# SIP Peer Profile

## Purpose

This SIP Peer Profile form is used to configure SIP trunks with the following:

- the local account information

- the outbound proxy server

- the Calling Line ID information

- the policies applicable to the SIP trunk

- the authentication information

- RTP (voice) stream packet rate

- SIP private networking trunks

- the Outgoing DID Ranges for CPN Substitution

- SIP profile information

NOTE: A SIP peer profile is automatically displayed here each time a new gateway trunk profile is added.

Use this form when performing the following tasks:

- Programming SIP Trunks

- SIP Maintenance Command

- Managing Configuration Profiles

- Program Call Billing for SIP Gateway

The default values in this form are those that are presented when a new SIP Peer Profile is being added. However, for each specific peer, some of those values must be changed. Changes are required for licensing, authentication, and site or peer specific features and requirements. Some defaults date back to when SIP support was first introduced. These defaults have remained largely unchanged to ensure backward compatibility with existing installations as they upgrade. When creating a new Peer Profile, view the recommended option changes (indicated in the following table with a '**' beside the original default value) and perform testing with the recommended - not the default - settings.

NOTE: The recommendations are based on extensive in-house testing but are not guaranteed to work in all installations. You are strongly urged to test your configuration before deploying it to ensure it meets your customer's needs.

## Conditions

A single system can support up to 250 SIP Peer Profiles.

