[cisco-voip] DTMF Issue with SIP TRUNK (CUCM -- SIP Trunk---- CUCME<---CUE)
vignesh sethuraman
sethuvignesh at yahoo.co.in
Wed Jan 29 17:32:14 EST 2014
sure Brian. Please find attached, the debug output.
At the end you will find MWI notification message, those are received only when I hang up the phone and not when I press # key to send the message. Just for your clarification.
Thanks & Regards,
Vignesh S
On Wednesday, 29 January 2014 11:25 PM, Brian Meade (brmeade) <brmeade at cisco.com> wrote:
Can you collect a “debug ccsip messages” for one of the calls?
From:vignesh sethuraman [mailto:sethuvignesh at yahoo.co.in]
Sent: Wednesday, January 29, 2014 5:24 PM
To: Brian Meade (brmeade); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] DTMF Issue with SIP TRUNK (CUCM -- SIP Trunk---- CUCME<---CUE)
Hello Brian,
I tried that now and reset the SIP trunk but no luck.
My CUCM version is 8.6.2. and CUCME is running 15.2(1)T3 and my CUE is running 8.6.7.
Thanks & Regards,
Vignesh S
On Wednesday, 29 January 2014 11:17 PM, Brian Meade (brmeade) <brmeade at cisco.com> wrote:
Viki,
What do you see getting negotiated for DTMF on the CUCM side? You might want to try forcing out-of-band on the CUCM side. Try changing the DTMF Signaling Method to “OOB and RFC 2833”.
Brian
From:cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of vignesh sethuraman
Sent: Wednesday, January 29, 2014 5:07 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] DTMF Issue with SIP TRUNK (CUCM -- SIP Trunk---- CUCME<---CUE)
Hello All,
I am facing an issue with dtmf-relay. PhoneA registered to CUCM (SiteA) calls PhoneD registered to CUCME (Site C). Between Regions G729 codec is negotiated. PhoneD call-forward no answer to Voicemail. CUE integrated with CME. After leaving the Voicemail from PhoneA to PhoneD, when I press # key to send the Voicemail, it is not recognized.
Here is my scenario and the configuration.
(PhoneA) -- CUCM ----SIP TRUNK ----CUCME (PhoneD) ---> CUE.
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
registrar server expires max 600 min 60
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
telephony-service
sdspfarm units 3
sdspfarm transcode sessions 4
max-ephones 5
max-dn 10
ip source-address 3.3.3.3 port 2000
load 7945 term45.default.loads
time-zone 28
time-format 24
date-format dd-mm-yy
voicemail 3600
max-conferences 8 gain -6
call-forward pattern .T
moh "music-on-hold.au"
dn-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files
!
sccp local Loopback0
sccp ccm 10.0.10.160 identifier 2 version 7.0
sccp ccm 3.3.3.3 identifier 1 version 7.0
sccp ip precedence 3
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register BR2-IOS-XCODE
associate profile 2 register BR2-IOS-CFB
keepalive retries 5
switchover method immediate
switchback method immediate
switchback interval 15
!
dspfarm profile 1 transcode
codec g729r8
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 4
associate application SCCP
!
dspfarm profile 2 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
!
dial-peer voice 1000 voip
destination-pattern [15]...$
session protocol sipv2
session target ipv4:10.0.10.160
incoming called-number .
voice-class codec 1
dtmf-relay sip-notify
no vad
!
dial-peer voice 3600 voip
destination-pattern 3[16]00$
session protocol sipv2
session target ipv4:10.10.202.100
incoming called-number 399[89]....
dtmf-relay sip-notify
codec g711ulaw
no vad
!
On the CUCM, I did the following,
Media Termination Point Required (Checked)
MTP Preferred Originating CodecRequired Field: g711ulaw
DTMF Signaling MethodRequired Field: No preference
Non Secure SIP Trunk Profile: I am using TCP+UDP for INCOMING + Accept Unsolicited Notification (Checked).
Please let me know what I am missing.
Thanks,
Viki
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