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CUBE Integration to ITSP (IP Public)

solihul.hadi9
Level 1
Level 1

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?

 

17 Replies 17

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

 

 



Response Signature


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  


@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?



Response Signature