08-31-2022 08:00 PM
Hi,
i have a cube which will connect to ITSP with Public IP . Do I have to configure SIP-UA first?
because I debug ccsip message when making incoming call but debug doesn't enter CUBE (Term Monitor already)Topology :
CUCM-> SIPTrunk -> CUBE -> ITSP
I tried configuring SIP-UA with the username, password and realm provided by ITSP. But when I check show sip-ua register status , the result in the register column is "No"
is there something wrong with my config? how do I know if the cube sends a request to ITSP?
Solved! Go to Solution.
09-07-2022 02:42 AM - edited 09-07-2022 07:55 AM
Please create separate dial peers that are used for calls from your CM and from your service provider and on these use information in the VIA header to match them. Something along the line with this should work.
voice class e164-pattern-map 1
description E164 Pattern Map for called number to CUCM
e164 4...
!
!
voice class e164-pattern-map 2000
description E164 Pattern Map for called number to ITSP
e164 0T
!
voice class uri CUCM sip
host ipv4:IP ADD CUCM 1
host ipv4:IP ADD CUCM 2
! add as many line as you need, one for each CM that handles call processing
!
voice class uri PSTN sip
host ipv4:IP ADD ITSP
! add as many line as you need
!
voice class server-group 1
ipv4 IP ADD CUCM 1 preference 1
ipv4 IP ADD CUCM 2 preference 2
! add as many line as you need, one for each CM that is in the CMG use by the trunk for the SBC
description Inbound calls from ITSP to CUCM
huntstop 1 resp-code 404 to 404
!
voice class sip-options-keepalive 1
description Used for Server Group SIP OPTIONS PING
!
dial-peer voice 101 voip
description *** Inbound Dial-Peer From ITSP ***
no incoming called-number +6221303022xx
incoming uri via PSTN
!
dial-peer voice 10 voip
description *** Inbound Dial-Peer from CUCM ***
session protocol sipv2
incoming uri via CUCM
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte sip-kpml
no vad
!
dial-peer voice 12 voip
description *** Outbound Dial-Peer for inbound calls from ITSP to CUCM ***
no destination-pattern 4...
no session target ipv4:IP ADD CUCM
session server-group 1
destination e164-pattern-map 1
voice-class sip options-keepalive profile 1
dtmf-relay rtp-nte sip-kpml
no vad
!
dial-peer voice 102 voip
description *** Outbound Dial-Peer to ITSP ***
no destination-pattern 0.
destination e164-pattern-map 2000
voice-class sip options-keepalive
09-07-2022 11:52 PM
Hi Roger,
Many thanks for respond , but before i am apply your suggest config , in and outgoing with my VG running properlly , dan sound hear from both (calling and called)
then i just receieved information now , suddenly Sip-Ua not register again , now i tried to upgrade IOS from your recommendation
09-08-2022 12:31 AM
@solihul.hadi9 wrote:
Hi Roger,
Many thanks for respond , but before i am apply your suggest config , in and outgoing with my VG running properlly , dan sound hear from both (calling and called)
then i just receieved information now , suddenly Sip-Ua not register again , now i tried to upgrade IOS from your recommendation
Not sure if I really understood your reply. Would you mind to clarify?
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