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New Member

Cisco CCME ephone-hunt

Hi guys,

 

I have problem with ephone-hunt I want to finish call of hunt group on external mobile number,

but I can't put for final number external mobile number How I can solve this problem?

Configuration of hunt group:

 

ephone-hunt 1 sequential
 pilot 101
 list 130, 131, 113, 135, 121
 final ------>external mobile number
 timeout 10, 5, 5, 5, 5
 statistics collect

 

Tomislav

1 ACCEPTED SOLUTION

Accepted Solutions
Cisco Employee

You can try to modify SIP

You can try to modify SIP-header using voice class sip-profiles.

Another way is configuration on dial-peer:

clid network number <number>

 

Regards,

Kirill

 

23 REPLIES
Cisco Employee

Hello!You should try to use

Hello!

You should try to use voice hunt-group instead of ephone-hunt.

voice hunt-group 1 sequential

pilot 101

list 130, 131, 113, 135, 121

timeout 10

final <external_phone>

Regards,

Kirill

New Member

Hi Kirill, I tried but

Hi Kirill,

 

I tried but without success 

voice hunt-group 1 sequential
 final 8xxxxxxxxx
 list 130,131,113,135,121
 timeout 10
 pilot 101

 

T.

Cisco Employee

Tomislav,When u dial mobile

Tomislav,

When u dial mobile phone from your IP-phone, do u use any prefix like 0?

Can u post your config?

Can u try to configure test voice hunt-group with one number? That number should be mobile phone number.

 

Regards,

Kirill

New Member

Hi Kirill,When u dial mobile

Hi Kirill,

When u dial mobile phone from your IP-phone, do u use any prefix like 0?

No when we calling mobile numbers you are calling 8xxxxxxxxxx

Configuration what you suggested :

ephone-dn  41
 number 126
 call-forward all 8xxxxxxxxxxxx

voice hunt-group 1 sequential
 final 126
 list 130,131,113,135,121
 timeout 10
 pilot 101

dial-peer voice 10 voip
 description Outgoing
 translation-profile outgoing OUT
 destination-pattern 8T
 session protocol sipv2
 session target dns:sip.provider
 session transport udp
 voice-class codec 1  
 dtmf-relay rtp-nte
 no vad

voice translation-rule 8
 rule 1 /^1.*/ /0096113001/
!
voice translation-rule 9
 rule 1 // //

voice translation-profile OUT
 translate calling 8
 translate called 9

I tried to call local num 126 and call forwarding is working but doesn't work in hunt group

 

Thank you in advance!

BR

Tomislav

Cisco Employee

Hi!I was not my suggestion :

Hi!

I was not my suggestion :)

My suggestion was next(according to your configuration):

voice hunt-group 3 sequential
 final 126
 list 8xxxxxxxxxxxx
 timeout 10
 pilot 101

 

dial-peer voice 10 voip
 description Outgoing
 translation-profile outgoing OUT
 destination-pattern 8T
 session protocol sipv2
 session target dns:sip.provider
 session transport udp
 voice-class codec 1  
 dtmf-relay rtp-nte
 no vad

voice translation-rule 8
 rule 1 /^1.*/ /0096113001/
!
voice translation-rule 9
 rule 1 // //

voice translation-profile OUT
 translate calling 8
 translate called 9

 

Regards,

Kirill

New Member

Yes you had right sorry

Yes you had right sorry Kirill :( clipboard wasn't clean

 

voice hunt-group 1 sequential
 final 126
 list 130,131,113,135,121
 timeout 10
 pilot 101

ephone-dn  41
 number 126
 call-forward all 8xxxxxxx

I tried like this too :

voice hunt-group 1 sequential
 final 89645270210
 list 130,131,113,135,121
 timeout 10
 pilot 101

Cisco Employee

Can u put Mobile Phone number

Can u put Mobile Phone number into list, not into the final?

Regards,

Kirill

New Member

I put it :voice hunt-group 1

I put it :

voice hunt-group 1 sequential
 list 130,131,113,135,121,8xxxxxxxx
 timeout 10
 pilot 101

 

And nothing :(

Cisco Employee

What kind of PSTN connection

What kind of PSTN connection do u have? through SIP/E1/T1?

Can u turn 'debug isdn q931' or 'debug ccsip messages' on and look, if CME tries to make a call to Mobile Phone and there is any problem with the connection.

 

Regards,

Kirill

New Member

Hi KirillPlease check config

Hi Kirill

Please check config in attach.

I have SIP trunk to ISTP.

 

BR

Tomislav

Cisco Employee

Please,Post debug voice ccapi

Please,

Post debug voice ccapi inout and debugi csip messages, while testing case 2.

 

Regards,

Kirill

New Member

Hi Kirill,Please check the

Hi Kirill,

Please check the log.

Case 2

voice hunt-group 1 sequential
 list 130,131,113,135,121,89645270xxx
 timeout 10
 pilot 101

Cisco Employee

Hello!

Hello!

I think, that hunt group works fine.

You get 

225839: *Jul  3 15:49:42.387: //287721/D6A67B0BB80F/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.34.1.2:5060;branch=z9hG4bK4563622A0
From: "9262999777" <sip:89262999777@sip.provider>;tag=49764274-655
To: <sip:89645270210@sip.provider>;tag=aprqngfrt-ddldl220000a6
Call-ID: F4778862-1DE11E4-B81AC96C-3BA39579@10.34.1.2
CSeq: 101 INVITE
Timestamp: 1404388182

from your sip provider...

Please, post debug ccsip messages for simple call to that mobile phone number from any IP-phone.

