10-02-2014 03:15 AM - edited 03-17-2019 12:24 AM
Dears
i configure mobile voice access and i can hear the IVR when i call from a PSTN Line and i input my Remote destination number then it ask me for my PIN and i put it then it ask to press one for a call and after that i dialed the number but it give busy tone
here is my configuration
my MVA number is 7999
my IOS COnfiguration
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 3
rule 1 /^9\(\)/ /\1/
!
voice translation-rule 4
rule 4 /^22217/ /7/
rule 5 /^2217/ /7/
rule 6 /^022217/ /7/
rule 7 /^0122217/ /7/
!
voice translation-rule 5
rule 1 /^5/ /905/
rule 2 /^1/ /901/
rule 3 /^2/ /902/
rule 4 /^3/ /903/
rule 5 /^4/ /904/
rule 6 /^6/ /906/
rule 7 /^7/ /907/
rule 8 /^8/ /908/
rule 10 /^00/ /900/
rule 11 /'+'/ /900/
!
voice translation-rule 9
rule 1 /^5/ /05/
!
!
voice translation-profile MVA
translate calling 9
!
voice translation-profile OUT
translate called 3
!
voice translation-profile REDIAL
translate calling 5
!
voice translation-profile SIP-NEW
translate called 4
dial-peer voice 802 voip
description ** SIP TO STC **
translation-profile outgoing OUT
destination-pattern 9T
session protocol sipv2
session target ipv4:10.208.9.69:5060
session transport udp
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 1
dtmf-relay sip-notify rtp-nte sip-kpml
fax rate disable
no vad
dial-peer voice 802 voip
description ** SIP TO STC **
translation-profile outgoing OUT
destination-pattern 9T
session protocol sipv2
session target ipv4:10.208.9.69:5060
session transport udp
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 1
dtmf-relay sip-notify rtp-nte sip-kpml
fax rate disable
no vad
dial-peer voice 811 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 022217...$
voice-class codec 1
dtmf-relay sip-notify rtp-nte sip-kpml
fax rate disable
no vad
!
dial-peer voice 812 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 7999 voip
Description "MVA"
translation-profile incoming SIP-NEW
service mva
session protocol sipv2
incoming called-number 2217999
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 7991 voip
translation-profile incoming SIP-NEW
translation-profile outgoing OUT
destination-pattern 7999
session protocol sipv2
session target ipv4:192.168.200.53
voice-class codec 1
dtmf-relay rtp-nte
fax rate disable
no vad
in the service parameter
False | ||
Complete Match | ||
10
Thanks in Advanced |
Solved! Go to Solution.
10-02-2014 05:16 AM
Hi.
Did you check that CSS of associated DN on Destination Profile an redirect CSS allows you to call out?
Let us know.
Regards
Carlo
10-02-2014 03:19 AM
10-02-2014 03:29 AM
Can you do another test call and send us a "debug ccsip messages"
Please include the calling and the called number
10-02-2014 08:20 AM
10-02-2014 08:48 AM
The logs showing 404 not found..
Can you send your config please..
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.200.86:5060;branch=z9hG4bK256124F2
From: <sip:90565285270@192.168.200.86>;tag=6D99910-1AD7
To: <sip:7999@192.168.200.53>;tag=173206~244641b0-36ac-434c-91c1-823f25a68b28-24866165
Date: Thu, 02 Oct 2014 15:15:35 GMT
Call-ID: CAE4A53F-497D11E4-AF9A9583-B978D042@192.168.200.86
CSeq: 101 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0
10-03-2014 04:43 AM
10-03-2014 05:06 AM
Hi.
Are you able to call internal extensions such as Voicemail Pilot or collegues extension?
Let us know
Regards
Carlo
10-03-2014 07:06 AM
if you mean after i dialed the MVA then the answer is no
but if you mean from my local extensuoin o normal extension then yes
10-03-2014 07:12 AM
Hi.
This usually happens when:
-MVA directory number is not correctly configured on CUCM (Both service parameter and media resource)
-Dial Peer to MVA is not or not correctly configured
-Configured Incoming and redirect CSS on SIP / H323 trunk is unable to reach MVA Extension partition
Check it and let us know
Regards
Carlo
10-03-2014 01:13 PM
Dears
im actually reaching the MVA Number and i hear the IVR and everything going right till the moment it ask me to dial a number after i dial it get busy tone and my phisical extension keep flashing
if any of the GW configuration is wrong it wouldnt go to the MVA
can you tell me the steps to the right configuration so i can go through it
10-03-2014 02:41 PM
Make sure you have "Inbound Calling Search Space for Remote Destination" CallManager Service Parameter set to "Remote Destination Profile + Line Calling Search Space".
By default, it will use the trunk's CSS to try to reach the external number you try to dial which might not work in your environment.
10-04-2014 02:37 AM
Hi.
Brian is correct (+5) but to tell te truth... I have the default value for that parameter but CUCM keeps using RDP/Line CSS.
What I suggest is to check those CSS.
Can you also please send the output of a debug ccsip message while calling an internal extesion from MVA?
Thanks
Regards
Carlo
10-05-2014 09:40 AM
10-06-2014 01:39 AM
Hi.
Which version of CUCM are you running?
Let us know
Regards
Carlo
10-06-2014 03:33 AM
Hi
it's
System version: 8.6.2.20000-2
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