New features has
been introduced in CUBE to support URI dialing
Enable URI Call Routing
By default CUBE will
lookup the user field in the Request-URI of the INVITE message which should be
made of digits. To support URI dialing, you need to enable domain-lookup call
routing in the CUBE.
Domain lookup should
be enabled globally or on
the incoming dial-peer. Domain lookup isn't
required on the outgoing dial-peer
voice service
voip
sip
call-route url
or
dial-peer
voice x voip
voice-class sip call-route url
Inbound Dial-Peers
Inbound dial-peers
can match based on SIP URI of the incoming INVITE message. The matching order
is:
- Top Most VIA header in INVITE message
- Request-URI header in INVITE message
- TO header in INVITE message
- FROM header in INVITE message
- Called Number (incoming called-number)
- Calling Number (answer-address)
- Calling Number (destination-pattern)
- Dialpeer 0
voice class
uri voice-class-uri-tag sip|tel
Specify a URI
field for the voice class:
host hostname-pattern
host ipv4:ipv4-address
host ipv6:ipv6-address
host dns:dns-address
pattern uri-pattern
user-id username-pattern
!
dial-peer
voice tag voip
session protocol sipv2
incoming uri { from | request | to | via} voice-class-uri-tag
You need to apply the command
'session protocol sipv2' in order to configure URI matching. Without this
command, URI matching commands won't be listed
In a
scenario where multiple SIP hops are involved in a call, there would be
multiple via headers involved, and the topmost via header of an incoming SIP
invite represents the last hop that forwarded the SIP request, and the
bottom-most via header would represent the originator of the SIP request. Match based on VIA header of
incoming INVITE message will match the last hop that forwarded the request
(neighboring SIP entity), which is the topmost via header. Other VIA headers
are ignored
INVITE
sip:123@1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.1:5093;branch=z9hG4bK-17716-1-0
Via:
SIP/2.0/TCP 10.10.14.20:5093;branch=z9hG4bK-28280-1-0
Outgoing Dial Peer
Similarly, outbound
dial-peers can match URIs of incoming INVITE
message ordered as follow:
- URI of an incoming INVITE message & Carrier-ID Target
- Called Number & Carrier-ID Target
- URI of an incoming INVITE message
- Called Number
voice class uri 2001 sip
host ipv4:10.2.1.1
!
dial-peer
voice 1 VoIP
destination uri 2001
carrier-id
target orange
!
dial-peer
voice 2 VoIP
destination-pattern
654321
carrier-id
target orange
!
dial-peer
voice 3 VoIP
destination
uri 2001
!
dial-peer
voice 4 voip
destination-pattern 654321
INVITE sip:654321@10.2.1.1
SIP/2.0
Via:
SIP/2.0/UDP 10.1.1.1:5060;x-route-tag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From:
"555" ;tag=1
Configuration
Example
voice class
uri InboundMatch sip
pattern hquser*
!
voice class
uri outgoingURI sip
host dns:hqcluster.local
!
dial-peer
voice 1913 voip
session protocol sipv2
session target ipv4:10.45.230.10
incoming uri request InboundMatch
voice-class sip call-route url
!
