Why Experts Exchange?

Experts Exchange always has the answer, or at the least points me in the correct direction! It is like having another employee that is extremely experienced.

Jim Murphy
Programmer at Smart IT Solutions

When asked, what has been your best career decision?

Deciding to stick with EE.

Mohamed Asif
Technical Department Head

Being involved with EE helped me to grow personally and professionally.

Carl Webster
CTP, Sr Infrastructure Consultant
Ask ANY Question

Connect with Certified Experts to gain insight and support on specific technology challenges including:

Troubleshooting
Research
Professional Opinions
Ask a Question
Did You Know?

We've partnered with two important charities to provide clean water and computer science education to those who need it most. READ MORE

troubleshooting Question

Call Manager 6 integration with H323 Gateway and CUE

Avatar of TiagoJC
TiagoJC asked on
Voice Over IPIP Telephony
6 Comments1 Solution1200 ViewsLast Modified:
Hi,

My H323 GW and users have registered successfully, but when users from Sydeny try to reach this remote site they can't be diverted to the users Voicemail (on CUE). the call seems to do the transfer to the Pilot number, but then i just get a busy tone and disconnected.

Calls via the PSTN and local acces to Voicemail work fine. Are there limitations with G729 WAN calls coming in and being transfered to the VM dial-peer which needs to be G711?

I've made sure the CTI route points, ports and Pilot number configured in CM have the correct CSS and the correct Voicemail partitions are configured for Sydney Devices
 
Hope someone can shed some light on the matter.
 
Cheers


Here's the relevant config on the H323 gateway.

voice call disc-pi-off
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol cisco
 h323
  call preserve limit-media-detection
 modem passthrough nse codec g711ulaw
!
!
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
 codec preference 3 g729r8
!
!
voice translation-rule 1
 rule 1 /0200/ /9201/
 rule 2 /^02/ /28/
!
voice translation-rule 2
 rule 1 /^6628/ //
!
voice translation-rule 66
 rule 1 /\([61].........\)/ /00061\1/
 rule 2 /\([1-9].*\)/ /0\1/
!
!
voice translation-profile SRST
 translate calling 66
 translate called 1
!
voice translation-profile SRST-VM
 translate calling 2
!
voice translation-profile calling
 translate calling 66
 translate called 1
!
!
voice class h323 1
  h225 timeout tcp establish 3
  call preserve limit-media-detection
!
dial-peer voice 1 pots
 voice cut-through alert
 description Inbound DID via PSTN
 translation-profile incoming calling
 incoming called-number 662802..
 direct-inward-dial
 port 0/0/0:15
 forward-digits all
!
dial-peer voice 3 pots
 description Outbound to PSTN
 destination-pattern 0T
 port 0/0/0:15
!
dial-peer voice 10 voip
 description Inbound CM-Subscriber Peer
 translation-profile outgoing calling
 destination-pattern ^28..
 voice-class h323 1
 max-redirects 10
 session target ipv4:10.1.201.11
 dtmf-relay h245-alphanumeric
 no vad
!
dial-peer voice 11 voip
 description Inbound CM-Publisher Peer
 translation-profile outgoing calling
 preference 2
 destination-pattern ^28..
 voice-class h323 1
 max-redirects 10
 session target ipv4:10.1.201.10
 dtmf-relay h245-alphanumeric
 no vad
!
dial-peer voice 66 voip
 description Inbound SRST Peer
 translation-profile outgoing SRST
 preference 3
 destination-pattern 662802..
 voice-class h323 1
 max-redirects 10
 session target ipv4:10.1.245.1
 dtmf-relay h245-alphanumeric
 no vad
!
dial-peer voice 67 voip
 description SRST CUE Voicemail
 translation-profile outgoing SRST-VM
 destination-pattern 9200
 session protocol sipv2
 session target ipv4:10.1.245.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
dial-peer voice 68 voip
 description SRST AA
 destination-pattern 9201
 session protocol sipv2
 session target ipv4:10.1.245.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
dial-peer voice 69 voip
 description SRST GMS
 destination-pattern 9202
 session protocol sipv2
 session target ipv4:10.1.245.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
dial-peer voice 70 voip
 description SRST MainNumber
 destination-pattern 2800
 session protocol sipv2
 session target ipv4:10.1.245.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
!
sip-ua
 mwi-server ipv4:10.1.245.2 expires 3600 port 5060 transport udp unsolicited
!
!
call-manager-fallback
 secondary-dialtone 0
 max-conferences 8 gain -6
 transfer-system full-consult
 ip source-address 10.1.245.1 port 2000
 max-ephones 10
 max-dn 20 preference 3
 system message primary SRST Mode: Phones in Fallback
 dialplan-pattern 1 662802.. extension-length 4 no-reg
 voicemail 9200
 no huntstop
 call-forward busy 9200
 call-forward noan 9200 timeout 10
 time-zone 39
 time-format 24
 date-format dd-mm-yy
ASKER CERTIFIED SOLUTION
Avatar of José Méndez
José Méndez

Our community of experts have been thoroughly vetted for their expertise and industry experience.

Commented:
This problem has been solved!
Unlock 1 Answer and 6 Comments.
See Answers