Mobile voice Access working but can't make outside calls

[aysar3000]

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

Enable Mobile Voice Access		False
Mobile Voice Access Number
Matching Caller ID with Remote Destination		Complete Match
Number of Digits for Caller ID Partial Match		10   Thanks in Advanced

[Carlo Poggiarelli]

Hi.

Go to Call Routing --> Class of control --> Access list

Add new

-Add a Name

-Select the user associated too your RDP as Owner

- Select "Allowed" 

- Click "Add member"

- As Filter mask  leave "Directory Number"

- As DN Mask put a  X!

Save and try again

HTH

Regards

Carlo

[aysar3000]

Hi Carlo

Still the same when i call from PSTN number i hear the whole authentication and press one to dial and then after i dialed i hear busy tone

but i find something that when i call from my remote destination which is my mobile number i cant hear anything no welcome message no authentication

[Carlo Poggiarelli]

Hi

Try to restart media streaming application service from cucm serviceability page and verify which locale is configured on RDP and which locale you added to Mobile Voice Access configuration.

Let us know

Regards

Carlo

[aysar3000]

Dears

i use same locale in both

[Carlo Poggiarelli]

Hi.

I don't know if you solved the problem but, in your sip trunk make sure you have "Redirecting Diversion Header Delivery Inbound" selected.

HTH

Regards

Carlo

[Ashish Bagla]

Hi,

I am too facing the same problem.

CUCM Version 10.5.2-10000-5 and SIP Gateway having normal E1 PRI termination.

MVA reaches to IVR and then when you try to reach a PSTN number it gives a fast busy tone and disconnects.

Any help would be appreciated.

Thanks in advance.

Regards,

Ashish Bagla

[aysar3000]

Check the Rerouting CSS

it should be able to reach the Partition of the PSTN Number you trying to call

it worked for me

[Carlo Poggiarelli]

Hi.

All leads to a call Routing/Permission issue.

To go in deep with this issue, we could setup a webex session.

Tell me if and when your are available and , if you agree, you can leave me your email address , even on a private message, where I can send you an invitation.

Let me know

Regards

Carlo

[aysar3000]

This would be very great out timming here is GMT +3

if it possible i can contact you in the next 2 days i will check which time i can go to the office coz it would be a holiday

and i wil contact you

my email is : infosec3000@gmail.com

Regards

Thanks in Advanced

[Carlo Poggiarelli]

Hi.

Ok let me know a possible date so I can organize my time ;)

Regards

Carlo

[aysar3000]

i change it to Remote Destination Profile + Line Calling Search Space"

still the same

[Brian Meade]

Can you attach the CallManager traces for a test call?

[Brian Meade]

You have to use H.323 gateway for MVA to work.  It doesn't work with SIP.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_0_1/ccmfeat/fsmobmgr.html#wp1125041

"Only H.323 VoIP gateways are supported for Mobile Voice Access."

Edit: Nevermind, it supports SIP in newer versions now:

"Both H.323 and SIP VoIP gateways are supported for Mobile Voice Access." from http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100_chapter_0110111.html#CUCM_TP_I193BE2E_00

Can you attach the CallManager traces for a test call?

[Carlo Poggiarelli]

Hi.

MVA works either with sip or H323 in the same way.

I have sip only VG at many customer's sites even in my head office.

did you add  "Media Resource" -->> Mobile Voice Access DN and secified language?.

In your sip trunk, did you configure redirect CSS?

Reset both Mobile Voce access and Media Streamin Application service

HTH

Regards

Carlo

[aysar3000]

Dears

yes i define the DN and the language

and i reset both link and configure the rerouting and still not working