01-23-2009 04:12 AM - edited 03-15-2019 03:43 PM
Guys,
I have a 2811 CME gateway connect to the Telco via SIP. I have no problem with inbound and outbound calls, they work fine but I have problem with DTMF. The Telco says they are not receiving any digits from our side. I have the below dial-peer config on my side, please let me know what I need to change...
dial-peer voice 9 voip
description <<< Outbound to Telco >>>
translation-profile outgoing outgoing_digits
destination-pattern 9.T
voice-class codec 1
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx
dtmf-relay sip-notify
no vad
!
AE-BAH-2811-01(config-dial-peer)#do sh ver
Cisco IOS Software, 2800 Software (C2800NM-SPSERVICESK9-M), Version 12.4(15)T8, RELEASE SOFTWARE (fc3)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2008 by Cisco Systems, Inc.
Compiled Mon 01-Dec-08 15:28 by prod_rel_team
ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)
AE-BAH-2811-01 uptime is 2 days, 19 hours, 19 minutes
System returned to ROM by Reload Command
System restarted at 19:53:18 AE Tue Jan 20 2009
System image file is "flash:c2800nm-spservicesk9-mz.124-15.T8.bin"
Cheers,
K
Solved! Go to Solution.
01-23-2009 06:37 AM
No DSP. You should use g.711 as I indicated before.
01-23-2009 06:43 AM
Hi Kiran,
Your voice class codec has g729 and g711 defined. The provider only advertises g711, so you are using g711 already for these calls.
If you think about this - without DSPs the router is not able to insert or change voice in the RTP packets. The SCCP IP phone is only going to send a Keypad message, and it doesn't send in-band information into the stream.
So, in order for in-band information to be inserted into the stream, you will need DSPs on the router to do this.
I would look into getting DSPs for this router, or more preferably, contacting your SIP provider and begging them to support RFC 2833 (rtp-nte).
If you have to transcode, it will be a pain because you will have to worry about your sessions, and troubleshooting DSPs if you ever have voice quality problems. RFC 2833 is the far more preferred option.
-nick
01-23-2009 06:56 AM
OK! We have just tested with rtp-nte on both ends and still doesn't work...does it mean I am sending OOB?
01-23-2009 07:06 AM
If SIP provider has changed, send the new messages log and we can confirm.
01-23-2009 07:47 AM
01-23-2009 07:56 AM
Provider isn't sending RFC 2833, capability 101:
v=0
o=2Connect-MSC4 0 0 IN IP4 22.22.222.2
s=sip call
c=IN IP4 80.88.246.2
t=0 0
m=audio 50976 RTP/AVP 0
This is ours:
m=audio 18738 RTP/AVP 0 8 18 101
c=IN IP4 11.11.111.11
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
101 is DTMF
hth,
nick
01-23-2009 08:35 AM
thanks nick & p.bevilacqua, I have sent the traces to the provider. I will update you guys when I hear from them.
Many thanks for your time.
01-27-2009 06:35 AM
Thanks Nick, I have provided the SIP traces to the Telco and they have finally accepted that we are sending DTMF in RFC 2833 and they are not able to respond to those. They are now in contact with there vendor (NexTone).
BR,
K
01-27-2009 06:51 AM
You can try putting this on your dial peer (hidden command):
voice-class sip dtmf-relay force rtp-nte
This will send it no matter what your SIP provider says.
hth,
nick
01-27-2009 08:36 AM
Hi Nick,
voice-class sip dtmf-relay force rtp-nte
The above command has solved my issue, dtmf is working fine... Great!!!
Thanks a million for your help!!!
Cheers,
Kiran
01-27-2009 09:53 AM
My rating to Nick for openly providing this great info.
01-23-2009 08:00 AM
Sorry, I could not see the rtpmap correctly. Definitely the cisco phones don't send dtmf in-band.
05-18-2013 10:40 PM
05-19-2013 12:41 AM
Capture:
deb ccsip messages
deb voip rtp session name-event (I think this is the correct sintax, I don't have a CLI close to me )
You should see the NSE events payload type 101 for the DTMF
--
Jorge Armijo
Please remember to rate helpful responses and identify helpful or correct answers.
05-19-2013 02:35 AM
I can call through sip trunk and voice is very good. But problem is that when I call any other call center trrough SIP trunk & she ask for dial extention number and I dial my desired extention but then the call is terminated. I observed that my extention dialing is not recevied by the CME. when I dial 110 it shows twice value like as:
//242092/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 1 to dstCallId 0x3B1AD
//242092/xxxxxxxxxxxx/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
//242092/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 1 to dstCallId 0x3B1AD
//242092/xxxxxxxxxxxx/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
//242092/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 1 to dstCallId 0x3B1AD
//242092/xxxxxxxxxxxx/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
//242092/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 1 to dstCallId 0x3B1AD
//242092/xxxxxxxxxxxx/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
//242092/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 0 to dstCallId 0x3B1AD
//242092/xxxxxxxxxxxx/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
//242092/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 0 to dstCallId 0x3B1AD
//242092/xxxxxxxxxxxx/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
//-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
I attach the following debuging log.
#deb voice ccapi in
#deb voice ccapi ino
#deb voice ccapi inout
Please assist me. I am in stuck about this CME configuration.
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