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

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.

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

CUBE - DTMF Interworking between SIP INFO and RFC2833

Hi all,
After some testing I noticed that when the cube receives dtmf via SIP INFO it doesn't translate them into rfc2833 on the other side. This happens when the incoming INVITE advertises both SIP INFO and RFC2833.
When the incoming INVITE advertises only SIP INFO then it does the translation to RFC2833 and sends the digits via RTP on the outgoing interface.
CUBE supports both RFC2833 and SIP INFO as it advertises both methods in it's sip messages. Is this the expected behaviour ?
IOS is 15.1(3)T1. CUBE is running on a 3945 router.
I have found bug CSCuh65102 which seems to be a match. Can somebody confirm this ?

Hi,
Please check the following bug
https://tools.cisco.com/bugsearch/bug/CSCtj93573/?reffering_site=dumpcr
CUBE not processing DTMF SIP INFO to RFC 2833 upstream to Network
You can try an upgrade to 15.1(3)T4 or higher and check if the issue is resolved.
HTH
Manish

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?

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.

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

DTMF not working between 2 CUCM SIP Trunks

Dears
We have configured 2 SIP trunks on 2 CUCM servers , all calls are working fine except the DTMF ?? any ideas what can enable the DTMF between the SIP Trunks??
Inter-Cluster Trunk (Non-Gatekeeper Controlled) On Both Sides and the Codec is G.711u
Best Regards

Ok, so as I understand following is the issue description.
There are 2 sites A & B. A has CUCM cluster for IPT users & Site B has CUCM Cluster for Contact Center users. These 2 clusters are connected using Non-GK controlled ICT.
Site A users when call Site B IVR, they hear the greeting but DTMF is not recognized hence they are unable to choose between options. Correct ?
1. What are the CUCM, IP IVR or UCCX/UCCE versions ?
2. Are the site B users able to choose options without any issues ?
3. When you said IVR, is it IPIVR/UCCX/CVP, Unity Connection, Unity, CUE or BACD on gateway ?
Please, always give full details of your setup & then ask the query, as it helps you only to get a quick & precise answers.
GP.
Pls rate helpful posts by clicking on stars below the post !!

Dtmf-relay sip-notify

Evening Cisco techs,
Having trouble enabling sip-notify
Using CME 3.3 via (C2691-IPVOICEK9-M), Version 12.4(25d), RELEASE SOFTWARE (fc1)
Have registered cisco phones set up in CME
SIP is enabled on router. So is H323
In conf t mode via voice service voip
Upon using dtmf-relay sip-notify I get the error of unrecognised command
If other information is required. Please let me know

Just go to your SIP dial-peers:
R01#conf t
R01#dial-peer voice XX voip    << XX is you SIP dial-peer
R01(config-dial-peer)#dtmf-relay ?
cisco-rtp          Cisco Proprietary RTP
h245-alphanumeric DTMF Relay via H245 Alphanumeric IE
h245-signal        DTMF Relay via H245 Signal IE
rtp-nte            RTP Named Telephone Event RFC 2833
sip-kpml           DTMF Relay via KPML over SIP SUBCRIBE/NOTIFY
sip-notify         DTMF Relay via SIP NOTIFY messages
dial-peer voice 555 voip
destination-pattern 55555
session protocol sipv2
session target ipv4:1.1.1.1
dtmf-relay sip-notify
-Terry

CME/CUE SIP Phones DTMF-Relay

Hi all,
Just looking for some clarification on this one. I'm seeing some conflicting advice about setting the DTMF-Relay on SIP Phones registered to CME with a CUE Module. I've read some documentation indicating that rtp-nte RFC2833 is the only dtmf-relay supported for SIP Phones registered to CME, however I've also read some documents indicating that sip-notify must be configured as the dtmf-relay on SIP phones when they are communicating to a CUE module. I'm assuming I'm going to need to configure an MTP on the CME, but just wondering what the official DTMF config should be under the voice register pool for SIP phones.
Thanks!

Hi logan
When doing lab with cme 7.0 and sip phones .sip phones are not recognizing the "sip-notify" dtmf-relay method .It can only recognize "rtp-nte" method and it does not matter weather you are using sip-notify or rtp-nte for a dial-peer pointing to cme .
i configured on cue
ccn subsystem sip
dtmf-relay sip-notify
end
on cme i configured a dial-peer pointing to cue
dial-peer v 3888 voip
destination-pattern 3888
session target ipv4:177.3.11.10
codec g711ulaw
no vad
session protocol sipv2
dtmf-relay sip-notiy
on my sip phones
voice register pool 1
dtmf-relay sip-notify ------> now in this case cue wont recognize dtmf tones
when i change this dtmf-relay method to rtp-nte it recognizes dtmf tones to when recording a message

