Configuring SIP Trunk Support Cisco Call Manager Express with 2N VoiceBlue Lite

Released on = July 3, 2006, 7:17 am

Press Release Author = Nicole Konstantinidisova

Industry = Telecommunications

Press Release Summary = This procedure enables four SIP trunk support parameters

Press Release Body = This procedure enables four SIP trunk support parameters:
.Call forwarding over SIP networks-call-forward pattern and calling-number local
commands
.Call transfer over SIP networks-transfer-system and transfer-pattern commands
.DTMF relay-dtmf-relay rtp-nte or dtmf-relay sip-notify command and notify
telephone-event max-duration command
.SIP registrar-registrar, retry, and timers commands

SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. call-forward pattern pattern
5. calling-number local
6. transfer-system {full-blind | full-consult}
7. transfer-pattern transfer-pattern
8. exit
9. dial-peer voice tag voip
10. dtmf-relay rtp-nte
11. dtmf-relay sip-notify
12. exit
13. sip-ua
14. notify telephone-event max-duration time
15. registrar {dns:host-name | ipv4:ip-address} expires seconds [tcp] [secondary]
16. retry register number
17. timers register time
18. exit

DETAILED STEPS
Command or Action
Purpose

Step 1
enable
Example:
Router> enable

Enables privileged EXEC mode.
.Enter your password if prompted.

Step 2
configure terminal
Example:
Router# configure terminal

Enters global configuration mode.













telephony-service



Step 3
telephony-service
Example:
Router(config)# telephony-service

Enters telephony-service configuration mode.

Step 4
call-forward pattern pattern
Example:
Router(config-telephony)# call-forward pattern 4...


Specifies the H.450.3 standard or SIP 302 redirection method for call forwarding.
Calling-party numbers that do not match the patterns defined with this command are
forwarded using Cisco-proprietary call forwarding for backward compatibility.

.pattern-Digits to match for call forwarding using the H.450.3 standard or SIP 302
redirection method. A pattern of .T matches all calling-party numbers.

Note when defining forwards to non local numbers, it is important to note that
pattern-digit matching is performed before translation-rule operations. Therefore,
you should specify in this command the digits actually entered by phone users before
they are translated. For more information, see the \"Voice Translation Rules and
Profiles\" section on page 117.

Step 5
calling-number local
Example:
Router(config-telephony)# calling-number local

(Optional) Replaces a calling-party number and name with the forwarding-party
(local) number and name.

Step 6
transfer-system {full-blind | full-consult}
Example:
Router(config-telephony)# transfer-system full-consult

Defines the call transfer method for all lines served by the router.
Note For SIP networks, use only the full-blind keyword or the full-consult keyword.
For more information, see the Cisco IOS SIP Configuration Guide.

.full-blind-Calls are transferred without consultation using H.450.2 standard methods.
.full-consult-Calls are transferred with consultation using H.450.2 standard methods
and a second phone line if available. The calls fall back to full-blind if the
second line is unavailable.

Step 7
transfer-pattern transfer-pattern
Example:
Router(config-telephony)# transfer-pattern 52540..




Allows transfer of telephone calls by Cisco Unified IP phones to specified phone
number patterns. If no transfer pattern is set, the default is that transfers are
permitted only to other local IP phones.

.transfer-pattern-String of digits for permitted call transfers. Wildcards are
allowed.

Note when defining transfers to non local numbers, it is important to note that
transfer-pattern digit matching is performed before translation-rule operations.
Therefore, you should specify in this command the digits that are actually entered
by phone users before they are translated. For more information, see the \"Voice
Translation Rules and Profiles\" section on page 117.

Step 8
exit
Example:
Router(config-telephony)# exit

Exits telephony-service configuration mode.

Configuration EXAMPLE:

telephony-service
load 7960-7940 P0030702T023
max-ephones 24
max-dn 24
ip source-address 172.24.34.160 port 2000
time-format 24
date-format dd-mm-yy
max-conferences 12 gain -6
moh music-on-hold.au
web admin system name xxx secret xxx
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 May 11 2006 17:52:56



dial-peer voice

Step 9
dial-peer voice tag voip
Example:
Router(config)# dial-peer voice 2 voip

Enters dial-peer configuration mode.

Step 10
dtmf-relay rtp-nte
Example:
Router(config-dial-peer)# dtmf-relay rtp-nte

Forwards DTMF tones by using Real-Time Transport Protocol (RTP) with the Named
Telephone Event (NTE) payload type. This enables DTMF relay using the RFC 2833
standard method.

Step 11
dtmf-relay sip-notify
Example:
Router(config-dial-peer)# dtmf-relay sip-notify

Forwards DTMF tones using SIP NOTIFY messages.

Step 12
exit
Example:
Router(config-dial-peer)# exit

Exits dial-peer configuration mode.

