DTMF Issues

IPhone 5s iOS 8.0.2 does not recognize DTMF tones when dialing an automated service. I end up losing the call because automated service does not received a response from me.
Is there a setting adjustment that needs to me made and if so, where is this located?

Hello Manish
Thanks for your reply. So here is the thing. It was just as i feared, by leaving this "REQUIRE MTP" check on the SIP trunk, now all video calls are being setup as Audio only. So this is not a good solution.
It would be nice is the MTP is only invoked for the audio only callers, from my reading this is how its supposed to work.
I suppose as another workaround, not elegant but should work I could go to the H.323 Gateway and make a specific dial-peer for this one pattern that points to Conductor for audio participants and change this to be a SIP dial-peer then setup a SIP trunk from CUCM to the same voice gateway.
Seems a little strange, i was hoping i can make this work just as it is.
Has anyone else run into this issue?

Similar Messages

DTMF issues on SIP trunk to Verizon

Were you able to resolve this problem? I am having an identical issue also with Verizon.

Our topology and symptoms are as follows:
Outside phone -> PSTN -> Vzn SBC -> Vzn SIP trunk -> CUBE -> CUCM / VM system
DTMF tones generated by an IP phone are heard and recognized by an outside (off-net) phone/system as you would expect. However, DTMF tones generated by an outside (off-net) phone are not recognized by our voice mail system. When listening to the DTMF tone on an IP phone, it sounds very distorted and faint. A sniffer trace performed on the CUBE shows RFC 2833 NTEs being received from Verizon, and they appear to be properly relayed by the CUBE to the destination. Payload type negotiated for both legs is 101.
We are running CUCM 6.1.5. We have a CUBE router between CUCM and the Verizon SIP trunk. The CUBE router is running 12.4(24)T3 with the IPIPGW feature set. Our voice mail system is an AVST CallXpress system running v7.9 software. To CUCM the AVST voice mail ports appear as DNs assigned to several SCCP 7940 phones (DNs are part of a hunt group, hunt pilot = vm pilot). The AVST masquerades and registers as the 7940 phones.
I tried applying the "dtmf-interworking rtp-nte" both globally and at the dial-peer level with no success. Attached is the debug output you suggested.

DTMF Issue in SIP

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.

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

Outbound DTMF Issue

We are experiencing a problem with DTMF tones on external attendants. We have a CCM 4.2 cluster connected to a CUCM 8.6.1 via a QSIG ICT.
CCM 4.2 <--> ICT <--> CUCM 8.6 <--> CUBE <--> SIP
DTMF works OK from phones in the CUCM 8.6 cluster. All inbound DTMF works all the way through to the CCM 4.2 cluster. However, DTMF from phones in the CCM 4.2 cluster does not work. It was working OK last week when some changes (adding CUC integration) were made. I have tried reversing most of the changes, but cant get it working again.
Any ideas or TS steps recommended? Thanks!

Two things to add on this thread:
The statement "these are outbound calls - so they don't hit a VoIP peer." is not true. There is always an inbound and outbound dial-peer matched on IOS. If you do not have an inbound dial-peer to match, IOS will use dial-peer 0 which is almost always a bad thing. Make sure you have a VoIP dial-peer that will match for inbound calls from UCM to the router.
CCX does not support in-band DMTF such as RFC2833 with SIP. If you do not have OOB DTMF such as KPML (RFC4730) or H.245 alphanumeric, then you need to invoke an MTP for conversion.

SIP phone DTMF issue registered on a CUCME H323 gateway

I have a CUCME 10 gateway that is registered on a callmanager as H.323 gateway.
On this a Cisco 8831 SIP conference bridge that is not generating DTMF. No sound heard in the other end when pressing digits, its just muting the sound. Anyone got a tip?
Callflow is 8831->CME->CM->PSTN
voice register pool 1
busy-trigger-per-button 1
id mac xxx.xxxx.xxxx
type 8831
number 1 dn 1
dtmf-relay sip-kpml
dial-peer voice 1201 voip
destination-pattern ...T
progress_ind setup enable 3
delay transport-address
session target ipv4: callmanager ip address
incoming called-number .
voice-class codec 100
dtmf-relay rtp-nte <- Also tried dtmf-relay h245-alphanumeric

