SIP Profile Field Descriptions
------------------------------
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-------------------------------------------------------------+
| Option | Description |
+================+=============================================================+
| Name | Enter a name to identify the SIP profile; for example, |
| (Mandatory) | SIP\_7905. The value can include 1 to 50 characters, |
| | including alphanumeric characters, dot, dash, and |
| | underscores. |
+----------------+-------------------------------------------------------------+
| Description | This field identifies the purpose of the SIP profile; for |
| (Optional) | example, SIP for 7970. The description can include up to |
| | 50 characters in any language, but it cannot include |
| | double-quotes ("), percentage sign (%), ampersand (&), |
| | back-slash (\\), or angle brackets (<>). |
+----------------+-------------------------------------------------------------+
| Default MTP | This field specifies the default payload type for RFC2833 |
| Telephony | telephony event. See RFC 2833 for more information. |
| Event Payload | Usually, the default value specifies the appropriate |
| Type | payload type. Be sure that you have a good understanding |
| (Optional) | of this parameter before changing it, as changes could |
| | result in DTMF tones not being received or generated. |
| | |
| | Default-101 |
| | |
| | Range-96 to 127 |
| | |
| | This parameter's value affects calls with the following |
| | conditions: |
| | |
| | - An outgoing SIP call from Cisco Unified Communications |
| | Manager |
| | - For the calling SIP trunk, the **Media Termination |
| | Point Required** check box is checked on the SIP Trunk |
| | Configuration window |
+----------------+-------------------------------------------------------------+
| Early Offer | This feature supports both standards-based G.Clear |
| for G.Clear | (CLEARMODE) and proprietary Cisco Session Description |
| Calls | Protocols (SDP). |
| (Optional) | |
| | To enable or disable Early Offer for G.Clear Calls, |
| | choose one of the following options: |
| | |
| | - Disabled |
| | - CLEARMODE |
| | - CCD |
| | - G.nX64 |
| | - X-CCD |
+----------------+-------------------------------------------------------------+
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-------------------------------------------------------------+
| Option | Description |
+================+=============================================================+
| User-Agent and | This feature indicates how Unified CM handles the |
| Server header | User-Agent and Server header information in a SIP |
| information | message. |
| (Mandatory) | |
| | Choose one of the following options: |
| | |
| | - **Send Unified CM Version Information as User-Agent |
| | Header** - For INVITE requests, the User-Agent header is |
| | included with the CM version header information. For |
| | responses, the Server header is omitted. Unified CM |
| | passes any contact headers through untouched. |
| | - **Pass Through Received Information as Contact Header |
| | Parameters** - If selected, the User-Agent and Server |
| | header information is passed as Contact header |
| | parameters. The User-Agent and Server header is |
| | derived from the received Contact header parameters, |
| | if present. Otherwise, they are taken from the |
| | received User-Agent and Server headers. |
| | - **Pass Through Received Information as User-Agent and |
| | Server Header** - If selected, the User-Agent and Server |
| | header information is passed as User-Agent and Server |
| | headers. The User-Agent and Server header is derived |
| | from the received Contact header parameters, if |
| | present. Otherwise, they are taken from the received |
| | User-Agent and Server headers. |
| | |
| | Default: Send Unified CM Version Information as |
| | User-Agent Header |
+----------------+-------------------------------------------------------------+
| Version in | This field specifies the portion of the installed build |
| User Agent and | version that is used as the value of the User Agent and |
| Server Header | Server Header in SIP requests. Possible values are: |
| (Mandatory) | |
| | - **Major and Minor**; for example, Cisco-CUCM10.6 |
| | - **Major:** for example, Cisco-CUCM10 |
| | - **Major, Minor and Revision**; for example, |
| | Cisco-CUCM10.6.2 |
| | - **Full Build**; for example, Cisco-CUCM10.6.2.98000-19 |
| | - **None**; header is omitted |
| | |
| | Default: Major and Minor |
+----------------+-------------------------------------------------------------+
| Dial String | Possible values are: |
| Interpretation | |
| (Mandatory) | - Phone number consists of characters 0-9, \*, #, and + |
| | (others treated as URI addresses). This is the default |
| | value. |
| | - Phone number consists of characters 0-9, A-D, \*, #, |
| | and + (others treated as URI addresses) |
| | - Always treat all dial strings as URI addresses |
+----------------+-------------------------------------------------------------+
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-------------------------------------------------------------+
| Option | Description |
+================+=============================================================+
| Redirect by | If you select this check box and configure this SIP Profile |
| Application | on the SIP trunk, the Unified CM administrator can: |
| (Optional) | |
| | - Apply a specific calling search space to redirected |
| | contacts that are received in the 3xx response. |
| | - Apply digit analysis to the redirected contacts to |
| | make sure that the calls get routed correctly. |
| | - Prevent a DOS attack by limiting the number of |
| | redirection (recursive redirection) that a service |
| | parameter can set. |
| | - Allow other features to be invoked while the |
| | redirection is taking place. |
| | |
| | Getting redirected to a restricted phone number (such as |
| | an international number) means that handling redirection |
| | at stack level causes the call to be routed, not blocked. |
| | This behavior occurs if you leave the **Redirect by |
| | Application** check box clear. |
+----------------+-------------------------------------------------------------+
