FonoSIP.com
lang world mobile site


See also CME config example


See also Cisco Auto QOS

Providing Cisco CME Support for SIP

Contents

DTMF Relay for SIP Applications and Voice Mail

SIP Register Support

Call Transfer over SIP Networks

Call Forwarding over SIP Networks


Providing Cisco CME Support for SIP


Cisco CallManager Express (Cisco CME) supports incoming and outgoing Session Initiation Protocol (SIP) calls to and from IP phones and router voice gateway voice ports, but does not support direct attachment of SIP phones to Cisco CME. Special configurations to support SIP calls are described in this appendix.

For more information about SIP, refer to the Cisco IOS SIP Configuration Guide.

Contents

DTMF Relay for SIP Applications and Voice Mail

SIP Register Support

Call Transfer over SIP Networks

Call Forwarding over SIP Networks

DTMF Relay for SIP Applications and Voice Mail

To use remote voice mail or interactive voice response (IVR) applications on SIP networks from Cisco CME phones, you must enable a Cisco-proprietary out-of-band dual tone multifrequency (DTMF) relay method for SIP.

Cisco Skinny Client Control Protocol (SCCP) phones, such as those used with Cisco CME systems, do not support the standard in-band DTMF relay mechanism used by SIP phones to send keypad digits. To enable SCCP phones to send digit information to SIP-based IVR applications, Cisco CME 3.0 and later versions use what is known as the SIP DTMF relay method. You select this method in the SIP VoIP dial peer using the dtmf-relay sip-notify command.

The SIP DTMF relay method is needed in the following situations:

When SIP is used to connect a Cisco CME system to a SIP-based IVR or voice-mail application.

When SIP is used to connect a Cisco CME system to a SIP-PSTN voice gateway that goes through the PSTN to a voice-mail or IVR application.

Note that the need to use out-of-band DTMF relay is limited to SCCP phones. SIP phones support in-band DTMF relay as specified in RFC 2833.

To enable SIP DTMF relay, the commands in this section must be used on both originating and terminating gateways. For more information, refer to SIP Gateway Enhancements, Cisco IOS Release 12.2(15)ZJ.

SUMMARY STEPS

1. dial-peer voice tag voip

2. dtmf-relay sip-notify

3. exit

4. sip-ua

5. notify telephone-event max-duration time

6. exit

DETAILED STEPS

 
Command or Action
Purpose

Step 1 

dial-peer voice tag voip

Example:

Router(config)# dial-peer voice 2 voip

Enters dial-peer configuration mode.

Step 2 

dtmf-relay sip-notify

Example:

Router(config-dial-peer)# dtmf-relay sip-notify

Forwards DTMF tones using SIP NOTIFY messages.

Step 3 

exit

Example:

Router(config-dial-peer)# exit

Exits dial-peer configuration mode.

Step 4 

sip-ua

Example:

Router(config)# sip-ua

Enables SIP user-agent configuration mode.

Step 5 

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 6 

exit

Example:

Router(config-sip-ua)# exit

Exits SIP user-agent configuration mode.

Troubleshooting Tips

The dial-peer section of the show running-config command output displays DTMF relay status when it is configured, as shown in this excerpt:

dial-peer voice 123 voip 
 destination-pattern [12]... 
 monitor probe icmp-ping 
 session protocol sipv2 
 session target ipv4:10.8.17.42 
 dtmf-relay sip-notify 

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=) required
 Owner line (o=) required
 Timespec line (t=) required
 Media supported:audio image 
 Network types supported:IN 
 Address types supported:IP4 
 Transport types supported:RTP/AVP udptl 

SIP Register Support

This section describes how to enable a SIP gateway to register E.164 numbers with a SIP proxy or SIP registrar, similar to the way that H.323 gateways can register E.164 numbers with a gatekeeper. SIP gateways allow registration of E.164 numbers to a SIP proxy or registrar on behalf of analog telephone voice ports (FXS) and IP phone virtual voice ports (EFXS) for local SCCP phones.

When registering E.164 numbers in dial peers with an external registrar, you can also register them with a secondary SIP proxy or registrar to provide redundancy. The secondary registration can be used if the primary registrar fails.

For more detailed information, refer to SIP Gateway Enhancements, Cisco IOS Release 12.2(15)ZJ.


Note There are no commands that allow registration between the H.323 and SIP protocols.


By default, SIP gateways do not generate SIP Register messages, so the following steps are needed to set up the gateway to register the gateway's E.164 telephone numbers with an external SIP registrar.