dial-peer
voice 1912 voip
session protocol sipv2
session target ipv4:10.170.10.200
destination uri outgoingURI
*********************************************************************
Received:
INVITE sip:hquser1@hqcluster.local
SIP/2.0
Via:
SIP/2.0/TCP 10.45.230.10:5060;branch=z9hG4bK2379042ad5
From:
"SBUSER1-CALLER"
;tag=28~8901f810-e3b7-43de-b8ef-31fe13e397a7-18206035
To:
Date: Mon, 22
Feb 2016 10:35:46 GMT
Call-ID:
6cfb280-6ca1e482-1a-ae62d0a@10.45.230.10
Supported:
timer,resource-priority,replaces
Min-SE: 1800
User-Agent:
Cisco-CUCM9.1
Allow: INVITE,
OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101
INVITE
Expires: 180
Allow-Events:
presence, kpml
Supported:
X-cisco-srtp-fallback,X-cisco-original-called
Call-Info:
;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid:
0114274944-0000065536-0000000026-0182856970
Session-Expires: 1800
P-Asserted-Identity:
"SBUSER1-CALLER"
Remote-Party-ID:
"SBUSER1-CALLER"
;party=calling;screen=yes;privacy=off
Contact:
Max-Forwards:
70
Content-Length:
0
Request URI Pass-Through
By default CUBE will
replace the host portion in Req-URI & TO headers of INVITE message with the
configured session target as follow:
dial-peer
voice 1913 voip
session protocol sipv2
session target ipv4:10.45.230.10
incoming uri request InboundMatch
voice-class sip call-route url
!
dial-peer
voice 1912 voip
session protocol sipv2
session target ipv4:10.170.10.200
destination uri outgoingURI
----->
Received:
INVITE sip:hquser1@hqcluster.local
SIP/2.0
Via:
SIP/2.0/TCP 10.45.230.10:5060;branch=z9hG4bK2379042ad5
From:
"SBUSER1-CALLER"
;tag=28~8901f810-e3b7-43de-b8ef-31fe13e397a7-18206035
To: <sip:hquser1@hqcluster.local>
Date: Mon, 22
Feb 2016 10:35:46 GMT
Call-ID:
6cfb280-6ca1e482-1a-ae62d0a@10.45.230.10
Supported:
timer,resource-priority,replaces
Min-SE: 1800
User-Agent:
Cisco-CUCM9.1
Allow: INVITE,
OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101
INVITE
Expires: 180
Allow-Events:
presence, kpml
Supported:
X-cisco-srtp-fallback,X-cisco-original-called
Call-Info:
;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid:
0114274944-0000065536-0000000026-0182856970
Session-Expires: 1800
P-Asserted-Identity:
"SBUSER1-CALLER"
Remote-Party-ID:
"SBUSER1-CALLER"
;party=calling;screen=yes;privacy=off
Contact:
Max-Forwards:
70
Content-Length:
0
<----- span="">
Sent:
INVITE sip:hquser1@10.170.10.200:5060
SIP/2.0
Via:
SIP/2.0/UDP 10.170.170.2:5060;branch=z9hG4bK44CE2352
Remote-Party-ID:
"SBUSER1-CALLER"
;party=calling;screen=yes;privacy=off
From:
"SBUSER1-CALLER" ;tag=970E872C-1947
To: <sip:hquser1@10.170.10.200>
Date: Mon, 22
Feb 2016 15:18:36 GMT
Call-ID:
61038AB0-D8AE11E5-AF45D397-811E4C2B@10.170.170.2
Supported:
timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid:
2310550400-0000065536-0000000046-0182856970
User-Agent:
Cisco-SIPGateway/IOS-15.4.3.M3
Allow: INVITE,
OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO,
REGISTER
CSeq: 101
INVITE
Timestamp:
1456154316
Contact:
Expires: 180
Allow-Events:
telephone-event
Max-Forwards:
69
Session-Expires: 1800
Content-Length:
0
In URI dialing this
isn't a desirable behavior as the preference is to maintain the domain-name for
the destination system to be able to perform domain-lookup. Typical example
when routing calls from CUCM to Lync or vice-versa.
