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Mobile voice Access working but can't make outside calls

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

Everyone's tags (1)
1 ACCEPTED SOLUTION

Accepted Solutions

Hi.Did you check that CSS of

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

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32 REPLIES
New Member

Sorry for the mistake in the

Sorry for the mistake in the picture

in the service parameter

 

Enable Mobile Voice Access  Required Field True
Mobile Voice Access Number 7999
Matching Caller ID with Remote Destination  Required Field Partial Match 
Number of Digits for Caller ID Partial Match  Required Field 7
VIP Super Bronze

Can you do another test call

Can you do another test call and send us a "debug ccsip messages"

Please include the calling and the called number

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Dearsthe RD Number is

Dears

the RD Number is 90565285270

i called the MVA 92217999

and then i dialed the number 90540533364

 

Thanks in Advanced

VIP Super Bronze

The logs showing 404 not

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

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Dearsi attached the SIP GW

Dears

i attached the SIP GW Configuration

Hi. Are you able to call

Hi. 

Are you able to call internal extensions such as Voicemail Pilot or collegues extension?

 

Let us know

 

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"
New Member

if you mean after i dialed

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

 

Hi.This usually happens when:

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

 

Please rate all helpful posts "The more you help the more you learn"
New Member

Dearsim actually reaching the

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

Make sure you have "Inbound

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.

Hi.Brian is correct (+5) but

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

Please rate all helpful posts "The more you help the more you learn"
New Member

Dearsi change it to Remote

Dears

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

but still nothing work

Hi.Which version of CUCM are

Hi.

Which version of CUCM are you running?

 

Let us know

 

 

Regards

 

Carlo

 

Please rate all helpful posts "The more you help the more you learn"
New Member

Hiit'sSystem version: 8.6.2

Hi

it's

System version: 8.6.2.20000-2

Hi.Go to Call Routing -->

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

 

Please rate all helpful posts "The more you help the more you learn"
New Member

Hi CarloStill the same when i

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

HiTry to restart media

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

Please rate all helpful posts "The more you help the more you learn"
New Member

Dearsi use same locale in

Dears

i use same locale in both

Hi.I don't know if you solved

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

Please rate all helpful posts "The more you help the more you learn"
New Member

Hi,

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

New Member

Check the Rerouting CSS

Check the Rerouting CSS

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

it worked for me

Hi.All leads to a call

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

Please rate all helpful posts "The more you help the more you learn"
New Member

This would be very great out

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

Hi.Ok let me know a possible

Hi.

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

 

Regards

 

Carlo

 

Please rate all helpful posts "The more you help the more you learn"
New Member

i change it to Remote

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

still the same

Can you attach the

Can you attach the CallManager traces for a test call?

You have to use H.323 gateway

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?

 

Hi.MVA works either with sip

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

Please rate all helpful posts "The more you help the more you learn"
New Member

Dearsyes i define the DN and

Dears

yes i define the DN and the language

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

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