| Disable Early | By default, Unified CM signals the calling phone to play |
| Media on 180 | local ringback if SDP is not received in the 180 or 183 |
| (Optional) | response. If SDP is included in these responses, instead |
| | of playing ringback locally, Unified CM connects media. |
| | The calling phone then plays whatever the called device |
| | is sending (such as ringback or busy signal). If you |
| | receive no ringback, the device you are connecting to may |
| | include SDP in the 180 response, but not send media |
| | before 200OK response. In this case, select this check box |
| | to play local ringback on the calling phone and connect the |
| | media upon receipt of the 200OK response. |
| | |
| | Note: |
| | |
| | Even though the phone that is receiving ringback is |
| | the calling phone, you need the configuration on the |
| | called device profile because it determines the |
| | behavior. |
+----------------+-------------------------------------------------------------+
| Outgoing T.38 | The parameter allows the system to accept a signal from |
| INVITE include | Microsoft Exchange that causes it to switch the call from |
| audio mline | audio to T.38 fax. To use this feature, configure a SIP |
| (Optional) | trunk with this SIP profile. |
| | |
| | Note: |
| | |
| | The parameter applies to SIP trunks only, not phones |
| | that are running SIP or other endpoints. |
+----------------+-------------------------------------------------------------+
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-------------------------------------------------------------+
| Option | Description |
+================+=============================================================+
| Use Fully | This feature enables Unified CM to relay a caller's |
| Qualified | alphanumeric hostname by passing it to the called device |
| Domain Name in | or outbound trunk as SIP header information. Enter one of |
| SIP Requests | the following: |
| (Optional) | |
| | **f** - To disable this option. The IP address for Unified |
| | CM is passed to the line device or outbound trunk instead |
| | of the user’s hostname. |
| | |
| | **t** - To enable this option. Unified CM relays an |
| | alphanumeric hostname of a caller by passing it through |
| | to the called endpoint as a part of the SIP header |
| | information. This enables the called endpoint to return |
| | the call using the received or missed call list. If the |
| | call originates from a line device on the Unified CM |
| | cluster, and is routed on a SIP trunk, then the |
| | configured Organizational Top-Level Domain (for example, |
| | Cisco.com) is used in the Identity headers, such as From, |
| | Remote-Party-ID, and P-Asserted-ID. If the call |
| | originates from a trunk on Unified CM and is being routed |
| | on a SIP trunk, then: |
| | |
| | - If the inbound call provides a host or domain in the |
| | caller’s information, the outbound SIP trunk messaging |
| | preserves the hostname in the Identity headers, such |
| | as From, Remote-Party-ID, and P-Asserted-ID. |
| | - If the inbound call does not provide a host or domain |
| | in the caller's information, the configured |
| | Organizational Top-Level Domain is used in the |
| | Identity headers, such as From, Remote-Party-ID, and |
| | P-Asserted-ID. |
| | |
| | Default: f - Disabled |
+----------------+-------------------------------------------------------------+
| Assured | Select this check box for third-party AS-SIP endpoints and |
| Services SIP | AS-SIP trunks to ensure proper Assured Service behavior. |
| conformance | This setting provides specific Assured Service behavior |
| (Optional) | that affects services such as Conference factory and |
| | SRTP. |
+----------------+-------------------------------------------------------------+
Table: SDP Information Tab
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-------------------------------------------------------------+
| Option | Description |
+================+=============================================================+
| SDP | Displays the SDP Transparency Profile Setting (read-only) |
| Transparency | |
| Profile | |
| (Optional) | |
+----------------+-------------------------------------------------------------+
| Accept Audio | Choose one of the following options: |
| Codec | |
| Preferences in | - **On** - Enables Unified CM to honor the preference of |
| Received Offer | audio codecs in the received offer and preserve it |
| (Optional) | while processing. |
| | - **Off** - Enables Unified CM to ignore the preference of |
| | audio codecs in a received offer and apply the locally |
| | configured Audio Codec Preference List. The default |
| | selects the service parameter configuration. |
| | - **Default** - Selects the service parameter |
| | configuration. |
| | |
| | Default: Default |
+----------------+-------------------------------------------------------------+
| Require SDP | This feature determines how Unified CM handles midcall |
| Inactive | updates to codecs or connection information such as IP |
| Exchange for | address or port numbers. |
| Mid-Call Media | |
| Change | If you select this check box, during midcall codec or |
| (Optional) | connection updates Unified CM sends an INVITE a-inactive SDP|
| | message to the endpoint to break the media exchange. This is|
| | required if an endpoint is not capable of reacting to |
| | changes in the codec or connection information without |
| | disconnecting the media. This applies only to audio and |
| | video streams within SIP-SIP calls. |
| | |
| | Note |
| | |
| | For early offer enabled SIP trunks, the Send |
| | send-receive SDP in midcall INVITE parameter |
| | overrides this parameter. |
| | |
| | If this check box is clear, Unified CM passes the midcall |
| | SDP to the peer leg without sending a prior Inactive SDP |
| | to break the media exchange. |
| | |
| | Default: Clear |
+----------------+-------------------------------------------------------------+
| Allow RR/RS | Specifies the RR (RTDP bandwidth allocated to other |
| bandwidth | participants in an RTP session) and RS (RTCP bandwidth |
| modifier (RFC | allocated to active data senders) in RFC 3556. Options |
| 3556) | are: |
| (Mandatory) | |