## Field Descriptions

Parameter | Description | Default Value
Basic
SIP Peer Profile Label | Mandatory. Enter an alphanumeric string up to nine characters for the SIP Peer Profile. | Blank
Network Element | Mandatory. Select the appropriate Network Element name (programmed in the Network Elements form) from the pull-down list. | Blank
Basic - Local Account Information
Registration User Name | For Service Providers and SIP Services that require MiVoice Business to register, this field is used to indicate the user names that they want the system to register with. The Registration User Name can be: a single user name a range of DIDs being supplied by the service provider a "bnc" (bulk number contact), to register all DIDs associated with this MiVoice Business system under a single registration name (RFC 6140). The field accepts a maximum of 60 characters (alphanumeric and special characters). The maximum number of alphanumeric characters allowed per user name is 26. If using a range, you must use only digits 0 through 9 with a dash (-) separating the first number in the range and the last. A valid range can represent at the most 1000 user names. Any range greater than 1000 is considered as a user name. For example the entry 6135551001-6135551999 is considered as a valid range and the entry 6135551001-6135551 is considered as a user name. Alphanumeric characters in a range is also considered as a user name. For example, the entry AB67-564 is considered as a user name. You can enter a mix of numerical ranges and single usernames (for example, "6135554000-6135554400, 6135554500"). Use a comma to separate user names and ranges. The first and last characters cannot be a comma or a dash. The field accepts the following special characters; !$%ab&'()*+-_./;=?~. If any other special characters are required, then they need to be escaped. For example, to include #, the escape sequence is %23. For user names that cannot be registered, check with your service provider if any special character needs to be escaped. | Blank
Address Type | Select the address type for the local host. Two types are available: FQDN - the Fully Qualified Domain Name (FQDN) of the local host, and the DNS name from the System IP Properties form. IP - from the System IP Properties form. NOTE: Being consistent in using either FQDNs or IP addresses at both the MiVoice Business and the Service Provider will minimize issues in the configuration and maintenance. Also verify that the Address Type for the peer in the Network Element form matches your selection here. | FQDN
Basic - Administration Options
Interconnect Restriction | Enter the Interconnect Restriction number that is used to restrict device interconnections. | 1
Maximum Simultaneous Calls | Enter the maximum allowable number of incoming and outgoing simultaneous calls for this peer. This value cannot exceed the number of SIP Trunks enabled in the License and Option Selection form. NOTE: Set the maximum number of calls per link the same on both ends of the link. | 2000
Minimum Reserved Call Licenses | Enter the minimum number of call licenses to be reserved for this peer and not to be shared with others. This value cannot exceed the number of Maximum Simultaneous Calls. If this value matches the Maximum Simultaneous Calls, all call licenses are reserved for this peer and none are shared. If this value is 0, all call licenses are shared. See Program SIP Trunks for more information. | 0
Outbound Proxy Server | Select the network element to be used for the Outbound Proxy Server. The Outbound Proxy may be defined as either an IP or an FQDN. If an FQDN is used, it may refer to more than a single Outbound Proxy. This is useful for resilient solutions where in case of failure (or maintenance) the other Outbound Proxy is used. |
SMDR Tag | Enter the SMDR tag number. The range is 1-9998. This tag number is used in SMDR logs for both incoming and outgoing calls when SMDR is enabled. | Blank
Trunk Service | Enter the Trunk Service number (Trunk Attributes form) where the COS/COR and incoming call handling are set. | 1
Zone | Devices are grouped together into a zone numbered from 1 to 999 to which compression and other policies can be applied. The Zone is assigned in the Network Elements form. | 1
Basic - Authentication Options
User Name | Enter the authentication user name (up to 48 characters long) used to authenticate incoming/outgoing calls. The user name is also used to authenticate the registration, if applicable. | Blank
Password | Enter the authentication password associated with the Authentication User Name (hidden). The password is also used to authenticate the registration, if applicable. Any combination of alphanumeric and special characters is permitted, up to 26 characters in length. | Blank
Confirm Password | Re-enter the authentication password to confirm it. | Blank
Authentication Option for Incoming Calls | Select the type of authentication challenges for incoming calls. It is best to use the standard SIP Challenge-Based Authentication, but some providers require that no authentication be performed. The options are: No Authentication - No authentication performed. Challenge - Based Authentication- All incoming calls are challenged with digest authentication utilizing the username and password programmed for this profile. Validate Address - The IP address or FQDN of all incoming calls is validated against the entry programmed in the Network Elements form. | No Authentication
Subscription User Name | Enter the username MiVoice Business uses to subscribe to the MiVoice Border Gateway or other SIP element that is performing KPML digit detection. Any combination of alphanumeric and special characters is permitted, up to 48 characters in length. | *******
Subscription Password | Enter the password MiVoice Business uses to subscribe to the MiVoice Border Gateway or other SIP element that is performing KPML digit detection. Any combination of alphanumeric and special characters is permitted, up to 20 characters in length. | *******
Subscription Confirm Password | Re-enter the password for KPML subscription. | *******
Gateway Options
Digital Trunk Licenses | Read-only. Indicates the total number of E1T1 cards in the group on an EX controller. |
Maximum Digital/Analog Channels | Read-only. Indicates the total number of channels in the E1 (30 channels), T1-ISDN (23 channels), T1-E&M (24 channels), and FXO (1 channel) trunks in the group associated with this profile on an EX controller. |
Call Routing
Alternate Destination Domain Enabled | From the drop-down list, select when to use Alternate Destination Domain: No - not to be used. for Calls -this option will insert the alternate FQDN or IP Address into the To header of outgoing calls. In addition, the FQDN or IP Address may be used to find the appropriate Peer Profile on incoming calls. for Calls and Registration - alternate FQDN or IP Address will be also used for Registration with the service provider. Enter the alternate domain name in the text box bellow. | No
