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New Member

PBX Integration

I have CCM 4.1(3) and Unity 4.0(3). A C1700-IPVoice-M Gateway Router is providing the integration with the PBX system to accomodate 4-digit dialing to/from both ends. 4-digit dialing from the VoIP side to the Non-VoIP side works fine. 4-digit dialing from the Non-VoIP end to the VoIP works but not the way it should. When I dial 4-digits from the Non-VoIP end it accepts the four digits, but it only gives me a dial tone after that. If I then press the four digits that I want to dial the call goes through. It is almost like the first four digits are for establishing a dial tone. I did debug isdn and it shows that it establishes a layer 3 connection after the first four digits dialed. This is very odd and I am not sure wether it is on the gateway or CCM.

I have attached a copy of the config file for the gateway.

Any suggestions would be appreciated.

1 ACCEPTED SOLUTION

Accepted Solutions
New Member

Re: PBX Integration

Direct inward dialing needed?

19 REPLIES
New Member

Re: PBX Integration

Direct inward dialing needed?

Re: PBX Integration

Absolutely. And you can verify that DID is the problem by dialing a valid voip extension when you get that dialtone back from the gateway.

In addition to a direct-inward-dial on your pots dial-peers, try adding an "incoming called x..." where x is the first digit of your voip 4 digit extensions ie, 8... This way you ensure that you match a dial-peer to an inbound call-leg.

New Member

Re: PBX Integration

Thanks,

Adding direct-inward-dial resolved it, I also added, as you recommended, the "incoming called-number ..." line to my config. I appreciate the help.

Carlos

New Member

Re: PBX Integration

A couple of hours after adding the "direct-inward-dial" config lines, I ran into another issue. What I am getting now is a busy/error tone after dialing the four digits. Every once in a while it does work fine and the calls goes through, but for the most part, it does not. I turned on "debug isdn standard" and I notice an "ERROR" on the debug.

.Sep 15 15:41:32.969: ISDN Se0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x5700

Bearer Capability i = 0x8090A2

Standard = CCITT

Transer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98397

Exclusive, Channel 23

Progress Ind i = 0x8183 - Origination address is non-ISDN

Calling Party Number i = 0x2180, '4805752099'

Plan:ISDN, Type:National

Called Party Number i = 0xA1, '8566'

Plan:ISDN, Type:National

.Sep 15 15:41:32.969: ISDN Se0/0:23 BACKHAUL: L3IF_rx_L2_pak: received data 0x08

0257000504038090A21803A983971E

.Sep 15 15:41:32.969: 0281836C0C2180343830353735323039

.Sep 15 15:41:32.969: 397005A138353636

.Sep 15 15:41:32.973: ISDN Se0/0:23 EVENT: process_rxstate: ces/callid 1/0x58 ca

lltype 2 CALL_INCOMING

.Sep 15 15:41:32.977: ISDN Se0/0:23 EVENTd: isdn_get_guid: Got Guid Se0/0:23

.Sep 15 15:41:32.977: ISDN Se0/0:23 EVENT: call_incoming: call_id 0x0058, Guid =

Se0/0:23

.Sep 15 15:41:32.977: ISDN

CC-VoIP-GW2# Se0/0:23 EVENTd: calltrkr_incoming_call: call_id=0x58

.Sep 15 15:41:32.977: ISDN Se0/0:23 EVENTd: calltrkr_setup_received: isdn_info=2

175790592l, call_id=0x58 ANSWER

.Sep 15 15:41:32.977: ISDN Se0/0:23 EVENTd: calltrkr_setup_received: calltracker

disabled

.Sep 15 15:41:32.977: ISDN Se0/0:23 EVENTd: calltrkr_setup_received: isdn_info=2

200470464l, call_id=0x58 ANSWER

.Sep 15 15:41:32.981: ISDN Se0/0:23 EVENTd: call_incoming: b channel 22, call ty

pe is VOICE ULAW

.Sep 15 15:41:32.981: ISDN Se0/0:23 EVENTd: call_incoming: Received a VOICE call

from 4805752099 on b channel 22 at 64 Kb/s

.Sep 15 15:41:32.981: ISDN CDAPI: cdapi_find_tsm found a GTD message IAM,

PRN,isdn*,,NI***,

USI,rate,c,s,c,1

USI,lay1,ulaw

TMR,00

CPN,04,,1,8566

CGN,04,,1,y,1,4805752099

CPC,09

FCI,,,,,,,y,

GCI,0694e307253611da8058000f90b91746

:

end of gtd length is 170

.Sep 15 15:41:32.985: ISDN Se0/0:23 **ERROR**: cdapi_process_connect_resp: cdapi

sez to reject the call (appl rejected?)