| | - Transport Independent Application Specific bandwidth |
| | modifier (TIAS) and AS |
| | - TIAS only |
| | - AS only |
| | - CT only |
| | |
| | Default: TIAS and AS |
+----------------+-------------------------------------------------------------+
Table: Parameters used in Phone Tab
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-----------------------------------------------------------+
| Option | Description |
+================+===========================================================+
| Timer Invite | This field specifies the time, in seconds, after which a |
| Expires | SIP INVITE expires. The Expires header uses this value. |
| (seconds) | |
| (Optional) | Valid values: Any positive number |
| | |
| | Default: 180 seconds |
+----------------+-----------------------------------------------------------+
| Timer Register | This field is intended to be used by SIP endpoints only. |
| Delta | The endpoint receives this value through a TFTP config |
| (seconds) | file. The endpoint reregisters Timer Register Delta |
| (Optional) | seconds before the registration period ends. The |
| | registration period gets determined by the value of the |
| | ``SIP Station KeepAlive Interval`` |
| | service parameter. |
| | |
| | Valid values: 0 to 32767 |
| | |
| | Default: 5 seconds |
+----------------+-----------------------------------------------------------+
| Timer Register | This field is intended to be used by SIP endpoints only. |
| Expires | The SIP endpoint receives the value through a TFTP config |
| (seconds) | file. This field specifies the value that the phone that |
| (Optional) | is running SIP sends in the Expires header of the |
| | REGISTER message. Valid values include any positive |
| | number; however, 3600 (1 hour) specifies the default |
| | value. |
| | |
| | Valid values: Any positive number |
| | |
| | Default: 3600 seconds (1 hour) |
| | |
| | If the endpoint sends a shorter Expires value than the |
| | ``SIP Station Keepalive Interval`` service parameter, |
| | Unified CM responds with a 423 "Interval Too Brief." |
| | |
| | If the endpoint sends a greater Expires value than the |
| | ``SIP Station Keepalive Interval`` service parameter, |
| | Unified CM responds with a 200 OK with the Keepalive |
| | Interval value for Expires. |
| | |
| | Note: |
| | |
| | For mobile phones running SIP, Unified CM uses this value |
| | instead of the ``SIP Station KeepAlive Interval`` service |
| | parameter to determine the registration period. |
| | |
| | Note: |
| | |
| | For TCP connections, the value for the |
| | ``Timer Register Expires`` field must be lower than the |
| | value for the ``SIP TCP Unused Connection`` service |
| | parameter. |
+----------------+-----------------------------------------------------------+
| Timer T1 | This field specifies the lowest value, in milliseconds, |
| (msec) | of the retransmission timer for SIP messages. |
| (Optional) | |
| | Valid values: Any positive number |
| | |
| | Default: 500 msec |
+----------------+-----------------------------------------------------------+
| Timer T2 | This field specifies the highest value, in milliseconds, |
| (msec) | of the retransmission timer for SIP messages. |
| (Optional) | |
| | Valid values: Any positive number |
| | |
| | Default: 4000 msec |
+----------------+-----------------------------------------------------------+
| Retry INVITE | This field specifies the maximum number of times that an |
| (Optional) | INVITE request gets retransmitted. |
| | |
| | Valid values: Any positive number |
| | |
| | Default: 6 |
+----------------+-----------------------------------------------------------+
| Retry | This field specifies the maximum number of times that a |
| Non-INVITE | SIP message other than an INVITE request gets |
| (Optional) | retransmitted. |
| | |
| | Valid values: Any positive number |
| | |
| | Default: 10 |
+----------------+-----------------------------------------------------------+
| Start Media | This field designates the start real-time protocol (RTP) |
| Port | port for media. |
| (Optional) | |
| | Range: 2048 to 65535 |
| | |
| | Default: 16384 |
+----------------+-----------------------------------------------------------+
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-------------------------------------------------------------+
| Option | Description |
+================+=============================================================+
| Stop Media | This field designates the stop real-time protocol (RTP) |
| Port | port for media. |
| (Optional) | |
| | Range: 2048 to 65535 |
| | |
| | Default: 32766 |
+----------------+-------------------------------------------------------------+
| Call Pickup | This URI provides a unique address that the phone that is |
| URI (Optional) | running SIP sends to Unified CM to invoke the call pickup |
| | feature. |
+----------------+-------------------------------------------------------------+
| Call Pickup | This URI provides a unique address that the phone that is |
| Group URI | running SIP sends to Unified CM to invoke the call pickup |
| (Optional) | group feature. |
+----------------+-------------------------------------------------------------+
| Meet Me | This URI provides a unique address that the phone that is |
| Service URI | running SIP sends to Unified CM to invoke the meet me |
| (Optional) | conference feature. |
+----------------+-------------------------------------------------------------+
| User Info | This field configures the user- parameter in the REGISTER |
| (Optional) | message. Valid values are: |
| | |
| | - **None** - No value is inserted |
| | - **Phone** - The value user-phone is inserted in the To, |
| | From, and Contact Header for REGISTER |
| | - **IP** - The value user-ip is inserted in the To, From, |
| | and Contact Header for REGISTER |
| | |
| | Default: None |
+----------------+-------------------------------------------------------------+
| DTMF DB Level | This field specifies the in-band DTMF digit tone level. |
| (Optional) | Valid values are: |
| | |
| | - 6 dB below nominal |
| | - 3 dB below nominal |
| | - Nominal |
| | - 3 dB above nominal |
| | - 6 dB above nominal |
| | |
| | Default: Nominal |
+----------------+-------------------------------------------------------------+
