
CALL ID 0 18

.09:13:38 01/04/2009
ACTIVE TN 001 12
ORIG 003 0 00 15 0 SCR MARP 0 xxxx (obscured number)
TERM 001 12 COT RMBR 5 12
TDTN 0 SLOT 13 PTY SLOT 8
DIAL DN 9xxxxxxxxxxx
MAIN_PM DIAL AUX_PM NOOUTPULSE
TALKSLOT ORIG 8 TERM 8
QUEU 128
CALL ID 0 18

09:13:46 01/04/2009
ACTIVE TN 001 12
ORIG NONE
TERM 001 12 COT RMBR 5 12
DIAL DN 9xxxxxxxxxxx
MAIN_PM HALFDISC
TALKSLOT NONE
QUEU NONE
CALL ID 0 18

-----------------
And a test call over the TIE lines:

DCH 4 IMSG SETUP REF 0000801E CH 2 1 TOD 9:14:04
09:14:04 01/04/2009
ACTIVE TN 001 11
CALLING #:xxxx NUM PLAN: E164
ORIG 002 01 TIE RMBR 12 1
CALLED #:9xxxxxxx NUM PLAN: E164
TERM 001 11 COT RMBR 5 11

DCH 4 OMSG CALLPROC REF 0000801E CH 2 1 TOD 9:14:04
DIAL DN 9xxxxxxx
MAIN_PM DIAL AUX_PM DDL
TALKSLOT ORIG 19 TERM 19
QUEU 128
CALL ID 0 30


DCH 4 OMSG PROGRESS REF 0000801E CH 2 1 TOD 9:14:04
PROGRESS: CALL IS NOT END TO END ISDN

---- ISDN PRA CALL (ORIG) ----
CALL REF # = 30
BEARER CAP = VOICE
CALL STATE = 7 CALL RCV
CALLING NO = xxxx
CALLED NO = 9xxxxxxx

.09:14:04 01/04/2009
ACTIVE TN 001 11
ORIG 002 01 TIE RMBR 12 1
TERM 001 11 COT RMBR 5 11
DIAL DN 9xxxxxxx
MAIN_PM DIAL AUX_PM TEMPPATH
TALKSLOT ORIG 19 TERM 19
QUEU 128
CALL ID 0 30


---- ISDN PRA CALL (ORIG) ----
CALL REF # = 30
BEARER CAP = VOICE
CALL STATE = 7 CALL RCV
CALLING NO = xxxx
CALLED NO = 9xxxxxxx




.09:14:04 01/04/2009
ACTIVE TN 001 11
ORIG 002 01 TIE RMBR 12 1
TERM 001 11 COT RMBR 5 11
TDTN 0 SLOT 10 PTY SLOT 19
DIAL DN 9xxxxxxx
MAIN_PM DIAL AUX_PM OUTPULSE
TALKSLOT ORIG 19 TERM 19
QUEU 128
CALL ID 0 30


---- ISDN PRA CALL (ORIG) ----
CALL REF # = 30
BEARER CAP = VOICE
CALL STATE = 7 CALL RCV
CALLING NO = xxxx
CALLED NO = 9xxxxxxx




.09:14:06 01/04/2009
ACTIVE TN 001 11
ORIG 002 01 TIE RMBR 12 1
TERM 001 11 COT RMBR 5 11
TDTN 0 SLOT 10 PTY SLOT 19
DIAL DN 9xxxxxxx
MAIN_PM DIAL AUX_PM NOOUTPULSE
TALKSLOT ORIG 19 TERM 19
QUEU 128
CALL ID 0 30


---- ISDN PRA CALL (ORIG) ----
CALL REF # = 30
BEARER CAP = VOICE
CALL STATE = 7 CALL RCV
CALLING NO = xxxx
CALLED NO = 9xxxxxxx




.09:14:20 01/04/2009
ACTIVE TN 001 11
ORIG 002 01 TIE RMBR 12 1
TERM 001 11 COT RMBR 5 11
DIAL DN 9xxxxxxx
MAIN_PM ESTD
TALKSLOT ORIG 19 TERM 19
QUEU NONE
CALL ID 0 30


---- ISDN PRA CALL (ORIG) ----
CALL REF # = 30
BEARER CAP = VOICE
CALL STATE = 7 CALL RCV
CALLING NO = xxxx
CALLED NO = 9xxxxxxx




.
DCH 4 OMSG DISC REF 0000801E CH 2 1 TOD 9:14:40
09:14:40 01/04/2009 CAUSE :NORMAL CALL CLEARING

ACTIVE TN 001 11

DCH 4 IMSG RELEASE REF 0000801E CH 2 1 TOD 9:14:40
ORIG NONE
CAUSE :NORMAL CALL CLEARING
TERM 001 11 COT RMBR 5 11

DCH 4 OMSG REL COMP REF 0000801E CH 2 1 TOD 9:14:40
DIAL DN 9xxxxxxx
MAIN_PM HALFDISC
TALKSLOT NONE
QUEU NONE
CALL ID 0 30

 

[telebub]

Your post mentioned you have looked at other threads but I copied a link to one below where I helped a gentleman with a PRI to Asterisk installation. You may have read through this one already but in case you haven't, here you go. Their setup was similar in that they had analog trunks from the PSTN with a PRI over to the Asterisk. I don't know what type of phones he had on the Asterisk. Hope this helps!

http://www.tek-tips.com/viewthread.cfm?qid=1533379&page=16

War Eagle!
Lions Baseball '09!

 

[jermudgeon]

telebub,
Thanks for posting that link. I have now read through it, but I'm not sure what's applicable.

I think TGAR and TARG are all OK, since calls placed do actually terminate.

I checked the DSEL on both routes; they're both set to VOD, although the COT lines (T1 to PSTN) == ISDN NO, and the TIE lines (T1 to Asterisk) == ISDN YES. I would try with them both set to VCE, but I don't want to have to rebuild the route. Must I do this anyway? The lines on the T1 to the PSTN get frequent use; the Asterisk is the new 'experimental' side.

I noticed this line: "ground start trunks do not need supervision." I have supervision on; I'm going to test it now with it off, since my PSTN trunks are ground start (although not analog).

Thanks,
Jeremy

 

[jermudgeon]

FIXED!

Supervision on ground start lines was the problem. Our telco + original programmer had supervision on, and it always 'worked', so wasn't a problem before adding the VOIP side.

With SUPN = OFF, the DCH message CONNECT comes through about 8 seconds after the call is actually answered, but it does arrive.

Thanks ghtrout, tnphoneman, and telebub!
jermudgeon

 

[telebub]

Good deal! I have seen 2 issues in the past with SUPN = YES on Ground Start trunks. I always recommend it be set to NO on those trunks as I have never seen an issue with them set to NO. Now I can add a 3rd issue to the list.



War Eagle!
Lions Baseball '09!

 

[GHTROUT]

Outstanding telebub. Now I converted too

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