Annekim: /* Table: Field Reference: SIP Configuration Page */ CSCtz67805

2012-05-04T17:17:13Z

Table: Field Reference: SIP Configuration Page: CSCtz67805

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-	~~Recommendation~~: Use ~~TCP~~ for the SIP signaling between Cisco Unified MeetingPlace and Cisco Unified Communications Manager~~. This is already the default~~ on ~~the Cisco Unified Communications Manager side~~.	+	Required: Use UDP for the SIP signaling between Cisco Unified MeetingPlace and Cisco Unified Communications Manager on all Hardware Media Server integrations.


	Default: UDP		Default: UDP
	\|}		\|}
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	===== Related Topics =====		===== Related Topics =====

Docwikibot: Bot: Adding {{Template:Required Metadata}}

2009-12-18T01:31:51Z

Bot: Adding {{Template:Required Metadata}}

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	'''Main page:''' [[Cisco Unified MeetingPlace, Release 7.0]]		'''Main page:''' [[Cisco Unified MeetingPlace, Release 7.0]]

MeetingPlace Express Moderator: 1 revision

2009-03-20T23:55:12Z

1 revision

← Older revision

Revision as of 23:55, 20 March 2009

MeetingPlace Express Moderator at 18:43, 13 March 2009

2009-03-13T18:43:49Z

New page

'''Main page:''' [[Cisco Unified MeetingPlace, Release 7.0]]

'''Up one level:''' [[Cisco Unified MeetingPlace, Release 7.0 -- Reference Information| Reference Information]]

To find this page, click '''System Configuration > Call Configuration > SIP Configuration'''.

===== Table: Field Reference: SIP Configuration Page =====

{| border = 1
|-
!Field
!Description
|-
|

Display name
|

Name that appears on Cisco Unified IP Phone screens during calls placed by Cisco Unified MeetingPlace.

Default: Cisco Unified MeetingPlace
|-
|

Username
|

Enter the same value that is in the [[Cisco Unified MeetingPlace, Release 7.0 -- Usage Configuration Page#Access phone number 1|Access phone number 1]] field on the [[Cisco Unified MeetingPlace, Release 7.0 -- Usage Configuration Page#Usage Configuration Page|Usage Configuration Page]].

The system uses this value in the "From" address for outgoing calls.

Default: 0000
|-
|

Only accept incoming calls from proxies
|

Whether to accept incoming calls from only the SIP proxy servers that are configured on the [[Cisco Unified MeetingPlace, Release 7.0 -- SIP Configuration Page#SIP Configuration Page|SIP Configuration Page]].

If you select Yes, then make sure that the configured SIP proxy servers include all Cisco Unified Communications Manager subscriber nodes that might send calls directly to this Cisco Unified MeetingPlace server.

Default: No
|-
|

SIP domain name
|

Domain name that is used in the "To" address for outgoing calls. If left blank, then the Application Server IP address is used instead.

Typically, this field is left blank when the [[Cisco Unified MeetingPlace, Release 7.0 -- SIP Configuration Page#SIP Proxy Server|SIP Proxy Server]] fields are populated with Cisco Unified Communications Manager information.

'''Caution! ''' If you enter a value that does not match the domain used by the SIP proxy servers, then Cisco Unified MeetingPlace will fail to dial out.

If outgoing calls fail, then configure this field to match the SIP domain used by the [[Cisco Unified MeetingPlace, Release 7.0 -- SIP Configuration Page#SIP Proxy Server|SIP Proxy Server]] or your local Cisco Unified Communications Manager node. In Cisco Unified Communications Manager, the SIP domain is specified under System > Enterprise Parameters in the Organization Top Level Domain field.

'''Note: ''' For details about any field in Cisco Unified Communications Manager, click the field name, or click '''Help > This Page'''.

Example: cisco.com
|-
|
'''SIP Proxy Server'''
|

Outgoing SIP calls are directed to these proxies. These proxies are also used to filter incoming calls if the [[Cisco Unified MeetingPlace, Release 7.0 -- SIP Configuration Page#Only accept incoming calls from proxies|Only accept incoming calls from proxies]] field is set to Yes.

'''Note: ''' At least one SIP proxy server must be configured to enable dial-out calls.

If multiple proxy servers are configured when placing a call, Cisco Unified MeetingPlace tries each one successively until it finds one that accepts the call. Note, however, that if a proxy does not respond at all (as opposed to rejecting the invite), the call attempt will probably time out before the next proxy is tried.
|-
|

Hostname or IP address
|

Hostname or IP address of the SIP proxy server.

In a Cisco Unified Communications Manager environment, the SIP proxy server should be the IP address of a Cisco Unified Communications Manager node.

Enter the FQDN (fully-qualified domain name) (or IP address) if the domain of the SIP proxy server differs from the domain that you configured in DNS when you installed Cisco Unified MeetingPlace.
|-
|

Port
|

TCP or UDP port on which the proxy listens for SIP calls.

Default: 5060
|-
|

Transport
|

Whether the port is TCP or UDP.

Recommendation: Use TCP for the SIP signaling between Cisco Unified MeetingPlace and Cisco Unified Communications Manager. This is already the default on the Cisco Unified Communications Manager side.

Default: UDP
|}

===== Related Topics =====

* [[Cisco Unified MeetingPlace, Release 7.0 -- Configuring Call Control for Cisco Unified MeetingPlace|Configuring Call Control for Cisco Unified MeetingPlace]]

[[Category: Reference Information for Cisco Unified MeetingPlace, Release 7.0]]

Cisco Unified MeetingPlace, Release 7.0 -- SIP Configuration Page - Revision history

Annekim: /* Table: Field Reference: SIP Configuration Page */ CSCtz67805

Docwikibot: Bot: Adding {{Template:Required Metadata}}

MeetingPlace Express Moderator: 1 revision

MeetingPlace Express Moderator at 18:43, 13 March 2009