You can pass-through
Req-URI and TO headers using the command 'voice-class
sip requri-passing' which needs to be applied globally or on the outgoing dial-peer
Note: Remote-Party-ID, From and
Contact headers will be modified to reflect the session target in the host
portion
dial-peer
voice 1913 voip
session protocol sipv2
session target ipv4:10.45.230.10
incoming uri request InboundMatch
voice-class sip call-route url
!
dial-peer
voice 1912 voip
session protocol sipv2
session target ipv4:10.170.10.200
destination uri outgoingURI
voice-class sip requri-passing
The debugs will look
as follow:
----->
Received:
INVITE sip:hquser1@hqcluster.local
SIP/2.0
Via:
SIP/2.0/TCP 10.45.230.10:5060;branch=z9hG4bK2379042ad5
From:
"SBUSER1-CALLER"
;tag=28~8901f810-e3b7-43de-b8ef-31fe13e397a7-18206035
To: <sip:hquser1@hqcluster.local>
Date: Mon, 22
Feb 2016 10:35:46 GMT
Call-ID:
6cfb280-6ca1e482-1a-ae62d0a@10.45.230.10
Supported:
timer,resource-priority,replaces
Min-SE: 1800
User-Agent:
Cisco-CUCM9.1
Allow: INVITE,
OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101
INVITE
Expires: 180
Allow-Events:
presence, kpml
Supported:
X-cisco-srtp-fallback,X-cisco-original-called
Call-Info:
;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid:
0114274944-0000065536-0000000026-0182856970
Session-Expires: 1800
P-Asserted-Identity:
"SBUSER1-CALLER"
Remote-Party-ID:
"SBUSER1-CALLER"
;party=calling;screen=yes;privacy=off
Contact:
Max-Forwards:
70
Content-Length:
0
<----- span="">
Sent:
INVITE sip:hquser1@hqcluster.local
SIP/2.0
Via:
SIP/2.0/UDP 10.170.170.2:5060;branch=z9hG4bK44CF3C
Remote-Party-ID:
"SBUSER1-CALLER"
;party=calling;screen=yes;privacy=off
From:
"SBUSER1-CALLER" ;tag=971004A4-14C8
To: <sip:hquser1@hqcluster.local>
Date: Mon, 22
Feb 2016 15:20:14 GMT
Call-ID:
9B388AED-D8AE11E5-AF4BD397-811E4C2B@10.170.170.2
Supported:
timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid:
3290550400-0000065536-0000000047-0182856970
User-Agent:
Cisco-SIPGateway/IOS-15.4.3.M3
Allow: INVITE,
OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO,
REGISTER
CSeq: 101
INVITE
Timestamp:
1456154414
Contact:
Expires: 180
Allow-Events:
telephone-event
Max-Forwards:
69
Session-Expires: 1800
Content-Length:
0
302 Contact Header Pass-Through
Similar to Req-URI,
you can pass-through contact header only in outgoing
302 messages. This is important in call forward scenarios in order to
maintain host portion of contact header instead of replacing it with session
target
voice service
voip
sip
contact-passing
or
dial-peer
voice tag voip
session protocol sipv2
voice-class sip contact-passing
Remote-Party-ID Normalization
The next step is to
fix the caller-ID display. Incoming INVITE
in below debug contains caller ID, caller URI and caller DN which is
basically blended addressing.
Received:
INVITE
sip:hquser1@hqcluster.local SIP/2.0
Via:
SIP/2.0/TCP 10.45.230.10:5060;branch=z9hG4bK2379042ad5
From:
"SBUSER1-CALLER"
;tag=28~8901f810-e3b7-43de-b8ef-31fe13e397a7-18206035
To:
Date: Mon, 22
Feb 2016 10:35:46 GMT
Call-ID:
6cfb280-6ca1e482-1a-ae62d0a@10.45.230.10
Supported:
timer,resource-priority,replaces
Min-SE: 1800
User-Agent:
Cisco-CUCM9.1
Allow: INVITE,
OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101
INVITE
Expires: 180
Allow-Events:
presence, kpml
Supported:
X-cisco-srtp-fallback,X-cisco-original-called
Call-Info:
;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid:
0114274944-0000065536-0000000026-0182856970
Session-Expires: 1800
P-Asserted-Identity:
"SBUSER1-CALLER"
Remote-Party-ID: "SBUSER1-CALLER"
;party=calling;screen=yes;privacy=off
Contact:
Max-Forwards:
70
Content-Length:
0
CUBE will override
Remote-Party-ID header to reflect the session target in the host portion. Also,
it will strip x-cisco-number as it is unsupported header. At the receiving
endpoint, the caller number won't be displayed and the caller URI will be
wrongly displayed.