.Sep 15 15:41:32.985: ISDN Se0/0:23 EVENTd: calltrkr_call_cleared: isdn_info=0x8

32883C0, call_id=0x58

.Sep 15 15:41:32.985: ISDN Se0/0:23 EVENTd: calltrkr_call_cleared: isdn_info=0x8

1AFEE00, call_id=0x58

.Sep 15 15:41:32.989: ISDN EVENTd: cc_clear_free_list freed 0x83273214

.Sep 15 15:41:32.989: ISDN Se0/0:23 EVENT: process_rxstate: ces/callid 1/0x58 ca

lltype 2 CALL_CLEARED

.Sep 15 15:41:32.989: ISDN Se0/0:23 EVENTd: calltrkr_call_cleared: isdn_info=0x8

1AFEE00, call_id=0x58

.Sep 15 15:41:32.989: ISDN Se0/0:23 EVENTd: calltrkr_call_cleared: isdn_info=0x8

32883C0, call_id=0x58

.Sep 15 15:41:32.993: ISDN Se0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0

xD700

Cause i = 0x82AF - Resource unavailable, unspecified

Do you have an idea of what could be causing this now?

Thanks in advance

Re: PBX Integration

So we have to figure out what is telling the gateway:

Cause i = 0x82AF - Resource unavailable, unspecified

Are all the phones in the same CCM Region? That is, using the same codec?

And does the gateway have routes to all IP phones?

Finally, you mentioned that some call do succeed. Does a given extension, 8566 in this example, always fail, or intermittently succeed?

New Member

Re: PBX Integration

If you are referring to all IP Phones, then yes all phones are using the same codec. I have two gateways, one at the VoIP campus and the other integrating with the PBX System. They both have the same codecs configured on the "dial-peer voice 8500 voip" lines.

The gateway does have a route on the table to all IP Phones.

When I am unable to get through, no extensions will work. When I can, any extension will work. There is no particular extension that always works and that does not work. For the most part,it does work.

Is there anything that I could provide that would help in resolving this issue?

What is the gateway contacting when it receives the request?

Thanks again in advance.

Carlos

Re: PBX Integration

I have a few paths to for you to investigate, and am hoping others will add their experience...

However, if all calls from CCM to the PBX are working correctly during the failures, then it almost must be dial-plan related.

Issues that can cause "Cause i = 0x82AF - Resource unavailable, unspecified" include the following:

1) IP Phone is intermittently not reachable

2) Callmanager intermittently not reachable

3) Other gateway intermittently not reachable

These connectivity issues could be because of routing failure, frame loss, speed/duplex mismatch, etc. Verify that you can continuously ping from gateway to problem phone without loss.

And items 1-3 may not be reachability issues. Be sure that they are accepting these problem calls.

"debug h225 asn1", "debug h245 asn1" & "debug ccapi inout" are useful here.

You can extend your h225 setup timeout from 3 seconds to 6 seconds and see if it mitigates the issue.

4) Location setting in callmanager / CAC preventing the call

5) Codec mismatch / transcoder not available

6) Callmanager is hairpinning some calls back to the gateway, looping them, and filling up the T1. (bad route pattern / dial-plan)

I would suggest that you use Dialed Number Analyzer to verify that all calls, especially broken calls, are routing from this gateway the way you expect it to.

7) Local GW B-Channel is not available

debug isdn q931 will help here, in conjunction with "show isdn active" & "show isdn status"

8) Local GW D-Channel experiencing errors

9) One likely possibility is a DSP issue. The DSP in Local Gateway might not be working or is in use. See the following:

Troubleshooting DSPs

http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_note09186a00800e66b1.shtml

New Member

Re: PBX Integration

Thanks I will look into your recommended steps and post after.

Thanks again.

New Member

Re: PBX Integration

Hello again,

I have troubleshooted this issue more deeply and have discovered a couples of things.

Here are my scenarios:

1. Call is iniated from Nortel(PBX) side to VoIP side.

If VoIP phone user or voicemail answers, call is connected and active. If any of the parties hang up, the call/leg is disconnected normally.

If Nortel(PBX) phone is hanged up before VoIP phone user or voicemail answers, the call/leg remains open and connected, but not active.

At this point no other calls can be made from Nortel(PBX) side to VoIP because the Gateway has the leg open and connected.

