CUBE Inbound call to Pilot

[SHQ]

Hello

I'm having issue configuring CUBE running IOS 15.2(4)M3 with an ITSP, incoming calls are landing on Pilot no.

The ITSP is sending INVITE with Username in the Request-URI, need to replace it with the To field.

dial-peer voice 100 voip (to ITSP)
destination-pattern .T
session protocol sipv2
session target ipv4:10.0.1.1
voice-class codec 1
dtmf-relay rtp-nte

dial-peer voice 101 voip (from ITSP)
session protocol sipv2
session target ipv4:10.0.1.1
incoming called-number .T
voice-class codec 1
dtmf-relay rtp-nte

dial-peer voice 102 voip (to CUCM)
destination-pattern 445566..
session protocol sipv2
session target ipv4:172.16.1.195
voice-class codec 1
dtmf-relay sip-notify

dial-peer voice 103 voip (from CUCM)
session protocol sipv2
session target ipv4:172.16.1.195
incoming called-number .T
voice-class codec 1
dtmf-relay sip-notify

Please let me know on which dial-peer to apply the sip-profile commands. I had tried some, but didnt work. 

Thanks

[Mohammed al Baqari]

Use the command 'debug voice dialpeer' to find which dialpeer is matched for inbound then apply the sip-profile to it. Because you are applying it on inbound dialpeer you need to enable this feature. 

voice service voip

 sip

 sip-profiles inbound 

[SHQ]

Hello

It's giving the below error, when i try to enable inbound. If its not supported, is there any other way to copy, for incoming call SIP To header to Request-URI for calls from ITSP to CUCM

SIP(config)#voice service voip

SIP(conf-voi-serv)#sip

SIP(conf-serv-sip)# sip-profiles inbound

                                                 ^

% Invalid input detected at '^' marker.

SIP(conf-serv-sip)# sip-profiles ?

  <1-10000>  The sip profiles tag number to be linked as global

SIP(conf-serv-sip)# sip-profiles 101 ?

<cr>

Thanks

[Mohammed al Baqari]

Ops. I am sorry I just saw that you are on IOS 15.2(4) which doesn't
support this feature. Its supported on Cisco IOS 15.4(2)T Cisco IOS XE
3.12S.

The easiest option is to upgrade (there are lot of new nice features in
newer versions for CUBE). You can do some work with corlists to get it
working but you have to create multiple duplicate dialpeers.

[SHQ]

Hello

UPDATE:

I have upgraded to 15.7(3)M 

Have  managed to apply the sip-profiles, incoming is working fine, but when external party disconnects the call, it doesnt disconnect on CUCM End. Attached debug

voice service voip
 ip address trusted list
 ipv4 172.16.1.195
 ipv4 10.0.1.46
 ipv4 10.0.1.1
 address-hiding 
 mode border-element license capacity 30
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
  sip
   rel1xx require "100rel"
   header-passing
   asserted-id pai
   midcall-signaling passthru
   privacy-policy passthru
   pass-thru content sdp
   sip-profiles inbound
   no call service stop

voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
 codec preference 3 g729r8

 codec preference 4 g729br8

voice class sip-profiles 1
 request INVITE sip-header P-Asserted-Identity remove
 request INVITE sip-header P-Preferred-Identity add "P-Preferred-Identity: <sip:44556600@10.0.1.1>"

voice class sip-profiles 2
request INVITE peer-header sip TO copy "sip:(.*)@" u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"

voice class sip-copylist 1
sip-header TO

dial-peer voice 100 voip
destination-pattern .T
session protocol sipv2
session target ipv4:10.0.1.1
voice-class codec 1
voice-class sip profiles 1
dtmf-relay rtp-nte

dial-peer voice 101 voip
session protocol sipv2
session target ipv4:10.0.1.1
incoming called-number .T
voice-class codec 1
voice class sip-copylist 1
dtmf-relay rtp-nte
!
dial-peer voice 102 voip
destination-pattern 445566..
session protocol sipv2
session target ipv4:172.16.1.195
voice class sip-profiles 2
voice-class codec 1
dtmf-relay sip-notify

dial-peer voice 103 voip
session protocol sipv2
session target ipv4:172.16.1.195
incoming called-number .T
voice-class codec 1
dtmf-relay sip-notify

Thanks

[SHQ]

Hello

Don't get it, the sip profile 2, as per this link https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/118825-technote-sip-00.html

should give me an output (invite towards CUCM), of request-uri  copied from the TO field (from ITSP invite), but TO field (in invite towards CUCM) is still Pilot number.

Require invite coming from ITSP with TO field to be copied to RURI and TO fields in the invite towards CUCM.

Please help.

[Mohammed al Baqari]

From debugs I see cube sent cancel to CUCM and CUCM acknowledge it. Is the call still connected after the external party drops the call.

[SHQ]

Yes, its still connected.

[SHQ]

Yes, its still connected, keeps ringing.

[Mohammed al Baqari]

In CUCM, in the SIP profile assigned to your sip trunk, set the option 'SIP Rel1XX Options' to 'Send PRACK for all 1xx messages'

[SHQ]

It's already enabled, for outgoing calls i get the PSTN messages:call wait,busy etc.

Shouldn't the Invite message sent towards CUCM contain TO field as

To: <sip:44556630@172.16.1.195> instead of To: <sip:44556600@172.16.1.195>

Thanks

[SHQ]

Hello

Modified the To successfully, but didnt help. Seems the Cancel RURI needs to be copied from To, is that correct?

Thanks

[SHQ]

Modyfing the CANCEL RURI resolved it, Thanks for your help.