| Call Hold Ring | This parameter causes the phone to ring in cases where |
| Back | you have another party on hold when you hang up a call. |
| (Optional) | Valid values are: |
| | |
| | - **Off** - Off permanently and cannot be turned on and |
| | off locally by the user interface |
| | - **On** - On permanently and cannot be turned on and off |
| | locally by the user interface |
+----------------+-------------------------------------------------------------+
| Anonymous Call | The field configures anonymous call block. Valid values |
| Block | are: |
| (Optional) | |
| | - **Off** - Disabled permanently and cannot be turned on |
| | and off locally by the user interface |
| | - **On** - Enabled permanently and cannot be turned on and |
| | off locally by the user interface |
+----------------+-------------------------------------------------------------+
| Caller ID | This field configures caller ID blocking. When blocking |
| Blocking | is enabled, the phone blocks its own number or email |
| (Optional) | address from phones that have caller identification |
| | enabled. Valid values are: |
| | |
| | - **Off** - Disabled permanently and cannot be turned on |
| | and off locally by the user interface |
| | - **On** - Enabled permanently and cannot be turned on and |
| | off locally by the user interface |
+----------------+-------------------------------------------------------------+
| Do Not Disturb | This field sets the Do Not Disturb (DND) feature. Valid |
| Control | values are: |
| (Optional) | |
| | - **User** - The dndControl parameter for the phone |
| | specifies 0. |
| | - **Admin** - The dndControl parameter for the phone |
| | specifies 2. |
+----------------+-------------------------------------------------------------+
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+------------------------------------------------------------+
| Option | Description |
+================+============================================================+
| Telnet Level | Cisco Unified IP Phones 7940 and 7960 do not support SSH |
| for 7940 and | for sign-in access or HTTP that is used to collect logs. |
| 7960 | However, these phones support Telnet, which lets the user |
| (Optional) | control the phone, collect debugs, and look at |
| | configuration settings. This field controls the |
| | telnet\_level configuration parameter with the following |
| | possible values: |
| | |
| | - **Disabled** - No access |
| | - **Limited** - Some access but cannot run privileged |
| | commands |
| | - **Enabled** - Full access |
+----------------+------------------------------------------------------------+
| Resource | This field enables the administrator to select one of the |
| Priority | cluster's defined Resource Priority Namespace network |
| Namespace | domains for assignment to a line using its SIP Profile. |
| (Optional) | |
+----------------+------------------------------------------------------------+
| Timer Keep | Unified CM requires a keepalive mechanism to support |
| Alive Expires | redundancy. This field specifies the interval between |
| (seconds) | keepalive messages sent to the backup Unified CM to |
| (Optional) | ensure its availability for failover. |
| | |
| | Default: 120 seconds |
+----------------+------------------------------------------------------------+
| Timer | This field specifies the time, in seconds, after which a |
| Subscribe | subscription expires. This value gets inserted into |
| Expires | the\ `` Expires`` header field. |
| (seconds) | |
| (Optional) | Valid values: Any positive number |
| | |
| | Default: 120 seconds |
+----------------+------------------------------------------------------------+
| Timer | Use this parameter with the ``Timer Subscribe Expires`` |
| Subscribe | setting. The phone resubscribes Timer Subscribe Delta |
| Delta | seconds before the subscription period ends, as governed |
| (seconds) | by ``Timer Subscribe Expires``. |
| (Optional) | |
| | Range: 3 to 15 seconds |
| | |
| | Default: 5 seconds |
+----------------+------------------------------------------------------------+
| Maximum | Use this configuration variable to determine the maximum |
| Redirections | number of times that the phone allows a call to be |
| (Optional) | redirected before dropping the call. |
| | |
| | Default: 70 redirections |
+----------------+------------------------------------------------------------+
| Off hook To | This field specifies the time in microseconds that passes |
| First Digit | when the phone goes off hook and the first digit timer |
| Timer (msec) | gets set. |
| (Optional) | |
| | Range: 0 to 15,000 microseconds |
| | |
| | Default: 15,000 microseconds |
+----------------+------------------------------------------------------------+
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+------------------------------------------------------------+
| Option | Description |
+================+============================================================+
| Call Forward | This URI provides a unique address that the phone that is |
| URI (Optional) | running SIP sends to Unified CM to invoke the call |
| | forward feature. |
+----------------+------------------------------------------------------------+
| Speed Dial | This URI provides a unique address that the phone that is |
| (Abbreviated | running SIP sends to Unified CM to invoke the abbreviated |
| Dial) URI | dial feature. |
| (Optional) | |
| | Speed dials that are not associated with a line key |
| | (abbreviated dial indices) do not download to the phone. |
| | The phone uses the feature indication mechanism (INVITE |
| | with Call-Info header) to indicate when an abbreviated |
| | dial number has been entered. The request URI contains |
| | the abbreviated dial digits (for example, 14), and the |
| | Call-Info header indicates the abbreviated dial feature. |
| | Unified CM translates the abbreviated dial digits into |
| | the configured digit string and extends the call with |
| | that string. If no digit string has been configured for |
| | the abbreviated dial digits, a 404 Not Found response |
| | gets returned to the phone. |
+----------------+------------------------------------------------------------+
| Conference | Select this check box to join the remaining conference |
