Incoming call..when calling number starts with 9. only they get dial tone..

[mshamkumar]

Hi Experts,

                   This is some thing weird problem...whose area code starts with 9 and when they try to call FXO pots line after first ring they get dial tone then ends with fast busy signal...

When I tried with all possibilities I figured out that its hitting SRST dial peers:

dial-peer voice 8 pots
preference 8
destination-pattern 9T
incoming called-number .
port 0/2/3

when I removed or shutdown this dial-peers it works fine..

Check Below debug these captured when we have dial tone..

    Line 62:    Peer Info Type=DIALPEER_INFO_SPEECH

    Line 68:    Dial String=, Expanded String=, Calling Number=9018012250T

    Line 69:    Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

    Line 78:    Dial String=, Expanded String=, Calling Number=9018012250T

    Line 79:    Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

    Line 81:    Result=Success(0); Incoming Dial-peer=1 Is Matched

    Line 83:    Result=Success(0); Incoming Dial-peer=2 Is Matched

    Line 85:    Result=Success(0); Incoming Dial-peer=3 Is Matched

    Line 87:    Result=Success(0); Incoming Dial-peer=4 Is Matched

    Line 89:    Result=Success(0); Incoming Dial-peer=5 Is Matched

    Line 91:    Result=Success(0); Incoming Dial-peer=6 Is Matched

    Line 93:    Result=Success(0); Incoming Dial-peer=7 Is Matched

    Line 95:    Result=Success(0); Incoming Dial-peer=8 Is Matched

    Line 98:    Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=1

    Line 150:    Incoming Dial-peer=1, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=TRUE,

    Line 183:    Stop Tone On Digit=TRUE, Tone=Dial Tone,

    Line 205:    Fax Relay=TRUE, Dial Tone=TRUE, Digit Collect=TRUE, Overlap=FALSE, DID=FALSE

    Line 329:    Peer Info Type=DIALPEER_INFO_SPEECH

    Line 335:    Dial String=, Expanded String=, Calling Number=9TT

    Line 336:    Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

    Line 345:    Dial String=, Expanded String=, Calling Number=9TT

    Line 346:    Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

    Line 348:    Result=Success(0); Incoming Dial-peer=1 Is Matched

    Line 350:    Result=Success(0); Incoming Dial-peer=2 Is Matched

    Line 352:    Result=Success(0); Incoming Dial-peer=3 Is Matched

    Line 354:    Result=Success(0); Incoming Dial-peer=4 Is Matched

    Line 357: SCOTRT002#Result=Success(0); Incoming Dial-peer=5 Is Matched

    Line 359:    Result=Success(0); Incoming Dial-peer=6 Is Matched

    Line 361:    Result=Success(0); Incoming Dial-peer=7 Is Matched

    Line 363:    Result=Success(0); Incoming Dial-peer=8 Is Matched

[Ayodeji Okanlawon]

What is really the problem? I am not sure your description is really clear.  Are you saying that people cannot make outbound calls? Can you post a sh run here?. Also can you  post the result of "debug voip ccapi inout" when you place a test call    

Is this CCM? what PROTOCOL ARE U USING ...H323 OR MGCP?

[mshamkumar]

Ok let me explain you guys clearly....

We are using CUCM 8.0 gateway 2811 MGCP.All are POT lines FXO.

When a user calls our DID's no(Incomming call) first he hear ringing tone and then immediately he hears dial tone and ends with fast busy tone.

Note:(if he punch down nos call will be placed out using other available port)

(Outsider)calling no 9xx-xxx-xxxx-->called no 4xx-xxx-xxxx-->ring.Dial tone........fast busy tone.

Actually it should be:

Calling no 9xx-xxx-xxxx-->called no DID(4xx-xxx-xxxx)-->Auto Attendent(6xxxx)-->Dial by name or by ext or operator.

I hope you got my problem...i attaced sh run and debugs...Have a look and let me know any thing else you need...Thank you.

