DTMF Issue in SIP

[LAJITH POTHU VARAKKAL]

Hi All,

I have an issue here. The DTMF is not recognized by the Unity when user wants to do remote login to voicemail box by pressing *

Call Flow : T1 --> AS5400 --> SIP Trunk --> CUCM 9.1.2 --> SCCP --> CUC 9.1.2

Time : Nov 12 20:06:56.417 UTC

Calling Party Number i = 0x1183, '914466553077'
Called Party Number i = 0xA1, '2067677' - 99992067677

I can see in CCAPI, * being pressed and NOTIFY message is sent to CUCM, and I get 403 Forbidden as response.

The dial-peer configuration point to CUCM is below

dial-peer voice 4320 voip
 tone ringback alert-no-PI
 description --- PSTN  to XXX  9999.XXXXXXX ---
 preference 1
 destination-pattern 9999.......$
 no modem passthrough
 session protocol sipv2
 session target ipv4:XXXXX
 voice-class codec 1
 voice-class sip early-offer forced
 voice-class sip options-keepalive
 dtmf-relay sip-notify rtp-nte
 fax rate 7200
 ip qos dscp cs3 signaling
 no vad

Logs are attached. Please help me to find out the issue.

[Joseph Rivera-Santiago]

So that fixes the internal dialing out DTMF issue:

The Main AutoAttendant to the 2523170388 is still functioning

But changing that incoming called-number 9T , broke the original issue that we just fixed.

Calling 9102857722 no longer accepts DTMF inputs now haha

Just to double check...I have:

!
dial-peer voice 15 voip
 description Incoming Calls To CUCM Subscriber
 preference 1
 destination-pattern [2-9].........
 voice-class codec 9999
 session target ipv4:10.1.1.11
 incoming called-number 9T
 dtmf-relay h245-alphanumeric
 no vad
!

!
dial-peer voice 110 voip
 description LD SIP Calls
 translation-profile outgoing Outgoing_LD_SIP
 destination-pattern 91[2-9]..[2-9]......
 voice-class sip early-offer forced
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte digit-drop
 codec g711ulaw
 no vad
!

!
dial-peer voice 100 voip
 description Local SIP Calls
 translation-profile outgoing Outgoing_SIP
 destination-pattern 9[2-9].........
 voice-class codec 9999
 session protocol sipv2
 session target sip-server
 incoming called-number [29].........
 dtmf-relay rtp-nte digit-drop
 no vad
!

[Ayodeji Okanlawon]

You have what is called overlapping dial plans. You use 9 to dial out and you also have remote location with extensions/DDI beginning with 9..Can you tell me specifically what the DDI range is for the remote location? is it 910....... or are there more?

[Joseph Rivera-Santiago]

Ah that makes sense.  We do use 9 to dial out.

We have locations all over the country and are currently expanding even further West.

For example, the 9102857722 rings to our Wallace, NC location

281-XXX-XXXX for a few Texas stores

205-XXX-XXXX for a few Alabama stores

And so forth

[Ayodeji Okanlawon]

ok..We need to use a different approach to resolve this..We need to prefix calls coming from cucm so as to break up the overlapping issue..

do this..

go to cucm, search for the Route list you use for outbound calls, click on the route group associated with it.

Under called party xformation

under discard digits: use to none

prefix digit outgoing calls: add 141 as shown below

Then do this...

voice translation-rule 141
 rule 1 /^141/ //

!
voice translation-profile 141
  translate called 141

dial-peer voice 15 voip
incoming called-number 141T
translation-profile incoming 141

Then..do this

dial-peer voice 100 voip
incoming called number [29].........

So what this does is that all calls coming from cucm are prefixed with 141 and will  match dial-peer 15. We then strip this 141 from the dial-peer and send the call as it should be out to the PSTN..This way we break the overlapping issue with calls coming from PSTN with 9..

[Joseph Rivera-Santiago]

I will make this change shortly, however does this break my existing setup?  For example, will users still be dialing 9 then the # to make calls?

I definitely understand where this is going, just making sure it doesn't cause any downtime, or change the current procedure.

Thanks again for all this help.  Above and beyond.

[Ayodeji Okanlawon]

Joe,

It doesn't break anything. This change is transparent to the end users. They continue to dial as normal. The only thing to watch out for is if the display on the ip phone shows the prefixed 141..If this happens then please apply this command and you should be good

voice service voip

no supplementary-service h225-notify cid-update

[Joseph Rivera-Santiago]

I will make these changes around 12:15pm, most should be at lunch.  And I will report back.

Again, thanks for all the help.

[Joseph Rivera-Santiago]

That worked like a champ.  I didn't see the '141' appended to any phones internally or get masked when calling externally, so that looks good as well.

The original issue I posted looks to be fixed, however, I am wondering if you could help with 2 other "issues".

1.  I have cleaned up that config as best as I know how, however I am trying to further that clean up, especially now that we have a more correct config in place.

 -Do we need any other dial-peers besides Voice 15 and Voice 100?   Are the original Voice Translation Rules/Profiles valid anymore?

2.  We've never been able to dial internationally from our phone system, but I do see the existing dial peer below:

!
dial-peer voice 120 voip
 description International SIP Calls
 destination-pattern 9011T
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!

Is this an easy fix to get working?

[Ayodeji Okanlawon]

Glad to help Joe..

Yes you do need those dial-peers. They all do different things..

dial-peer 15 is for inbound from cucm and for outbound to cucm sub1 and 16 is for outbound to cucm sub2 incase sub1 fails..

The other outbound dial-peers are used for routing different numbers..

I am sure this should be easy to fix. what happens when you dial international call with this dial-peer in place? The previous config you sent to me doesn't seem to have this..

[Joseph Rivera-Santiago]

Well, if I am dialing it out correctly:

9-004-072-420-5837

But it simply immediately goes "Your call can't be completed as dialed.  Try again....etc"

That is a cell phone in Romania (coworker)

The current dial peer I am seeing is:

!
dial-peer voice 120 voip
 description International SIP Calls
 destination-pattern 9011T
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!

[Ayodeji Okanlawon]

You are not dialing correctly. The IDD for the is is 011, so you should be dialling

9-0114-072-420-5837

[Joseph Rivera-Santiago]

Well now I feel silly for not looking at that dial peer closer.  However, now we receive

"The numbers are not in service, please try again"

I do know they are active numbers (coworkers mom cell and land line in Romania)

[Ayodeji Okanlawon]

Have you tried dialling another? Any other international number

[Joseph Rivera-Santiago]

I have:

4165034259

4162588513

(289) 997-2241

(416) 998-8298

All Canadian numbers.  If I use 9011 - then the number it gives the above message.

If I simply try and dial normally ->  9-1-416-503-4259 , I get:

"This type of call requires the collection of a valid passcode to proceed.  Please enter your passcode, followed by the pound key"

Possibly something in CUCM?

[Ayodeji Okanlawon]

That looks like FAC in cucm. Even if you put in the FAC code, this wouldn't match any dial peer on the gateway