| Join Enabled | participants when a conference initiator using a Cisco |
| (Optional) | Unified IP Phone 7940 or 7960 hangs up. Leave it |
| | clear if you do not want to join the remaining |
| | conference participants. |
| | |
| | Note: |
| | |
| | This check box applies to the Cisco Unified IP Phones |
| | 7941/61/70/71/11 when they are in SRST mode only. |
+----------------+------------------------------------------------------------+
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+------------------------------------------------------------+
| Option | Description |
+================+============================================================+
| RFC 2543 Hold | Select this check box to enable setting connection address |
| (Optional) | to 0.0.0.0 per RFC2543 when call hold is signaled to |
| | Unified CM. This allows backward compatibility with |
| | endpoints that do not support RFC3264. |
+----------------+------------------------------------------------------------+
| Semi Attended | This check box determines whether the Cisco Unified IP |
| Transfer | Phones 7940 and 7960 caller can transfer an attended |
| (Optional) | transfer's second leg while the call is ringing. Select |
| | the check box if you want semi attended transfer enabled; |
| | leave it clear if you want semi attended transfer disabled.|
| | |
| | Note: |
| | |
| | This check box applies to the Cisco Unified IP Phones |
| | 7941/61/70/71/11 when they are in SRST mode only. |
+----------------+------------------------------------------------------------+
| Enable VAD | Select this check box if you want voice activation |
| (Optional) | detection (VAD) enabled; leave it clear if you want VAD |
| | disabled. When VAD is enabled, no media is sent when voice |
| | is detected. |
+----------------+------------------------------------------------------------+
| Stutter | Select this check box if you want stutter dial tone when |
| Message | the phone goes off hook and a message is waiting. Leave |
| Waiting | clear if you do not want a stutter dial tone when a |
| (Optional) | message is waiting. |
| | |
| | This setting supports Cisco Unified IP Phones 7960 and |
| | 7940 that run SIP. |
+----------------+------------------------------------------------------------+
| MLPP User | Select this check box to enable MLPP User Authorization. |
| Authorization | MLPP User Authorization requires the phone to send in an |
| (Optional) | MLPP username and password. |
+----------------+------------------------------------------------------------+
Table: Normalization Script Tab
.. tabularcolumns:: |p{4cm}|p{11cm}|
+---------------+-----------------------------------------------------------+
| Option | Description |
+===============+===========================================================+
| Normalization | From the drop-down list, choose the script that you |
| Script | want to apply to this SIP profile. |
| | |
| | To import another script from Unified CM, go to the SIP |
| | Normalization Configuration window (Device Device |
| | Settings SIP Normalization Script), and import a new |
| | script. |
+---------------+-----------------------------------------------------------+
| Enable Trace | Select this check box to enable tracing within the script |
| | or clear this check box to disable tracing. When selected,|
| | the trace.output API provided to the Lua scripter produces|
| | SDI trace. |
| | |
| | Note: |
| | |
| | We recommend that you only enable tracing while |
| | debugging a script. Tracing impacts performance and |
| | is not recommended under normal operating conditions. |
+---------------+-----------------------------------------------------------+
| Script | Enter parameter names and parameter values in the |
| Parameters | **Script Parameters** box as comma-delineated key-value |
| | pairs. Valid values include all characters except equals |
| | signs (-), semicolons (;), and nonprintable characters, |
| | such as tabs. You can enter a parameter name with no |
| | value. |
| | |
| | Alternatively, to add another parameter line from Unified |
| | CM, click the + (plus) button. To delete a parameter |
| | line, click the - (minus) button. |
+---------------+-----------------------------------------------------------+
Table: Incoming Requests FROM URI Settings Tab
.. tabularcolumns:: |p{4cm}|p{11cm}|
+--------------+----------------------------------------------------------+
| Option | Description |
+==============+==========================================================+
| Caller ID DN | Enter the pattern that you want to use for calling line |
| | ID, from 0 to 24 digits. For example, in North America: |
| | |
| | - 555XXXX - Variable calling line ID, where X equals an |
| | extension number. The CO appends the number with the |
| | area code if you do not specify it. |
| | - 55000 - Fixed calling line ID, where you want the |
| | Corporate number to be sent instead of the exact |
| | extension from which the call is placed. The CO |
| | appends the number with the area code if you do not |
| | specify it. |
| | |
| | You can also enter the international escape character +. |
+--------------+----------------------------------------------------------+
| Caller Name | Enter a caller name to override the caller name that is |
| | received from the originating SIP Device. |
+--------------+----------------------------------------------------------+
Table: Trunk Specific Configuration Tab
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+------------------------------------------------------------+
| Option | Description |
+================+============================================================+
| Reroute | Unified CM only accepts calls from a SIP device whose IP |
| Incoming | address matches the destination address of the configured |
| Request to new | SIP trunk. In addition, the port on which the SIP message |
| Trunk based on | arrives must match the one that is configured on the SIP |
| | trunk. After Unified CM accepts the call, Unified CM uses |
| | the configuration for this setting to determine whether |
| | to reroute the call to another trunk. |
| | |
| | From the drop-down list, choose the method that |
| | Unified CM uses to identify the SIP trunk where the call |