There are two ways that I can clear the leg. By reloading the Gateway or by simply dialing from the VoIP side to the Nortel(PBX) side. This seems to clear the leg on the gateway and then I can again place calls from the Nortel end, but only until again someone calls and hangs up before the other party or the voicemail system picks up. Weird???

So I am wondering if the reason that gateway is reporting "Cause i = 0x82AF - Resource unavailable, unspecified" is because the leg is connected but not active???

Are there any specific troubleshooting steps that I can take to pinpoint why the legs are being kept connected??? The Nortel techs did verified that the connection was being close on their end.

I have debug the calls while they were failing and attached them to see if maybe they can help.

I also performed a ping test and if I ping from the gateway to the access switch where the phones are connected, I get a 100% success. I get an 89% success rate when I ping the phones.

Thanks again and I apologize for the delayed response.

Carlos

Silver

Re: PBX Integration

Turn on debug isdn q931 and call from the Nortel side over to ip phone - you should see your isdn setup message come in -then hang up from the Nortel side and see if you see a disconnect message come in. That is how the router side will know to clear the call, which it does not seem to be doing. Once the call is hung, the Nortel side thinks it cleared it, and keeps trying to send a call on a channel your side thinks is unavailable.

Mary Beth

New Member

Re: PBX Integration

Hello,

I debugged isdn q931 and observed the output. I do get a disconnect message, which tells me that the router is receiving it but releasing, here is the ouput:

CC-VoIP-GW2#

Nov 8 11:47:56.428: ISDN Se0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x4407

Bearer Capability i = 0x8090A2

Standard = CCITT

Transer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98397

Exclusive, Channel 23

Progress Ind i = 0x8183 - Origination address is non-ISDN

Calling Party Number i = 0x2180, '4805752068'

Plan:ISDN, Type:National

Called Party Number i = 0xA1, '8515'

Plan:ISDN, Type:National

Nov 8 11:47:56.456: ISDN Se0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0xC40

7

Channel ID i = 0xA98397

Exclusive, Channel 23

CC-VoIP-GW2#

Nov 8 11:47:56.692: ISDN Se0/0:23 Q931: TX -> ALERTING pd = 8 callref = 0xC407

Progress Ind i = 0x8188 - In-band info or appropriate now available

CC-VoIP-GW2#

Nov 8 11:48:02.504: ISDN Se0/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x44

07

Cause i = 0x8190 - Normal call clearing

Nov 8 11:48:02.508: ISDN Se0/0:23 Q931: TX -> RELEASE pd = 8 callref = 0xC407

Nov 8 11:48:02.540: ISDN Se0/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x

4407

CC-VoIP-GW2#sh call active voice compact

A/O FAX T Codec type Peer Address IP R:

Total call-legs: 1

7 ANS T0 None TELE P4805752068

CC-VoIP-GW2#sh call active voice brief

: hs. + pid:

dur hh:mm:ss tx:/ rx:/

IP : rtt:

delay://ms

media inactive detected: media cntrl rcvd: timestamp:

MODEMPASS buf:/ loss /

last s dur:/s

FR [int dlci cid] vad: dtmf: seq:

(payload size)

ATM [int vpi/vci cid] vad: dtmf: seq:

(payload size)

Tele (callID) [channel_id] tx://ms noise: acom:

> i/o:/ dBm

MODEMRELAY info:// xid:/ total://<

drops>

speeds(bps): local / remote /

Proxy :

anf>

bw: / codec:

tx:

tes>

rx:

es>

Telephony call-legs: 1

SIP call-legs: 0

H323 call-legs: 0

MGCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 1

11DE : 7 37527hs.1 +-1 pid:2000 Answer 4805752068 connected

dur 00:00:00 tx:0/0 rx:0/0

Tele 0/0:23 (7) [0/0.23] tx:0/0/0ms None noise:0 acom:0 i/0:0/0 dBm

Telephony call-legs: 1

SIP call-legs: 0

H323 call-legs: 0

MGCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 1

CC-VoIP-GW2#sh isdn status

Global ISDN Switchtype = primary-ni

ISDN Serial0/0:23 interface

******* Network side configuration *******

dsl 0, interface ISDN Switchtype = primary-ni

Layer 1 Status:

ACTIVE

Layer 2 Status:

TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED

Layer 3 Status:

0 Active Layer 3 Call(s)

Active dsl 0 CCBs = 0

The Free Channel Mask: 0x807FFFFF

Number of L2 Discards = 0, L2 Session ID = 1

Total Allocated ISDN CCBs = 0

Silver

Re: PBX Integration

That is really weird, you seem to get the disconnect and go through the whole process to hang up, but just don't. If you do a sh voice call sum, do you show the channel still in use? This may be a long shot, but sometimes having the wrong signalling type will cause really weird issues - you could discuss with the PBX people, or maybe just run through a few others, dms-100, etc, and see what happens. Might not work at all, but worth a try.