In my opinion, you need to modify calling number from

9262999777

to 

0096113001

Or ask your SIP-provider to modify...

Regards,

Kirill

New Member

Hi Kirill Please check this

Hi Kirill

 

Please check this file

 

BR

Tomislav

Cisco Employee

Hello!

Hello!

U have different Invite messages.

Good call:

 

From: "9262483188"<sip:89262483188@sip.wiland.ru;user=phone>

Bad call:

From: "9262999777"<sip:89262999777@sip.provider;user=phone>

What are this numbers?

U call from outside...try to call from IP-phone to your hunt-pilot...it should work..

 

Regards,

Kirill

 

New Member

Hi Kirill,This second file is

Hi Kirill,

This second file is ok in the first i did changes

and 483188 replace 999777 I wanted to make more secure but I saw how you

didn't understand and share this original from GW.

​For me is interesting if calls from outside can be redirected to external mobile because this is the case when one operator is alone at night or for a weekend.

I will try to test and call from inside this pilot number but do you know how I can solve my problem.

 

Thanks in advance!

Tomislav

Cisco Employee

You can try to modify SIP

You can try to modify SIP-header using voice class sip-profiles.

Another way is configuration on dial-peer:

clid network number <number>

 

Regards,

Kirill

 

New Member

Hi Kirill,I didn't understand

Hi Kirill,

I didn't understand totally what is the problem, please can you explain me little bit.

I changed phone numbers manually in the log after capturing just for security and after that provided you with orginal.

 

Tomislav

New Member

Hi All, Kirill thank you for

Hi All,

 

Kirill thank you for your help, After more debugging I found what is the problem :)

I needed to change two things in sip INVITE msg Diversion and From

Please find configuration:

voice class sip-profiles 1
 request INVITE sip-header Diversion modify "<sip:(.*)@(.*>)" "<sip:sip:0123456@10.3.1.1>"
 request INVITE sip-header From modify "<sip:(.*)@(.*>)" "<sip:0123456@sip.wiland.ru>"

Can you post your telephony

Can you post your telephony-service config? You may need to add the following:

transfer-pattern .T
transfer-system full-blind

Regards,

Yosh

HTH Regards, Yosh
New Member

Hi Yosh,Please look at

Hi Yosh,

Please look at:

telephony-service
 sdspfarm units 2
 sdspfarm tag 1 cfb001f6cd3c5a0
 sdspfarm tag 2 xcode9927
 conference hardware
 conference transfer-pattern
 max-ephones 150
 max-dn 450
 ip source-address 172.16.11.1 port 2000
 max-redirect 13
 service local-directory authenticate
 cnf-file location flash:
 cnf-file perphone
 user-locale RU
 network-locale RU
 network-locale 1 RU
 network-locale 2 RU
 network-locale 3 RU
 network-locale 4 RU
 load 7914 S00105000400
 load 7911 SCCP11.9-0-3S
 load 7960-7940 P00308010200
 load 7941 SCCP41.8-5-3S
 load 7961 SCCP41.8-5-3S
 time-zone 32
 time-format 24
 date-format dd-mm-yy
 max-conferences 6 gain 3
 call-forward pattern .T
 moh "music-on-hold.au"
 dn-webedit
 time-webedit
 transfer-system full-consult dss
 transfer-pattern .T
 create cnf-files version-stamp Jan 01 2002 00:00:00

BR

Tomislav

New Member

Hi Yosh,I tried how you said,

Hi Yosh,

I tried how you said, but nothing look at conf below:

telephony-service
 sdspfarm units 2
 sdspfarm tag 1 cfb001f6cd3c5a0
 sdspfarm tag 2 xcode9927
 conference hardware
 conference transfer-pattern
 max-ephones 150
 max-dn 450
 ip source-address 172.16.11.1 port 2000
 max-redirect 13
 service local-directory authenticate
 cnf-file location flash:
 cnf-file perphone
 user-locale RU
 network-locale RU
 network-locale 1 RU
 network-locale 2 RU
 network-locale 3 RU
 network-locale 4 RU
 load 7914 S00105000400
 load 7911 SCCP11.9-0-3S
 load 7960-7940 P00308010200
 load 7941 SCCP41.8-5-3S
 load 7961 SCCP41.8-5-3S
 time-zone 32
 time-format 24
 date-format dd-mm-yy
 max-conferences 6 gain 3
 call-forward pattern .T
 moh "music-on-hold.au"
 dn-webedit
 time-webedit
 transfer-system full-blind
 transfer-pattern .T
 create cnf-files version-stamp Jan 01 2002 00:00:00

 

Case1

voice hunt-group 1 sequential
 final 8xxxxxxxx
 list 130,131,113,135,121
 timeout 10
 pilot 101


Case 2
voice hunt-group 1 sequential
 list 130,131,113,135,121,8xxxxxxxx
 timeout 10
 pilot 101


Case3
voice hunt-group 1 sequential
 final 126
 list 130,131,113,135,121
 timeout 10
 pilot 101

ephone-dn  41  dual-line
 number 126
 call-forward all 8xxxxxxxxxx

 

I would verify your dialing

I would verify your dialing prefix like Krill mentioned (+5) and also you can consider having a place holder DN that is set to forward all calls to the mobile number.

ephone-dn 99
number "dummy number"
call-forward all "prefix and mobile number"

voice hunt-group 1 sequential
 final "dummy number"
 list 130,131,113,135,121
 timeout 10
 pilot 101

Regards,

Yosh

HTH Regards, Yosh
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