DTMF Fails on outbound calls with SIP Trunk to PSTN

[jasunf]

I have a SIP gateway that works with both inbound/outbound calls.  However when I place an outbound call to an external source through the PSTN I can't pass DTMF.  In call manager the route pattern sends calls to the SIP gateway via a SIP trunk in call manager.  It matches a POTS dial-peer and sends the calls out to the PSTN fine.  In this config I can't pass DTMF.

I can configure h323 on the sip router and an h323 gateway in call manager and it works fine.  So I have a hybrid inbound SIP to call manager and an outbound h323 to PSTN type gateway. 

Most searches return DTMF issues with CUBE type routers that talk to an SBC.  However mine is SIP to PSTN directly using a 2811.  So I'm using a POTS dial peer to send calls out the router through the serial interfaces.

Is my hybrid setup the thing to do or will SIP from CUCM to PSTN send DTMF?

[glenn.martindell]

I'm not a DTMF relay expert but I've run into this before and this is how it was fixed.   You need to make sure you specify the correct DTMF relay in the dial peers.   The DTMF relay should be different for the SIP dial peers than they are for H323.

Your SIP dial peers should have:       dtmf-relay rtp-nte

Your H323 dial peers should have:     dtmf-relay h245-alphanumeric h245-signal

Thanks,

Glenn

[clileikis]

Hi there,

Is the MTP required selected on the sip trunk in CUCM?

HTH,

Chris

Sent from Cisco Technical Support iPhone App

[Jonathan Schulenberg]

Mixing H.323. and SIP depending on the call direction seems needlessly complicated to me.

Can you confirm the intended outbound call path is: SCCP Phone > CUCM > SIP trunk > Cisco ISR > ISDN PRI.

What model phone and firmware are you using? What version of CUCM and IOS? Not all phone/software versions support RFC2833 (i.e. rtp-nte).

Have you verified that the incoming call from CUCM is matching the SIP dial-peer? The 'show call active voice brief' command will show you what dial-peer was matched for the VoIP- and POTS-side call legs. Make sure you're not matching dial-peer 0!

[Chad Stachowicz]

Chris likely has nailed your problem.

Depending on your version of CUCM and IOS you will probably need to have MTP required to get DTMF to work.  12.4.20 is the first IOS train that could do it without MTP.

Chad

[jasunf]

Hey all thanks for the replies!

A bit more version history.  We are running call manager 7.1.31900-1.  Our phones are a mixture of 7940s and 7942s running P00308010100 and SCCP42.9-0-2SR1S.

The call path is indeed SCCP Phone> CUCM > Sip Trunk > Router> PRI for outbound calls.

The current test gateway is running code 12.4(25d)

Both my debug and show voice cal stat show it matching dial peer 1 which sends the call to the PRI:

dial-peer voice 1 pots

description OUTGOING TO PSTN PRI

destination-pattern 8T

port 1/0/0:23

I've tried with and without MTP and neither work.  The h323/sip combo isn't to bad to setup its still way less than MGCP alone   Just have to keep straight your CUCM configuration piece.  So far it's the only configuration that provides me DTMF outbound and the ability to block callers by ANI at the GW level.

[jleehawkins]

Can you send a list of all your dial-peers?

[jasunf]

No problem!  Pretty basic right now as I'm just testing.  I have one for inbound calls from PSTN for a TFN and one for outbound to the PSTN.

!

dial-peer voice 1 pots

description OUTGOING TO PSTN PRI

destination-pattern 8T

port 1/0/0:23

!

dial-peer voice 2 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 1002 voip

description INCOMING FROM PSTN TO CUCM SERVER

preference 1

destination-pattern 877.......

voice-class codec 1

session protocol sipv2

session target ipv4:10.16.241.52

dtmf-relay sip-notify cisco-rtp rtp-nte

no vad

[Chad Stachowicz · Accepted solution]

add a peer like this...

dial-peer voice 2000 voip

incoming called-number .

dtmf-relay cisco-rtp rtp-nte

!

if it doesnt work still please post confirmation that you actually HAVE MTP's in the MRG's the sip trunk is using, as well as the phones...

As well post a screenshot of the sip trunk config in CUCM, specifically around the DTMF setting at the bottom.

Chad

[jleehawkins]

Chad,

You beat me to it.

I'd also add your voice-class codec 1 and no vad to the inbound voip dial-peer unless you want to use the default g.729 with vad enabled.

You might also need to change the inbound voip dial-peers protocol to SIP.

Jason

[jasunf]

Thanks Chad that did it.  I've always been using MGCP gateways so still trying to get my head around these dial peers.  Is the explanation on why that 2000 dial peer fixing the issue easy?  To me if I matched an outbound dial peer POTS how does that VOIP dial peer resolve this DTMF issue?  This may help me more down the road as I push towards learning the SIP gateways.

Thanks Chad and everyone for your replies!  I appreciate it

[Chad Stachowicz]

I would also suggest running a more normal IOS train for Voice..  15.1(3)T1 is ok from what I have seen. 

Chad

[Chad Stachowicz]

ohh well think about the gateway as a house, and you to properly get in your need a front door and a back door.  So to walk through the house you have to go through 2 doors.  Pretend 1 door is in and one is out... no matter which way you enter...

I just gave you an entry voip peer from CUCM to the gateway, you had the one from the gateway to the PSTN...

Cisco lets it still work best effort, but you lose some features as you just saw

Chad

[jleehawkins]

You have to understand call legs.  Every call to a gateway has two call legs, an inbound and outbound.

If you don't have an inbound dial-peer to control your inbound call leg then the gateway will use the default dial-peer or dial-peer zero.  You can't see or configure dial-peer zero and it doesn't support dtmf-relay.

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#topic4

Also the inbound dial-peer is what negioates the parameters of a call such as codec, vad, and dtmf-relay when it receives a call.

[fedor.solovev]

Hello,

it is almost helped me:

I have jabber -> CUCM 11.5 -> SIP Trunk -> 2911 GW15.3 -> PSTN E1.
First, I was thinking that cisco-rtp helped me.

However, eventually, it was not work again, until I changed MTP at CUCM SIP Trunk Profile Insert MTP (best effort).

New Topic Here