Can you try this..
On your voice register pool configure
dtmf-relay rtp-nte digit drop
And on your dial peer 1201
Dtmf-relay h245-alpha

SPA112 and Webex (DTMF issue)

When I try to join a Webex, and have the system call me on my VoIP line through my SPA112, it seems that the * key is not recognized. I can use the phone to interact with other menu systems, such as my bank. But for some reason, I can't get into a Webex without dialing manually into the conference.
I use Vitelity as my SIP provider. They support G.711u, G.711a, and G.729a codecs. They prefer RFC2833 for DTMF Tx method, but I've also had success in the past with inband.
I should also mention that I've had sporadic luck getting into Webex in the past with this same configuration using the SPA112 with G.729a and Inband (though that really shouldn't work consistently because of the compression).
Just as a troubleshooting step, I tried calling the VoIP line from another phone to make sure I could hear the * key. I could hear it, even when I was using ATV (RFC2833) which I didn't think was going to be the case. However, I don't seem to be able to control the length of the tone - though I have no idea whether the length is a problem or not.
At the moment, I'm trying to generate a log from the SPA112 that would include the DTMF tones. Can anyone help me with the right settings so that I can see these in my log? Thanks.

Just turn on syslog and debug to the highest level possible, send the messages to an external server and catch them all here (with either a syslogd server software or by a generic packet catcher like tcpdump or Wireshark). There is no more detailed log possible.
You can catch SIP packets and RTP stream (just the part with DTMF) as well.

DTMF issue, Alestra Mexico

Hi All, recently deployed an UC320, now I have problems because they do not receive DTMF tones in my AA, my SIP provider tells me that it is problem of UC320, exisite some PMF file that can solve it?
Thanks in advance

Hi Gabriel,
Please capture SIP trunk syslog and send it to me at [email protected]
Best regards,
Wendy

CUCM - Webex Callback DTMF issue

Hi!
Im having some trouble with webex conferences. When i use the "Callback" feature, i recieve the call but when it asks to accept it, i press the digits and nothing. its lik a DTMF-relay problem but only fails with the Webex callback feature.
Any Ideas??

Hi Juan,
Did you tried from another device?
You can also take that option off, in the admin side you can disable this feature...
see image...
Thanks

Fxo card is not connecting dtmf calls

I have an analogue line that is being set up from GSM modem and connected to fxo card on VG router .The fxo card is configured to go to an automated message with option of pressing digits to take one to the right operator.i.e press 1 for....,2 for......,etc. Unfortunately,this is not working when i callled the number.Its looking like a DTMF issue,but i ve adjusted all the seeting that has to do with dtmf on the fxo card......Has anyone encounter any familiar issue?

Check to see if you have the voice license bundle for the router.
show license
Index 3 Feature: uck9
Period left: Life time
License Type: Permanent
License State: Active, In Use
License Count: Non-Counted
License Priority: Medium
Without the voice license, no voice ports can be configured on 2900 series routers.

SPA112 Outbound DTMF Problem

I am having trouble getting the SPA112 to recognize DTMF tones from my phone in AVT mode. Using InBand mode DTMF works for the most part, but there are some IVRs that don't seem to like the audio DTMF either. Currently my phone plays DTMF tones for 80 ms, increasing that value to 120 ms resolves the issue that I am having, but I don't have time to make that change and I'm hoping to find an ATA configuration fix to my problem. I have tried several other ATAs, none of which seem to have any problem with the DTMF tones at 80 ms. Why is the SPA112 different, are there any other settings I can tweak to make this work?

Updating firmware to 1.3.1 or higher will fix DTMF issues,
I updated firmware and now I can dial tones even from wireless phones. Use auto mode
Solved!!

