TiagoJC
asked on
Call Manager 6 integration with H323 Gateway and CUE
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
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
Last Comment
ASKER CERTIFIED SOLUTION
THIS SOLUTION IS ONLY AVAILABLE TO MEMBERS.
View this solution by signing up for a free trial.
Members can start a 7-Day free trial and enjoy unlimited access to the platform.
ASKER
I've requested that this question be deleted for the following reason:
sdfgsfg
sdfgsfg
"sdfgsfg" is not a reason to delete the question. A question was made and an answer given.
ASKER
I found my own answer
All,
Following an 'Objection' by willlywilburwonka (at https://www.experts-exchange.com/questions/27081819/10-Jun-11-05-Automated-Request-for-Review-Objection-to-Delete-Q-27050036.html) to the intended closure of this question, it has been reviewed by at least one Moderator and is being closed as recommended by the Expert.
At this point I am going to re-start the auto-close procedure.
Thank you,
SouthMod
Community Support Moderator
Following an 'Objection' by willlywilburwonka (at https://www.experts-exchange.com/questions/27081819/10-Jun-11-05-Automated-Request-for-Review-Objection-to-Delete-Q-27050036.html) to the intended closure of this question, it has been reviewed by at least one Moderator and is being closed as recommended by the Expert.
At this point I am going to re-start the auto-close procedure.
Thank you,
SouthMod
Community Support Moderator
Voice Over IP
Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN). Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).
9K
Questions
--
Followers
--
Top Experts
Get a personalized solution from industry experts
TRUSTED BY
ASKER