| | gets rerouted: |
| | |
| | - **Never** - If the SIP trunk matches the IP address of |
| | the originating device, choose this option. Unified |
| | CM, which identifies the trunk by the incoming |
| | packet's source IP address and the signaling port |
| | number, does not route the call to a different (new) |
| | SIP trunk. The call occurs on the SIP trunk on which |
| | the call arrived. |
| | - **Contact Info Header** - If the SIP trunk uses a SIP |
| | proxy, choose this option. Unified CM parses the IP |
| | address or domain name and the signaling port number |
| | in the incoming request's header. Unified CM then |
| | reroutes the call to the SIP trunk using that IP |
| | address and port. If no SIP trunk is identified, the |
| | call occurs on the trunk where the call arrived. |
| | - **Call-Info Header with purpose-x-cisco-origIP** - If |
| | the SIP trunk uses a Customer Voice Portal (CVP) or a |
| | Back-to-Back User Agent (B2BUA), choose this option. |
| | When the incoming request is received, Unified CM |
| | performs the following: |
| | |
| | - parses the Call-Info header |
| | - looks for the parameter ``purpose-x-cisco-origIP`` |
| | - uses the IP address or domain name and signaling |
| | port number in the header to reroute the call to |
| | the SIP trunk using the IP address and port |
| | |
| | If the parameter is not in the header, or no SIP trunk |
| | is identified, the call occurs on the SIP trunk where |
| | the call arrived. |
| | |
| | Default: Never |
| | |
| | Note: |
| | |
| | This setting does not work for SIP trunks connected to: |
| | |
| | - A Unified CM IM and Presence Service proxy server. |
| | - Originating gateways in different Unified CM groups |
+----------------+------------------------------------------------------------+
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-------------------------------------------------------------+
| Option | Description |
+================+=============================================================+
| RSVP Over SIP | This field configures RSVP over SIP trunks. From the |
| | drop-down list, choose the method that Unified CM |
| | uses to configure RSVP over SIP trunks: |
| | |
| | - **Local RSVP** - In a local configuration, RSVP occurs |
| | within each cluster, between the endpoint and the |
| | local SIP trunk, but not on the WAN link between the |
| | clusters. |
| | - **E2E** - In an end-to-end (E2E) configuration, RSVP |
| | occurs on the entire path between the endpoints, |
| | including within the local cluster and over the WAN. |
+----------------+-------------------------------------------------------------+
| Resource | Select a configured Resource Priority Namespace list from |
| Priority | the drop-down menu. The Namespace List is configured in |
| Namespace List | Unified CM in the Resource Priority Namespace List menu. |
| | You can access the menu in Unified CM from System MLPP > |
| | Namespace. |
+----------------+-------------------------------------------------------------+
| Fall back to | Select this check box if you want to allow failed end-to-end|
| local RSVP | RSVP calls to fall back to local RSVP to establish the |
| | call. If this check box is clear, end-to-end RSVP calls that|
| | cannot establish an end-to-end connection fail. |
+----------------+-------------------------------------------------------------+
| SIP Rel1XX | This field configures SIP Rel1XX, which determines |
| Options | whether all SIP provisional responses (other than 100 |
| | Trying messages) are sent reliably to the remote SIP |
| | endpoint. Valid values are: |
| | |
| | - **Disabled** - Disables SIP Rel1XX. |
| | - **Send PRACK if 1XX contains SDP** - Acknowledges a 1XX |
| | message with PRACK, only if the 1XX message contains SDP. |
| | - **Send PRACK for all 1XX messages** - Acknowledges |
| | all1XX messages with PRACK. |
| | |
| | If you set the RSVP Over SIP field to E2E, you cannot |
| | choose Disabled. |
+----------------+-------------------------------------------------------------+
| Video Call | Video Call Traffic Class determines the type of video |
| Traffic Class | endpoint or trunk that the SIP Profile is associated |
| | with. From the drop-down list, select one of: |
| | |
| | - **Immersive** - High-definition immersive video. |
| | - **Desktop** - Standard desktop video. |
| | - **Mixed** - A mix of immersive and desktop video. |
| | |
| | Unified CM Locations Call Admission Control (CAC) |
| | reserves bandwidth from two Locations video bandwidth |
| | pools, Video Bandwidth and Immersive Bandwidth. The pool |
| | used depends on the type of call determined by the Video |
| | Call Traffic Class. Refer to the “Call Admission Control†|
| | chapter of the Cisco Unified Communications Manager |
| | System Guide for more information. |
+----------------+-------------------------------------------------------------+
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-------------------------------------------------------------+
| Option | Description |
+================+=============================================================+
| Calling Line | Select one of: |
| Identification | |
| Presentation | - **Strict From URI presentation Only** - To select the |
| (Mandatory) | network-provided identity |
| | - **Strict Identity Headers presentation Only** - To |
| | select the user-provided identity |
| | - **Default** - To select the system default calling line |
| | identification |
| | |
| | Default: Default |
+----------------+-------------------------------------------------------------+
| Session | Session Timer with Update: The session refresh timer |
| Refresh Method | allows for periodic refresh of SIP sessions. This allows |
| (Mandatory) | the Unified CM and remote agents to determine whether the |
| | SIP session is still active. Prior to Release 10.01, when |
| | the Unified CM received a refresh command, it supported |
| | receiving either Invite or Update SIP requests to refresh |
| | the session. When the Unified CM initiated a refresh, it |
| | supported sending only Invite SIP requests to refresh the |
| | session. With Release 10.01, this feature extends the |
| | refresh capability so that Unified CM can send both |