AutoAttendant not transferring calls in CUE 8.6 to CME 9.1

**Issue is calls are not transferring once extension or directory is pressed on the phone.
I upgraded our 2811 CME 7.1/CUE 7.0 router to 2911 15.2 CME 9.1/CUE8.6. It has CME and V/k9 license enabled. I uploaded the configuration used on the 2811 to the 2911. I added the ip addresses in the ip trusted list. Calls come in and out if I point the translation list to the phone and not to the auto attendant pilot number. Incase if it was toll fraud issue I disabled it but still a no go.
Everything worked fine on the 2811. I can't understand why it would not work on the 2911 15.2. Has anyone had any issues when upgrading from 12.4 to 15.x with auto attendant not transferring calls?
I ran a trace on the Cue when placed a call to extension 114:
20.10.10.1- CME interface address
20.10.10.5- CUE interface address
659 09/12 11:43:39.418 ACCN SIPS 0 Call.transferFailed(114, RESOURCE_NOT_ACKNOWLEDGING) SIPCallContact[id=33,type=Cisco SIP Call,implId=5738D3Dse-20-10-10-5# [email protected],active=true,state=CALL_ANSWERED,inbound=true,handled=false,locale=en_US

Ok I'm starting to think its a dtmf issue. I checked to see if I call from internal to the AA and dial an extension if it works but the same issue. I ran the debug voip ccapi inout and I see consume mask is not set. What would that indicate? Could that be the problem?
This is right when I dialed extension 114
2811-TEST#
001288: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001289: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
001290: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001291: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
001292: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001293: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
001294: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001295: Sep 16 14:59:23.198: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
001296: Sep 16 14:59:23.614: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 4 to dstCallId 0x38
2811-TEST#
001297: Sep 16 14:59:23.614: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
001298: Sep 16 14:59:23.618: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 4 to dstCallId 0x38
001299: Sep 16 14:59:23.618: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
2811-TEST#
At this point theres silence on the phone I see this message:
2811-TEST#
001300: Sep 16 14:59:28.914: //55/D62BA3D180C8/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=FALSE, Tone=Null,
   Tone Direction=Sum Network, Params=0x0, Call Id=55
001301: Sep 16 14:59:28.918: //56/D62BA3D180C8/CCAPI/cc_api_call_feature:
   Feature Type=50, Interface=0x49FC2B80, Call Id=56
2811-TEST#
2811-TEST#
At this point I get the message "the phone number you are trying to reach" then the call disconnects
2811-TEST#
001242: Sep 16 14:48:57.719: %VOICE_IEC-3-GW: SIP: Internal Error (ACK wait timeout): IEC=1.1.129.7.67.0 on callID 52 GUID=5D21AC143CE711E480BCEEC6B624839
Also I checked show voice iec description:
2811-TEST#show voice iec description 1.1.129.7.67.0
    IEC Version: 1
    Entity: 1 (Gateway)
    Category: 129 (Call setup timeout)
    Subsystem: 7 (SIP)
    Error: 67 (ACK wait timeout)
    Diagnostic Code: 0
2811-TEST#

MTP Insertion for IP Communicator

Hello,
Last time I was troubleshooting for soft phone call failure issue, where it gets a busy after a ring.
Call Flow
IP Communicator --> CUCM 9.1 --> SIP Trunk (early offer) --> SME 9.1 --> MGCP GW --> PBX
In the CCM traces I see soft phone was looking for an MTP resource, I added a MTP in MRGL and the issue got resolved. The SRND also suggest to add an MTP in MRGL for soft phone like legacy devices to make calls work with SIP early offer.
1. Could someone tell me what's the exact reason for the soft phone requires an MTP?
2. An MTP is in place, when there is an capabilities mismatch, most likely a DTMF issue. But considering we don't require DTMF and all for a basic voice call, can we make the call work without an MTP?
Thanks in advance for your answers.

Hello Sukh,
You need to be able to route to the voice network from the data network. What is the default gateway on your network, the UC or ASA? If it is the ASA, you will probably need to add a static route on that device to the voice network.
Also, you might need to define the TFTP server IP address in the IP Communicator application. Finally, pay close attention to the NIC you are using on the PC. The MAC of the NIC being used must be the MAC associated with the ephone. If you have multiple NICs, ie one for wired and one for wireless, you can set in IP Communicator which NIC to use.
Hope this helps. Let me know if you have additional questions regarding this.
Thanks,
-john

Wireshark capture rtp packets on Cisco CUBE.