Problem connecting two trunks to sip provider using same CUBE

We need to connect two SIP trunks from service provider to Cisco CUCM 7.1 using CUBE “Cisco 2821”, SP using the following configuration:
First SIP PSTN Link Configuration(In-Out DID/DOD 218 7700 – 218 7799)
Customer IP Address =   10.196.191.158/30
SP IP Address = 10.196.191.157/30
Protocol= SIP
SIP Port = 5060
Transport Protocol=UDP
Voice Codec= G711 A-Law
DTMF = IN-Band DTMF without RFC2833
Signaling IP address = 10.201.20.49
IP Address 10.201.20.10 (Media IP) must be visible from IP PABX
Second SIP PSTN Link Configuration( Inbound Only 920009999)
Customer IP Address =   10.196.192.94/30
SP IP Address = 10.196.192.93/30
Protocol= SIP
SIP Port = 5060
Transport Protocol=UDP
Voice Codec= G711 A-Law
DTMF = IN-Band DTMF without RFC2833
SIP server IP address = 10.201.20.49
IP Address 10.201.20.10 (Media IP) must be visible from IP PABX
When we tried to configure both links on the same CUBE we faced two problems:
-          Routing issue, as we can’t route traffic using single CUBE through two different interfaces to the same destination “ i.e we have to configure static route commend (ip route 10.201.20.49 255.255.255.255 10.196.191.157 & ip route 10.201.20.49 255.255.255.255 10.196.192.93), sip traffic coming from one link can’t be sure to send it back to the same link.
-          SIP media & signaling control binding issue, as CUBE support sip binding using one interface only “one IP Address”, if we not using binding commands on the CUBE we can’t receive any calls though any link.
We have two options:
SP to send both traffic on the same trunk link
Or
Have another CUBE for the second link.
Attached network diagram.
Any solution?????
Regards,
Ahmed Rizk

I didn't mean NAT CUCM, I meant the interface towards it. But since you're using a single interface then yes that is what you NAT. You have a lot going on in that config. Probably a lot more than you need. Like I said you should work on this in two legs. CUBE to ITSP, and then CUCM to CUBE. You're trying to make the whole thing work in one shot which is going to cause you some headaches.
Install XLite free version. In the account settings set your UserID to a generic 10 digit phone number, domain to something generic, then at the bottom set the Proxy Address to the IP of your CUBE. The media ports will be negotiated dynamically between the CUBE and the ITSP. Since you said you're not registering you will also need to give the ITSP YOUR peer IP (this is how they secure the trunk) which is whatever IP you're sourcing from when you leave your network (what you're NAT'ing the CUBE to).
For testing, reduce your config to something like this:
voice service voip
allow-connections sip to sip
allow-connections h323 to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
signaling forward none
sip
dial-peer voice 10 voip
description CUBE_TO_ITSP
session protocol sipv2
session target ipv4:SIGNALING IP PROVIDED BY ITSP
destination-pattern [2-9].........
codec g711ulaw
dtmf-relay rtp-nte sip-notify
no vad
dial-peer voice 20 voip
description ITSP_TO_CUBE
destination-pattern .
session protocol sipv2
session target ipv4:Eventually your CUCM IP...for now set it to your computers IP.
codec g711ulaw
dtmf-relay rtp-nte sip-notify
no vad
Use XLite to place a phone call from your PC (if you have a mic and speakers you can have audio if the call connects). This should come pretty close to getting your outward leg established. Once you get this part working you can add in more codecs and translation profiles if you want. Let me know what happens. Include any debug or packet cap results if you can.

RV042 & VOIP Issues

Hello everyone. I have a client that recently purchased some RV042 routers to establish VPN connections between their offices. The VPN works great and the only issue we have right now is with their IP phone system. The problem that is occuring is that the phones work fine, but they only get 1 way audio on the phones from the remote locations. Discussing the phone system with their phone support guy we found that ALG (Application Layer Gateway) is usually the culprit here. Here's my problem with this, there doesn't seem to be a setting in the RV042 router to shut off ALG. Am I missing something here? The current firmware on the main router is version 4.0.4.02-tm does anyone know if the latest firmware has the option of turning off ALG?
Thank you for any help you can provide. I appreciate it.

RV0xx does not support SIP ALG.
Firmware 4.1.1.01 fixed an issue with SIP. Perhaps you want to give it a try.

CMR SIP alias via Productvity tools

Recently added CMR and running into an issue with SIP Alias for video endpoints. When scheduled via the Webex portal, the SIP alias is correct ([email protected]). However when we schedule using Productivity Tools or via Jabber, it schedules the meeting fine but then assigns the video alias differently. Rather than users firstname.lastname, it assigns the meeting number as the video alias. So now its [email protected] I need to figure out how to create a consistent meeting invite.

Hi,
So your current situation is that you are using AnyConnect with Split-tunneling? So you can access the LAN through VPN and Internet through the users local Internet connection?
I wonder if the problem with Full-Tunnel + Internet traffic is due to problem with NATing the traffic from VPN Client pool to the "outside" interface IP?
I think that NAT could be done by the following configuration
object network VPN-POOL
subnet 192.168.0.200 255.255.255.248
nat (outside,outside) after-auto source dynamic VPN-POOL interface
EDIT: The "after-auto" should take the rule at the bottom of the NAT rules, but I'm still thinking should it interfere with the configurations even if it didnt have the parameter.
I've only done this in software 8.2 and below, where it was
global (outside) 1 x.x.x.x
nat (inside) 1 10.10.10.0 255.255.255.0
nat (outside) 1 192.168.0.200 255.255.255.248
But as I said I havent done this with the new software. I can probably test this tomorrow at work though.
- Jouni
EDIT2:
I'm not 100% sure but is the following NAT statement even needed at the current configuration?
nat (inside,outside) source static any any destination static NETWORK_OBJ_192.168.0.200_29 NETWORK_OBJ_192.168.0.200_29
EDIT3:
I also usually use a totally different network for the VPN-pool. Your current LAN seems to be 192.168.0.0/24 and the VPN pool is 192.168.0.200/29. Maybe the top NAT statements are causing problems with the VPN Client Internet traffic.
What I would try to do is change the VPN Client pool to some different network, do the NAT0 between LAN and POOL like you have in the current configuration (just with new VPN Pool) and use the PAT configuration mentioned earlier. Leave out the NAT statement in the "EDIT2" in this post.
Sorry this is alot of guessing from my part before I get to test it myself with a lab ASA.

DTMF Issue in SIP

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