| | Update and Invite requests. |
| | |
| | Specify whether to use **Invite** or **Update** as the |
| | Session Refresh Method. |
| | |
| | Default: Invite |
| | |
| | Note: |
| | |
| | Sending a midcall Invite request requires specifying |
| | an offer SDP in the request. This means that the far |
| | end must send an answer SDP in the Invite response. |
| | |
| | Update: Unified CM requests a SIP Update if the SIP |
| | session's far end supports the Update method in the |
| | Supported or Require headers. When sending the Update |
| | request, the Unified CM includes an SDP. This |
| | simplifies the session refresh since no SDP offer or |
| | answer exchange is required. |
| | |
| | Note: |
| | |
| | If the far end of the SIP session does not support |
| | the Update method, the Unified CM continues using the |
| | Invite method for session refresh. |
+----------------+-------------------------------------------------------------+
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-------------------------------------------------------------+
| Option | Description |
+================+=============================================================+
| Enable ANAT | This option allows a dual-stack SIP trunk to offer both |
| | IPv4 and IPv6 media. |
| | |
| | Selecting the **Enable ANAT** and **MTP Required** check |
| | boxes sets Unified CM to insert a dual-stack MTP and send an|
| | offer with two m-lines, for IPv4 and IPv6. If a dual- stack |
| | MTP cannot be allocated, Unified CM sends an INVITE without |
| | SDP. |
| | |
| | When you select the **Enable ANAT** check box and the |
| | **Media Termination Point Required** check box is clear, |
| | Unified CM sends an INVITE without SDP. |
| | |
| | When the **Enable ANAT** and **MTP Required** check boxes |
| | are cleared (or when an MTP cannot be allocated), Unified |
| | CM sends an INVITE without SDP. |
| | |
| | When you clear the **Enable ANAT** check box but you |
| | select the **MPT Required** check box, consider the |
| | information, which assumes that an MTP can be allocated: |
| | |
| | - Unified CM sends an IPv4 address in the SDP for SIP |
| | trunks with an IP Addressing Mode of IPv4 Only. |
| | - Unified CM sends an IPv6 address in the SDP for SIP |
| | trunks with an IP Addressing Mode of IPv6 Only. |
| | - For dual-stack SIP trunks, Unified CM determines which |
| | IP address type to send in the SDP based on the |
| | configuration for the IP Addressing Mode Preference |
| | for Media enterprise parameter. |
+----------------+-------------------------------------------------------------+
| Deliver | When checked, the SIP trunk passes the b-number |
| Conference | identifying the conference bridge across the trunk |
| Bridge | instead of changing the b-number to the null value. |
| Identifier | |
| | The terminating side does not require this field. |
| | |
| | Selecting this check box is not required for Open |
| | Recording Architecture (ORA) SIP header enhancements to |
| | the Recording feature to work. |
| | |
| | Selecting this check box allows the recorder to coordinate |
| | recording sessions where the parties are participating in |
| | a conference. |
+----------------+-------------------------------------------------------------+
| Allow | Select this check box to allow passthrough of configured |
| Passthrough of | line device caller information from the SIP trunk. |
| Configured | |
| Line Device | |
| Caller | |
| Information | |
+----------------+-------------------------------------------------------------+
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-------------------------------------------------------------+
| Option | Description |
+================+=============================================================+
| Reject | Select this check box to reject anonymous incoming calls. |
| Anonymous | |
| Incoming Calls | |
+----------------+-------------------------------------------------------------+
| Reject | Select this check box to reject anonymous outgoing calls. |
| Anonymous | |
| Outgoing Calls | |
+----------------+-------------------------------------------------------------+
| Send ILS | When this check box is selected, for calls routed to a |
| Learned | learned directory URI, learned number, or learned pattern, |
| Destination | Unified CM: |
| Route String | |
| | - adds the ``x-cisco-dest-route-string`` header to |
| | outgoing SIP INVITE and SUBSCRIBE messages |
| | - inserts the destination route string into the header |
| | |
| | When this check box is clear, Unified CM does not add the |
| | x-cisco-dest-route-string header to any SIP messages. |
| | |
| | The x-cisco-dest-route-string header allows Unified CM to |
| | route calls across a Session Border Controller. |
+----------------+-------------------------------------------------------------+
Table: Trunk SIP OPTIONS Ping Tab
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-----------------------------------------------------------+
| Option | Description |
+================+===========================================================+
| Enable OPTIONS | Select this check box if you want to enable the SIP |
| Ping to | OPTIONS feature. SIP OPTIONS are requests to the |
| monitor | configured destination address on the SIP trunk. If the |
| destination | remote SIP device is unresponsive or returns a SIP error |
| status for | response such as 503 Service Unavailable or 408 Timeout, |
| Trunks with | Unified CM reroutes the calls by using other trunks or a |
| Service Type | different address. |
| "None | |
| (Default)" | If this check box is clear, the SIP trunk does not track |
| | the status of SIP trunk destinations. |
| | |
| | When this check box is selected, you can configure two |
| | request timers. |
+----------------+-----------------------------------------------------------+
| Ping Interval | This field configures the time duration between SIP |
| for In-service | OPTIONS requests when the remote peer is responding and |
| and Partially | the trunk is marked as In Service. If at least one IP |
| In-service | address is available, the trunk is In Service; if all IP |
| Trunks | addresses are unavailable, the trunk is Out of Service. |
| (seconds) | |
| | Default: 60 seconds |
| | |