Hello all,
We have this call flow and we are having intermittent DTMF issue
CUCM 10.5--->CUBE(10.1.1.10--->AVAYA(10.1.1.11)--->PSTN
I am trying to capture RTP packets between CUBE and AVAYA, How can we capture RTP packets between(10.1.1.10 and 10.1.1.11)??
I followed below steps and I can see the traffic only from AVAYA to CUBE and that too only SIP and TCP not RTP.
Router(config)# access-list 140 permit ip host 32.55.55.32 any
Router(config)# access-list 140 permit ip any host 32.55.55.32
This ACL will capture all traffic to and from this IP address.
Next we need to enable the Cisco packet monitoring service:
Router# monitor capture buffer holdpackets
Now we can filter the monitored traffic by filtering it through our access-list:
Router# monitor capture buffer holdpackets filter access-list 140
Now we need to name our particular packet capture. I have called mine "testcap"
Router# monitor capture point ip cef testcap all both
Router# monitor capture point associate testcap holdpackets
Now we can start our capture!
Router# monitor capture point start testcap
Once you think you have acquired enough packets, to stop the capture, type:
Router# monitor capture point stop testcap
Now you can export your data to your tftp server by typing in the following command. You can then open the .pcap file in Wireshark for viewing
Router# monitor capture buffer holdpackets export tftp://10.0.0.55/testcap.pcap
Once uploaded you can clear your capture buffer by typing the following:
Router# no monitor capture buffer holdpackets
Any help is much appreciated
Thanks!

But when i configure the destination as USB0 my pendrive, it fails.
Could be a bug but I wouldn't recommend configuring the destination as your USB drive because no one has the same luxury as you to have the USB sit there all the time.
Store to the flash and transfer to USB is probably the best solution.

Ringback Tone not heard in Arc Connect queue

Hi All,
Got a strange problem thats been ongoing for a while now.
Our customer has recently migrated to SIP and since this point I am having to force the external ringback tone using command disable-early media 180. In addition they have an Arc connect solution which now when transferring callers into the queues and then being passed to an agent, they do not hear ringback there is just silence. In queue messages and MOH is heard within the queue.
I'm pretty sure this is a DTMF issue but I have tried everything to rectify this with no luck.
We are unable to create a SIP trunk to the CUBE as CUCM 7.1.2 does not support this hence h323 gateway.
The flow is as follows :-
SIP > CUBE > h323 > CUCM > Arc
CUCM Version :- 7.1.2
Arc Version :- 5.1.0.403
I've attached sanitised config.
Any help would be much appreciated.
Cheers

You most likely right by saying this is DTMF related, I had a similar issue in the past, going H323<>SIP on a CUBE and played with the dtmf settings on the dial peers, and ended up with:
dtmf-relay h245-alphanumeric rtp-nte

Issue with DTMF for TBAT

We have noticed that something changed between ICM 7.5 and 8.5 and/or CVP 7.0 and 8.5. We have one or two situations where a call comes into the CVP Ingress Gateway, does the normal CUSP>CVP stuff, kicks off an ICM script and ICM immediately responds with a Label that evokes a DTMF *8 for Take-Back And Transfer. In ICM 7.5/CVP 7.0, this worked fine even without setting any VXML parameters or extending the RTP from the Ingress GW to the VXML GW. Ever since we migrated to ICM 8.5/CVP 8.5 and new IOS on the Ingress (15.0(1)M7) and VXML GWs (15.1(3)T1), the only way we can make the TBAT work is by playing a silence.wav file via a simple VXML app. It seems like the new versions moved the code needed for DTMF from the Ingress GW to the VXML GW, or something. I’ve tried searching Cisco.com for this issue but haven’t found anything yet.

Hi,
In BPS, you can use user specific variables or you can set up a Variable of type exit. You can also have a variable of type authorization which uses the security / authorization of the BW system.
Hope it helps...
Cheers,
Tanish

DTMF Issues

Similar Messages

Maybe you are looking for