[lusandi]

Hello mshamkumar

I hope you are doing great

I would like to confirm if the MGCP have access to the route patterns on the CUCM

Because I think the issue you are having could be related to that.

If the gateway do have access to the route patterns would you mind trying to remove them

Regards

Luis Sandi

[mshamkumar]

How to check that route patters are access to router and how to remove that.

...and can you expalin me how its related to routepatterns please....

Thank you.

[lusandi]

Hello mshamkumar,

You can verify that on the configuration of the gateway on the CUCM web page, the fxo port will have an incomming CSS.

You need to go to: device --> gateway --> select the mgcp gw --> select the FXO port --> verify the calling search space in there.

after that go to call routing --> class of control --> CSS and verify the partitions that the CSS have.

after that go to call routing ---> route/hunt ---> route patterns.

In there do this:

Search route patterns where (change it to partition )   beging with   and select the partitions you find.

Most likely you will find a route pattern that start with 9.(and something)

if you do have this I will strongly recommend you to remove the access to the route patterns to the gateway, because this can cause toll fraud.

this will be because of this:

The call it is comming to the gateway with the caller id like 912-345-6789

so if the gateway have access to the route pattern that it is 9.! it will think it is a call to 123-456-789 and will wait untill the interdigit time out expire.

[Ayodeji Okanlawon]

FROM THE DEBUG YOU HAVE SENT, THERE IS NO CALLED NUMBER...

  cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=9018012250
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-lastrdn=
   cisco-rdntype=0
   cisco-rdnplan=0

The calling number is 9018.... but there i sno number called. The destination filed "dest=" is empty.

Is this the full debug? and is this the correct debug?

[mshamkumar]

Yes this is full and correct debug logs....destination no should be DID as per u but no called is only to connect the port mean to generate signal to reach thats it...so why ur looking for called no...look dial-peers,port etc...if u see clearly debugs you can find callCall Info(Calling Number=9018012250(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=(TON=Unknown, NPI=Unknown))

Dec 13 14:22:15.450 EST: //1861/20C366C08F52/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=TRUE, Tone=Dial Tone,

Initial Digit Timeout=-1000(ms), Inter Digit Timeout=-1000(ms)

Dec 13 14:22:30.454 EST: //1861/20C366C08F52/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null

Dec 13 14:22:30.458 EST: //1861/20C366C08F52/CCAPI/cc_api_call_disc_cause_update:
   Call Entry(Disconnect Cause=16)

Try to explain me about these logs please.Thank you.

[Ayodeji Okanlawon]

How can a call be routed by the gateway if there is no destination number?

What number is dialled? WHy is the dialled number not showing on the debug.

[mshamkumar]

I dont know why its not showing...i have couple of debugs logs which took for caller-id purpose if u wana look them you can...

[mshamkumar]

When i removed dial peers from 1-8 which are for SRST every thing works fine.....so can any one explan me why incoming call is hitting SRST dial-peers..in which i gave perferences too...

Actually it should hit mgcpapps dial peers only coz preference is 0...but its not..why?whats wrong in config and CUCM config...?

[Ayodeji Okanlawon · Accepted solution]

Ok. You need to try the following..

on dial-peer voice 1 -8, you need to remove the follwing command "incoming called number ."

Remove this on all of them.

Then add a seperate dial-peer like this. Dont assign any port to it ust add it like this.

dial-peer voice 99998 pots
incoming called-number .
direct-inward-dial

Try again!

[mshamkumar]

I am trying it...but doubt with out port nos will these work in SRST mode.

Let me try and get back to you.Thank you.

[mshamkumar]

Ok i tried it and i dont have any issue of dial tone..but my Q? is will these dial-peers works in fall-back mode or not?...

[Ayodeji Okanlawon]

Did it work or not? Yes it will work in SRST. Rate useful posts and try and be polite with your english