Alternate Destination Domain FQDN or IP Address | Enter the Fully Qualified Domain Name or IP Address of the Alternate Destination Domain. | Blank
Enable Special Re-invite Collision Handling | Select 'Yes' to enable special handling of re-invite collisions. Normally, when a re-invite collision is detected, both re-invite messages are rejected with a 491. With this option enabled, the incoming re-invite wins. | No
Only Allow Outgoing Calls | Enable this option to have the peer allow only outgoing calls (up to the licensed limit), and reject all incoming calls with a 403 Forbidden message for this peer. This option should be enabled for trunks that are only used for outgoing emergency calls. | No
Private SIP Trunk | If set to 'Yes', calls received on this trunk will be considered private or non-public when presented to call control. This option is used when connecting to a SIP service that deals with local (non-public) numbers such as a local SIP voicemail service. When enabled, on outgoing calls, the system does not perform public CPN substitutions, which are used to present a related public number for the local number placing the call. If the call is directed to an ISDN type interface, the calling party number may pass "as is" out to the network if this option is Set to ‘No’. If set to ‘Yes’, the private/non-public number may undergo CPN substitution before being sent out to the network. | No
Reject Incoming Anonymous Calls | Select 'Yes' to prevent the following call types from entering the network on this trunk: Anonymous calls Calls that do contain have a username Calls that contain a username that cannot be understood (i.e. an unknown URI) The system sends a response code (433-Anonymity Disallowed) to rejected callers, alerting them to the fact that they need to change their configuration. | No
Route Call Using P-Called-Party-ID (if present) | When this option is set to 'Yes' all incoming calls will be routed based on the user information present in the P-Called-Party-ID header. If this header does not exist, then the call will be routed using the Request-URI or, if option 'Route Call Using To Header' is enabled, the To header. For most users the recommended setting is 'No' as this header is not often used. Verify with the service provider if routing based on the P-Called-Party-ID header is expected. | Yes **
Route Call Using To Header | Enable this option to route all incoming calls to the called individual user based on the information present in the To URI instead of the Request URI. In most cases, the information in these two lines is the same. When the two lines differ, this option is used to select which one should be used. | No
Calling Line ID
Default CPN | Enter the default CPN up to 26 alphanumeric characters (a-z, A-Z and 0-9). This alphanumeric string is used in outgoing calls to replace the calling party number and on incoming calls to replace the called party number when a match is not found in the URI/Number Translation form or in any of the various Calling Party Number (CPN) Substitution forms: Associated Directory Numbers CPN Substitution DID Ranges for CPN Substitution Normally on outgoing calls (unless restricted or private) the default CPN would appear in the From header if a match is not found elsewhere. If the "Use P-Asserted-Identity Header" option is set, the P-Asserted-Identity will be included and it will contain the Default CPN if a match is not found elsewhere. A "Privacy: id" header will also be included if the CPN is restricted or the number is marked private. If the "Use P-Preferred-Identity Header" option is set to ‘Peer Profile Default CPN’, the P-Preferred-Identity will be included and will contain the Default CPN. For incoming calls, the ringing or answering party's number is replaced with the default CPN when a match is not found in the URI/Number Translation form or in any of the various Calling Party Number (CPN) Substitution forms listed above. This number will only be included in the P-Asserted-Identity header if the option is enabled. To substitute specific numbers to DIDs, you may add numbers into the DID Ranges for CPN Substitution form and then select the appropriate numbers in the Outgoing DID Ranges panel located at the bottom of this form (SIP Peer Profile). Public numbers may also be assigned to users in the User and Services Configuration form as CPN Substitution Numbers, or in their associated Network Zones. NOTE: Public numbers may be substituted by the Default CPN unless the option "Public Calling Party Number Passthrough" is set. | Blank
Default CPN Name | Enter the default CPN name up to 60 alphanumeric characters (a-z, A-Z and 0-9 but not special characters). This alphabetic string is used in outgoing calls to replace the calling party name when a match is not found in the URI/Number Translation form or the Outgoing DID ranges on the SIP Peer Profile Form and the Default CPN is programmed. For example, if the Default CPN Name is "Mitel" and the Default CPN is 6135922122, this name and number will replace the original user's name and number. | Blank
CPN Restriction | Set this option to 'Yes' to force anonymous@anonymous.invalid in the From header on outgoing calls and to prevent CPN substitution on incoming calls. If the "Use P-Asserted-Identity Header" option is set the P-Asserted-Identity will be included and it will contain the calling party number or some substitution. A "Privacy: id" header will also be included in this case to indicate that the identity should be kept private. | No
Override From Header with Default CPN | Set this option to 'YES' to override the From header with the Default CPN on outbound SIP Calls. | No
Public Calling Party Number Passthrough | If an incoming call is received by MiVoice Business through an ISDN trunk or a Public SIP trunk you can allow the public number to be passed through MiVoice Business when it leaves via a SIP trunk. Enable this option to allow the public CPN to be passed through MiVoice Business and not substituted with the default CPN (normal behavior). NOTE: Passing public numbers through MiVoice Business is restricted in some areas. | No
Strip PNI | Select 'Yes' to strip the Primary Node Identifier (as defined in the Cluster Element form) from the calling party numbers leaving just the extension number for use in CPN substitution. | No
Use Diverting Party Number as Calling Party Number | Set this option to 'Yes' to use the diverting/forwarding party as the CPN on the outbound SIP call instead of the original calling party number. This option also applies to outbound EHDU SIP calls, where the EHDU number is used instead of the original calling party number. The party at the final call destination sees the call as being from the diverting/EHDU party because the original calling party information is not provided. NOTE: If Include Diversion Header for EHDU is enabled, it is the EHDU information that will be used as the diverting party for purposes of substituting the calling party number. | No