Configuration EXAMPLE:

dial-peer voice 20 voip
destination-pattern 004219[01]T
session protocol sipv2
session target ipv4:172.24.34.169
dtmf-relay sip-notify
codec g711ulaw
no vad


sip-ua



Step 13
sip-ua
Example:
Router(config)# sip-ua

Enters SIP user-agent configuration mode.

Step 14
notify telephone-event max-duration time
Example:
Router(config-sip-ua)# notify telephone-event max-duration 2000

Configures the maximum time interval allowed between two consecutive NOTIFY messages
for a single DTMF event.

.max-duration time-Time interval between consecutive NOTIFY messages for a single
DTMF event, in milliseconds. Range is from 500 to 3000. Default is 2000.

Step 15
registrar {dns:host-name | ipv4:ip-address} expires seconds [tcp] [secondary]
Example:
Router(config-sip-ua)# registrar ipv4:10.8.17.40 expires 3600 secondary

Registers E.164 numbers on behalf of analog telephone voice ports (FXS) and IP phone
virtual voice ports (EFXS) with an external SIP proxy or SIP registrar server.

.dns:host-name-Domain name server that resolves the name of the dial peer to receive
calls.
.ipv4:ip-address-IP address of the dial peer to receive calls.
.expires seconds-Default registration time, in seconds.
.tcp-(Optional) Sets the transport layer protocol to TCP. UDP is the default.
.secondary-(Optional) Specifies registration with a secondary SIP proxy or registrar
for redundancy purposes.




Step 16
retry register number
Example:
Router(config-sip-ua)# retry register 10

Sets the total number of SIP Register messages that the gateway should send.

.number-Number of Register message retries. Range is from 1 to 10. Default is 10.

Step 17
timers register time
Example:
Router(config-sip-ua)# timers register 500

Sets how long the SIP user agent (UA) waits before sending Register requests.

.time-Waiting time, in milliseconds. Range is from 100 to 1000. Default is 500.


Step 18
exit
Example:
Router(config-sip-ua)# exit

Exits SIP user-agent configuration mode.

Configuration EXAMPLE:

sip-ua
!
!
!
voice register global
mode cme
source-address 172.24.34.160 port 5060
load 7960-7940 P0S3-07-4-00
create profile sync 0052141959334142



Verifying SIP Trunk Support Features


Step 1. Use the show running-config command to verify dial-peer, telephony-service,
and SIP UA parameter values.


Call Forwarding over SIP Networks: Example


The following example enables call forwarding using the H.450.3 standard or SIP 302
response:

dial-peer voice 100 pots
destination-pattern 9.T
port 1/0/0
!
dial-peer voice 4000 voip
destination-pattern 4...
session protocol sipv2
session-target ipv4:1.1.1.1
!
telephony-service
call-forward pattern 4...

Call Transfer over SIP Networks: Example

The following example specifies transfer with consultation using the H.450.2
standard for all IP phones serviced by the router:
!
dial-peer voice 100 pots
destination-pattern 9.T
port 1/0/0
!
dial-peer voice 4000 voip
destination-pattern 4...
session protocol sipv2
session-target ipv4:1.1.1.1
!
telephony-service
transfer-pattern 4...
transfer-system full-consult


DTMF Relay using RFC 2833: Example

The following example specifies use of the RFC 2833 method for in-band DTMF relay
for calls using dial peer 2.

dial-peer voice 2 voip
dtmf-relay rtp-nte

sip-ua
notify telephone-event max-duration 2000


DTMF Relay using SIP Notify:Example

The following example specifies use of the SIP notify method for in-band DTMF relay
for calls using dial peer 4.

dial-peer voice 4 voip
dtmf-relay sip-notify

sip-ua
notify telephone-event max-duration 2000


SIP Register Support: Example

The following example sets up the gateway to register the gateway\'s E.164 telephone
numbers with an external SIP registrar.

sip-ua
registrar ipv4:10.8.17.40 expires 3600 secondary
retry register 10
timers register 500

Troubleshooting SIP Trunk Support Features

Step 1 The show sip-ua status command output displays the time interval between
consecutive NOTIFY messages for a telephone event. In the following example, the
time interval is 2000 ms.


Router#
show sip-ua status
SIP User Agent Status
SIP User Agent for UDP :ENABLED
SIP User Agent for TCP :ENABLED
SIP User Agent bind status(signaling):DISABLED
SIP User Agent bind status(media):DISABLED
SIP early-media for 180 responses with SDP:ENABLED
SIP max-forwards :6
SIP DNS SRV version:2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP:NONE
Check media source packets:DISABLED
Maximum duration for a telephone-event in NOTIFYs:2000 ms
SIP support for ISDN SUSPEND/RESUME:ENABLED
Redirection (3xx) message handling:ENABLED
SDP application configuration:
Version line (v

Web Site = http://www.2n.cz

Contact Details = Nicole
Modranska 621
Praha 4; 143 01
Prague, Czech Republic

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