Mary Beth

Silver

Re: PBX Integration

Also, check that link for slips - that can cause weird troubles, and clocking on that platform is a little weird - and the attached PBX probably has a specific way they have to do it, so you will need to accomodate them.

Mary Beth

New Member

Re: PBX Integration

Mary Beth,

Here is the output of the "#sh voice call sum" command:

Thanks again.

CC-VoIP-GW2#sh voice call sum

PORT CODEC VAD VTSP STATE VPM STATE

============== ======== === ==================== ======================

0/0:23.1 - - -

0/0:23.2 - - -

0/0:23.3 - - -

0/0:23.4 - - -

0/0:23.5 - - -

0/0:23.6 - - -

0/0:23.7 - - -

0/0:23.8 - - -

0/0:23.9 - - -

0/0:23.10 - - -

0/0:23.11 - - -

0/0:23.12 - - -

0/0:23.13 - - -

0/0:23.14 - - -

0/0:23.15 - - -

0/0:23.16 - - -

0/0:23.17 - - -

0/0:23.18 - - -

0/0:23.19 - - -

0/0:23.20 - - -

0/0:23.21 - - -

0/0:23.22 - - -

0/0:23.23 None y S_WAIT_STATS S_TSP_WAIT_RELEASE

New Member

Re: PBX Integration

Thanks,

Adding direct-inward-dial to the pots config resolved my issue.

Hall of Fame Super Red

Re: PBX Integration

Hi Carlos,

I agree with Mary-Beth that you should perhaps try a different signalling type. Our CCM is connected to a Nortel 61c using DMS 100 and works great. When we first set it up we were trying ESS5 and NI2 and had alot of weird problems. Changing to DMS 100 solved them all.If you can, maybe you could also ask the PBX Techs to provide a print of the RDB (Route Data Block) for the PRI Route between the CCM and Nortel that you could post here.

HTH Rob

New Member

Re: PBX Integration

Rob,

Here is the Nortel RDB from the Nortel Switch to the Cisco Gateway.

REQ: prt

TYPE: rdb

CUST 0

ROUT 11

TYPE RDB

CUST 00

ROUT 11

DES CISCO

TKTP DID

NPID_TBL_NUM 0

SAT NO

RCLS EXT

DTRK YES

DGTP PRI

ISDN YES

MODE PRA

IFC NI2

CBCR NO

NCOS 0

SBN NO

PNI 00001

NCNA YES

NCRD NO

CHTY BCH

CPFXS YES

CPUB OFF

DAPC NO

BCOT 0

INTC NO

DSEL 3VCE

PTYP PRI

AUTO NO

DNIS NO

DCDR NO

ICOG IAO

RANX NO

SRCH LIN

TRMB YES

STEP

ACOD 8011

TCPP NO

PII NO

TARG 01

CLEN 1

BILN NO

OABS

INST

IDC NO

DCNO 0 *

NDNO 0

DEXT NO

ICIS YES

TIMR ICF 512

OGF 512

EOD 13952

NRD 10112

DDL 70

ODT 4096

RGV 640

FLH 510

GRD 896

SFB 3

NBS 2048

NBL 4096

TFD 0

DRNG NO

CDR NO

PAGE 002

MUS 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

OHQ NO

OHQT 00

TTBL 0

ATAN NO

PLEV 2

MCTS NO

ALRM NO

ART 0

SGRP 0

AACR NO

Hall of Fame Super Red

Re: PBX Integration

Hi Carlos,

Sorry for the delay in getting back to you, it's been one of those days. I have only had a quick look at the Nortel RDB and have noted that you are using NI2 interface type. In my original notes from way back I have **Do Not Use** by the NI2 Nortel settings.I would really try DMS100 if I were you.I will look at the RDB further later on,to see if there is anything else.

Thanks Rob

New Member

Re: PBX Integration

Hello,

I finally got the Nortel Tech to come down and troubleshoot this issue with me on site. The problem all along was that the Nortel switch was configured to use a Dial Pulse instead of a Dial Tone. Changing to Dial Tone resolved my weird problem.

I want to thank all for the help and guidance in troubleshooting this issue.

Thanks again,

Carlos

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