SUMMARY STEPS

1. sip-ua

2. registrar {dns:host-name | ipv4:ip-address} expires seconds [tcp] [secondary]

3. retry register number

4. timers register time

5. exit

DETAILED STEPS

 
Command or Action
Purpose

Step 1 

sip-ua

Example:

Router(config)# sip-ua

Enables SIP user-agent configuration mode.

Step 2 

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 3 

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 4 

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 5 

exit

Example:

Router(config-sip-ua)# exit

Exits SIP user-agent configuration mode.

Troubleshooting Tips

Use the show sip-ua timers command to show the waiting time before Register requests are sent; that is, the value that has been set with the timers register command.

Use the show sip-ua register status command to show the status of local E.164 registrations.

Use the show sip-ua statistics command to show the Register messages that have been sent.

Call Transfer over SIP Networks

Cisco CME supports all SIP Refer method call transfer scenarios. Before configuring the SIP Refer method, follow the steps in this section to enable call transfer using H.450.2 standards. Note that the transfer-system command must be configured with the full-blind or full-consult keyword for SIP Refer to be invoked.

SUMMARY STEPS

1. telephony-service

2. transfer-system {full-blind | full-consult}

3. transfer-pattern transfer-pattern

4. exit

DETAILED STEPS

 
Command or Action
Purpose

Step 1 

telephony-service

Example:

Router(config)# telephony-service

Enters telephony-service configuration mode.

Step 2 

transfer-system {full-blind | full-consult}

Example:

Router(config-telephony-service)# 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 a second phone line if available. The calls fall back to full-blind if the second line is unavailable.

Step 3 

transfer-pattern transfer-pattern

Example:

Router(config-telephony-service)# transfer-pattern 52540..

Allows transfer of telephone calls by Cisco 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 nonlocal 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, refer to the "Translation Rules" section in the " Setting Up Phones in a Cisco CME System" chapter.

Step 4 

exit

Example:

Router(config-telephony-service)# exit

Exits telephony-service configuration mode.

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

What to Do Next

After using the call transfer commands for Cisco CME, you need to configure SIP call transfer, which is described in the " Configuring SIP Call Transfer" chapter of the Cisco IOS SIP Configuration Guide.

Call Forwarding over SIP Networks

Call forwarding over SIP networks uses the 302 Moved Temporarily SIP response, which works in a manner similar to the way in which the H.450.3 standard is used for H.323 networks. To enable call forwarding, use the call-forward pattern command and specify a pattern that matches the calling-party numbers of the calls that you want to be able to forward. Use the call-forward pattern command with the .T pattern to allow all calls for all possible SIP calling parties to be forwarded using the SIP 302 response.

SUMMARY STEPS

1. telephony-service

2. call-forward pattern pattern

3. calling-number local

4. exit

DETAILED STEPS

 
Command or Action
Purpose

Step 1 

telephony-service

Example:

Router(config)# telephony-service

Enters telephony-service configuration mode.

Step 2 

call-forward pattern pattern

Example:

Router(config-telephony-service)# 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 (as described in the " Configuring Call Forwarding" chapter in the Cisco IOS Telephony Services V2.1 guide).

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 nonlocal 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 "Translation Rules" section in the " Setting Up Phones in a Cisco CME System" chapter.

Step 3 

calling-number local

Example:

Router(config-telephony-service)# calling-number local

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

Note This command applies to hairpin-forwarded calls only and requires installation of the app-h450-transfer.2.0.0.8.tcl script or a later version. The local-hairpin attribute-value (AV) pair must be set to 1.

Step 4 

exit

Example:

Router(config-telephony-service)# exit

Exits telephony-service configuration mode.

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...

What to Do Next

After using the call forwarding commands for Cisco CME, you need to configure SIP call forwarding, which is described in the " Configuring SIP Call Transfer" chapter of the Cisco IOS SIP Configuration Guide.



-- FonoSIP.com -- Search Site -- About Us -- Add Credit -- Mobile Site -- Login -- Service Numbers -- Help -- Swag -- Write to us info@fonosip.com -- Voicemail 1-360-226-7665 -- FonoSIP.com © 2010

Powered by BA.net iGames


FonoSIP.com supports Xten / Counterpath SIP softphones for Windows, Mac, iPhone and Linux. Internet telephony equipment such as Linksys PAP2, Cisco, Sipura, Nokia N71. Apple iPhone, iPod Touch, iPad. Android. Codec G729. Fring Mobile. Bring your Own Device BYOD. We also support SIP Trunking, replace your phone lines. Asterisk PBX, TrixBox.