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Replies

DTMF Digits not be sent through SIP

kwaldecker
Level 1
Level 1

Hello all,

I have customer running Call Manager Express.  We just brought up the SIP trunk to the telco and all inbound and outbound calls work just fine.

However, once a call is connected the DTMF digits are not being sent.  So if they call a number that has a menu they cannot press any of the options in the menu.

Here is the Dial Peers i am using.

Am i missing something? is there something that needs to change with the carrier?

thanks for any help

Ken

*****

dial-peer voice 100 voip

description PSTN

translation-profile outgoing SIP-OUT

destination-pattern 9.T

progress_ind setup enable 3

session protocol sipv2

session target ipv4:208.53.0.5

incoming called-number 0297

dtmf-relay rtp-nte

codec g711ulaw

ip qos dscp cs5 media

ip qos dscp cs5 signaling

no vad

dial-peer voice 1000 voip

description **Incoming Call from SIP Trunk**

service session

destination-pattern 0297

session protocol sipv2

session target ipv4:192.168.130.253

incoming called-number 0297

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 200 voip

description PSTN to *****

translation-profile outgoing SIP-OUT

destination-pattern 9.T

progress_ind setup enable 3

session protocol sipv2

session target ipv4:208.53.0.5

incoming called-number 0787

dtmf-relay rtp-nte

codec g711ulaw

ip qos dscp cs5 signaling

no vad

!

dial-peer voice 2000 voip

description **Incoming Call from SIP Trunk**

service session

destination-pattern 0787

session protocol sipv2

session target ipv4:192.168.130.253

incoming called-number 0787

dtmf-relay rtp-nte

codec g711ulaw

no vad

11 Replies 11

Chad Stachowicz
Level 6
Level 6

Hello,

   Undoubtedly this is due to the lack of using an MTP for anything going over the SIP Trunk.  I don't have any relevent CME config next to me, however using and MTP and registering it to you CME is going to be imperative to get DTMF to work over a SIP Trunk.

I am confident if you get MTP working on you SIP Trunk calls, you will be aok.

HTH,


Chad

hi Chad,

thanks for the response.  aren't the MTP only needed when connecting to regular call manager?

I have done a few call manager expresses with SIP before and never had to create a MTP for the dtmf to work.

the dtmf tones work with their voicemail system (a non cisco system)  just not with the SIP

Is this something new in the ios?

thanks for any help

Ken

kwaldecker
Level 1
Level 1
Hey guys,
  I worked with the Telco and they are not seeing any digits when i press them.
Here is the message i get in the debug when i press a digit.  Have you guys seen this before?  it looks like its trying to start a conference call?
*Feb  3 16:46:38.409: //2023/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x7E9
*Feb  3 16:46:38.409: //2023/xxxxxxxxxxxx/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
*Feb  3 16:46:38.409: //2023/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x7E9
attached is the full debug
thanks for any insight!

Hi,

   Once again, you may not have needed an MTP before, however in my expereinces this can really only be one of 2 things.  dtmf-relay configuration, which i see is set to rtp-nte, are you sure its not sip-notify with this carrier?  worth checking.  If its not this, I highly suggest trying some MTP's you can even just use software session directly on the router without needing a PVDM.

Chad

hi Chad,

can you give me a config to use?

also, i did check with the telco.  they are using that form of relay and they watched the trunk and do not see the digits being sent at all,

on the cisco debug it looks like the system is trying a conference call.

*Feb  3 16:46:38.409: //2023/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x7E9
*Feb  3 16:46:38.409: //2023/xxxxxxxxxxxx/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
*Feb  3 16:46:38.409: //2023/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x7E9

thanks!

Ken

something like this should do

voice-card 1
dsp services dspfarm
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
!
sccp local GigabitEthernet0/0
sccp ccm 9.13.29.30 identifier 1 version 4.0 -- Use your own IP address
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register MTP123456

keepalive retries 1
keepalive timeout 10
switchover method immediate
switchback method immediate
!
dspfarm profile 1 mtp
description MTP123456

codec g711ulaw

codec g729r8
maximum sessions software 400
associate application SCCP
!
!
telephony-service
max-ephones 1
max-dn 1
ip source-address 9.13.29.30 port 2000 --> Use your own IP address
sdspfarm units 1
sdspfarm mtp sessions 10
sdspfarm tag 1 MTP1234356

Hi Chad,

I addeed the MTP but still no luck :/

there seems to be some progress, before i added the MTP's i added a command to the dial-peer to force the dtmf

and now if i call the TAC guy at cisco, the DTMF digits seem to work however if i call other numbers and try their menus it still does not work.

any other options to try?

thanks!

Any resolution to this yet?  I'm running into this problem using ATAs (SCCP) and already have MTP configured as well.

Thanks.

It ended up being an issue with the SIP Telco carrier.   they made some changes and all began to work.

which ata version are you using? is it the old 186/188 or the new slim looking 187 that just came out.

Thanks for the quick reply!

ATA 186 running v3.1.0. Do you know the change the Telco made?

I've tried upgrading it to the latest firmware, but for whatever reason I can't get it upgraded via TFTP.

they made a change to the way the call was being routed outside their system.  for some reason when they passed the SIP call from us to a normal PRI at their location the DTMF would not pass through.  So they change the calls to stay SIP out of their location.  once they did that, it all worked fine

here is a link from cisco that shows how to manually upgrade those things as well

http://www.cisco.com/en/US/docs/voice_ip_comm/cata/186_188/2_15_ms/english/administration/guide/sccp/sccpaapc.html

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