Sent:
INVITE
sip:hquser1@hqcluster.local SIP/2.0
Via:
SIP/2.0/UDP 10.170.170.2:5060;branch=z9hG4bK44CF3C
Remote-Party-ID: "SBUSER1-CALLER"
;party=calling;screen=yes;privacy=off
From:
"SBUSER1-CALLER" ;tag=971004A4-14C8
To:
Date: Mon, 22
Feb 2016 15:20:14 GMT
Call-ID:
9B388AED-D8AE11E5-AF4BD397-811E4C2B@10.170.170.2
Supported:
timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid:
3290550400-0000065536-0000000047-0182856970
User-Agent:
Cisco-SIPGateway/IOS-15.4.3.M3
Allow: INVITE,
OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO,
REGISTER
CSeq: 101
INVITE
Timestamp:
1456154414
Contact:
Expires: 180
Allow-Events:
telephone-event
Max-Forwards:
69
Session-Expires: 1800
Content-Length:
0
This can be fixed by
applying normalization.
- Copy Remote-Party-ID header from incoming INVITE message to u01 variable
- Paste Remote-Party-ID header to outgoing INVITE message from u01 variable
!!! …
Define which headers are allowed to be copied from incoming SIP message
voice class
sip-copylist 1
sip-header Remote-Party-ID
!!! …
Create normalization rule to modify the required header. First copy the
peer-header (which is incoming header) to variable then paste it as outgoing
header
voice class
sip-profiles 10
request INVITE peer-header sip Remote-Party-ID
copy "(.*)" u01
request INVITE sip-header Remote-Party-ID
modify "(.*)" "Remote-Party-ID: \u01"
!!! …
Assign the copy-list to inbound dial-peer
dial-peer
voice 1913 voip
session protocol sipv2
session target ipv4:10.45.230.10
incoming uri request InboundMatch
voice-class sip call-route url
voice-class sip copy-list 1
!!! …
Assign the normalization rule to outgoing dial-peer
dial-peer
voice 1912 voip
session protocol sipv2
session target ipv4:10.170.10.200
destination uri outgoingURI
voice-class sip profiles 10
voice-class sip requri-passing
Sent:
INVITE
sip:hquser1@hqcluster.local SIP/2.0
Via:
SIP/2.0/UDP 10.170.170.2:5060;branch=z9hG4bK47AF844
Remote-Party-ID: "SBUSER1-CALLER"
;party=calling;screen=yes;privacy=off
From:
"SBUSER1-CALLER" ;tag=9CF9483C-13B4
To:
Date: Tue, 23
Feb 2016 18:53:06 GMT
Call-ID:
82872A36-D99511E5-BF28D397-811E4C2B@10.170.170.2
Supported:
timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid:
2873105024-0000065536-0000000111-0182856970
User-Agent:
Cisco-SIPGateway/IOS-15.4.3.M3
Allow: INVITE,
OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO,
REGISTER
CSeq: 101
INVITE
Timestamp:
1456253586
Contact:
Expires: 180
Allow-Events:
telephone-event
Max-Forwards:
69
Session-Expires: 1800
Content-Length:
0
Derive Session Target from SIP-URI
CUBE can use the
host portion of Req-URI header in incoming INVITE message to findout the
session target. If the host portion is an IP address, CUBE will directly signal
the IP. However, in URI Dialing the host portion usually is a domain-name.
Hence, CUBE will resolve the domain name and signal the result IP.
dial-peer
voice 1912 voip
session protocol sipv2
session target sip-uri
destination uri outgoingURI
voice-class sip profiles 10
voice-class sip requri-passing