| | Range: 5 to 600 seconds |
+----------------+-----------------------------------------------------------+
| Ping Interval | This field configures the time duration between SIP |
| for | OPTIONS requests when the remote peer is not responding |
| Out-of-service | and the trunk is marked as Out of Service. The remote |
| Trunks | peer may be marked as Out of Service if: |
| (seconds) | |
| | - it fails to respond to OPTIONS |
| | - it sends 503 or 408 responses |
| | - the Transport Control Protocol (TCP) connection cannot |
| | be established |
| | |
| | If at least one IP address is available, the trunk is In |
| | Service; if all IP addresses are unavailable, the trunk |
| | is Out of Service. |
| | |
| | Default: 120 seconds |
| | |
| | Range: 5 to 600 seconds |
+----------------+-----------------------------------------------------------+
| Ping Retry | This field specifies the maximum waiting time before |
| Timer (msec) | retransmitting the OPTIONS request. |
| | |
| | Range: 100 to 1000 milliseconds |
| | |
| | Default: 500 milliseconds |
+----------------+-----------------------------------------------------------+
| Ping Retry | This field specifies the number of times that Unified CM |
| Count | resends the OPTIONS request to the remote peer. After the |
| | configured retry attempts are used, the destination is |
| | considered to have failed. To obtain faster failure |
| | detection, keep the retry count low. |
| | |
| | Range: 1 to 10 |
| | |
| | Default: 6 |
+----------------+-----------------------------------------------------------+
Table: Trunk SDP Information Tab
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-----------------------------------------------------------+
| Option | Description |
+================+===========================================================+
| Send | Select this check box to prevent Unified CM from sending |
| send-receive | an INVITE a-inactive SDP message during call hold or media|
| SDP in midcall | break during supplementary services. |
| INVITE | |
| | Note: |
| | |
| | This check box applies only to early offer enabled |
| | SIP trunks and has no impact on SIP line calls. |
| | |
| | When you enable Send send-receive SDP in midcall INVITE |
| | for an early offer SIP trunk in tandem mode, Unified CM |
| | inserts MTP to provide sendrecv SDP when a SIP device |
| | sends offer SDP with a-inactive or sendonly or recvonly |
| | in audio media line. In tandem mode, Unified CM depends |
| | on the SIP devices to reestablish media path by sending |
| | either a delayed INVITE or midcall INVITE with send-recv |
| | SDP. |
| | |
| | When you enable Send send-receive SDP in midcall INVITE |
| | and Require SDP Inactive Exchange for Mid-Call Media |
| | Change on the same SIP Profile, the Send send-receive SDP |
| | in midcall INVITE overrides the Require SDP Inactive |
| | Exchange for Mid-Call Media Change, so Unified CM does |
| | not send an INVITE with a-inactive SDP in midcall codec |
| | updates. For SIP line side calls, the Require SDP |
| | Inactive Exchange for Mid-Call Media Change check box |
| | applies when enabled. |
| | |
| | Note: |
| | |
| | To prevent the SDP mode from being set to inactive in |
| | a multiple-hold scenario, set the Duplex Streaming |
| | Enabled clusterwide service parameter in Unified CM |
| | (System Service Parameters) to True. |
+----------------+-----------------------------------------------------------+
.. tabularcolumns:: |p{4cm}|p{11cm}|
+----------------+-----------------------------------------------------------------+
| Option | Description |
+================+=================================================================+
| Allow | If the check box is selected, Unified CM allows supported SIP |
| Presentation | endpoints to use the Binary Floor Control Protocol (BFCP) |
| Sharing using | to enable presentation sharing. |
| BFCP | |
| | The use of BFCP creates an added media stream in addition |
| | to the existing audio and video streams. This additional |
| | stream is used to stream a presentation, such as a |
| | PowerPoint presentation from someone’s laptop, into a SIP |
| | videophone. |
| | |
| | If the check box is clear, Unified CM rejects BFCP offers |
| | from devices associated with the SIP profile. The BFCP |
| | application line and associated media line ports are set |
| | to 0 in the answering SDP message. |
| | |
| | Default: Clear |
| | |
| | Note: |
| | |
| | BFCP is only supported on SIP networks. BFCP must be |
| | enabled on all SIP trunks, lines, and endpoints for |
| | presentation sharing to work. BFCP is not supported |
| | if the SIP line or SIP trunk uses MTP, RSVP, TRP, or |
| | Transcoder. |
| | |
| | For more information on BFCP, refer to the Cisco Unified |
| | Communications Manager System Guide. |
+----------------+-----------------------------------------------------------------+
| Allow iX | Select this check box to enable support for iX media |
| Application | channel. |
| Media | |
+----------------+-----------------------------------------------------------------+
| Allow multiple | This option applies when incoming SIP signals do not |
| codecs in | indicate support for multiple codec negotiation and |
| answer SDP | Unified CM can finalize the negotiated codec. |
| | |
| | When this check box is selected, the endpoint behind the |
| | trunk can handle multiple codecs in the answer SDP. |
| | |
| | For example, an endpoint that supports multiple codec |
| | negotiation calls the SIP trunk, and Unified CM sends a |
| | Delay Offer request to a trunk. The endpoint behind the |
| | trunk returns all support codecs without the Contact |
| | header to indicate the support of multiple codec |
| | negotiation. |
| | |
| | In this case, Unified CM identifies that the trunk can |
| | handle multiple codec negotiation, and sends SIP response |
| | messages to both endpoints with multiple common codecs. |
| | |
| | When clear, Unified CM identifies that the endpoint |
| | behind the trunk cannot handle multiple codec |
| | negotiation, unless SIP contact header URI states it can. |
| | Unified CM continues the call with single codec |
| | negotiation. |
+----------------+-----------------------------------------------------------------+