Use Original Calling Party Number If Available | Applies to incoming external calls with CLI routed out of MiVoice Business. Set this option to 'Yes' to use the original calling number for PRI- or SIP-to-SIP outgoing public calls. In some cases the calling party number may be modified and this option allows the original number to be used. For this option to function properly, the SIP trunk must be public ('Private SIP Trunk' set to No) and option 'Public Calling Party Number Passthrough' must be set to Yes. IMPORTANT: This option does not apply to the CPN presented internally in alarm logs (system and consoles) and Emergency Response Adviser logs (system and SNMP trap). Internal CPN is always the number received, whether it was modified or not. | No
SDP Options
Allow Peer to Use Multiple Active M-Lines | Set this option to 'No' to limit the number of m-lines (media description line) exchanged to 1 with audio as the preferred service. For service providers that prefer a single m-line, which may be either audio or image (for T.38), this option should be set to 'No'. If the incoming call from the trunk will contain more than 1 m-line this option should be set to 'Yes'. For peers capable of negotiating multiple m-lines, such as audio, image, video, and applications, the recommended setting is 'Yes.' NOTE: For SRTP negotiations generally this option is enabled for those that negotiate both SAVP and AVP m-lines. | Yes
Allow Using UPDATE For Early Media Renegotiation | Select 'Yes' to allow the media source to be switched using an Update message prior to the call being answered. NOTE: Enable this option if the SIP trunk supports early media negotiation. See RFC 3311 for details. | No **
Avoid Signaling Hold to the Peer | If this option is set to 'Yes', when a user places a call on hold, MiVoice Business suspends the call but does not send a Hold indication to the peer. SIP-SDP renegotiation is performed using "sendrecv" attributes, and services such as Music on Hold are provided by MiVoice Business itself. If this option is set to 'No', when a user places a call on hold, MiVoice Business suspends the call and sends a Hold indication to the peer. SIP-SDP renegotiation is performed using "sendonly" attributes. If the service provider acts upon the Hold indication that it receives from MiVoice Business, it will suspend the call a second time and be in charge of providing services such as Music on Hold. | Yes
AVP Only Peer | The default setting 'Yes' indicates that the device is to be used for unencrypted media only (either the device is not capable or not configured to use Secure Real-time Protocol [SRTP]). This is the same mode (unencrypted) that trunks used in pre-7.0 releases. If you want the SIP trunks to allow SRTP encryption, set this option to "No" and option Allow Peer to Use Multiple Active M-Lines to "Yes". This is necessary because many SIP implementations use two or more m-lines to negotiate either SRTP or non-SRTP (one m-line for each). The System Option "Voice/Video SRTP Encryption Enabled" is also required to be set for SRTP to be negotiated. If this option is set to "No", calls may negotiate as either SRTP or non-SRTP depending on the capabilities of the devices that are being connected. | Yes
Enable Mitel Proprietary SDP | Select 'Yes' only if communicating with a MiVoice Border Gateway. This enables proprietary encryption to be negotiated if SRTP is not in use. | No
Force sending SDP in initial Invite message | Select 'Yes' to always insert SDP into the initial Invite message. This option allows inter-working with SIP peers that require SDP in the Initial Invite. In some situations, MiVoice Business already includes SDP in the initial invite. The forced SDP may contain a default ‘inactive’ SDP if the SDP is not available at call setup time. If the real SDP is not available, a renegotiation is necessary to get media working. | No
Force sending SDP in Initial Invite - Early Answer | Select 'Yes' if the Initial Outgoing Invite must contain a "sendrecv" Session Description Protocol (SDP) message that identifies the IP address/port of the calling device. This option takes precedence over the “Force sending SDP in initial Invite message” option. MiVoice Business provides the connection information on the outgoing call by pretending the call was answered before it ever left the controller. Avoid this option if at all possible. In some call scenarios (EHDU and call forwarding) this option may not work at all. | No
Ignore SDP Answers in Provisional Responses | Select "Yes" to improve interoperability with SIP systems that adhere to RFC 3960 and do not always provide audio when sending SDP information in the 18x messages. As a result, the device placing the call may hear silence. With this option enabled, the SDP Answer is ignored during ringback and local ringback tone is provided to the device placing the call. Once the call is answered the media is connected. A side effect of this option being enabled is that if real cut-through audio is available it will not be played – just the locally generated ringback. This option may also be used in cases where you are connecting an external incoming and outgoing call together. Without SIP PRACK, early media cannot be reconnected and the caller does not hear cut-through audio from the new outgoing call. With this option enabled, at least local ringback can be provided when PRACK is not an option for the provider. | No
IP Media Default | Select the addressing type for SDP. Supported options are: ipv4 ipv6 dual-anat dual-altc If the SIP peer supports dual media (IPv4 and IPv6), you can choose one of the following methods to provide the SDP : If your SIP peer supports RFC 4091 or RFC 4092, then select dual-anat (Alternative Network Address Type). If your SIP peer supports RFC 6947, then select dual-altc (Alternate Connectivity Attribute) | ipv4
Limit to one Offer/Answer per INVITE | Most vendors expect just one Offer/Answer exchange in the initial Invite transaction. If the device can begin a second offer in the 200ok and expects a response, this option should be set to 'No'. | Yes
NAT Keepalive | Select to send UDP packets every 30 seconds to the Peer’s Audio IP Address and port to keep a pinhole open on a NAT firewall. The packets are sent whether the connection is one-way or two-way. | Yes
Prevent Codec Selection on Answer | Select 'Yes' to prevent the MiVoice Business from making a codec selection on a list of codecs in the received SDP before sending an SDP Answer. Select 'No' to allow the MiVoice Business to apply a default policy to make an audio/image codec selection before sending an SDP Answer. | No
Prevent the Use of IP Address 0.0.0.0 in SDP Messages | IP Address 0.0.0.0 is the deprecated indicator of hold. When this option is set to 'Yes', the media signaling includes the direction indicators, such as ‘sendonly’, ‘inactive’, etc., as well as the real IP address. When this option is set to 'No', in some cases it can prevent audio from being available when prompts or music-on-hold is being played to specific devices which require knowledge of the transmitters IP address. | Yes
Reject Call without telephone-event payload | Select 'Yes' to reject an incoming call with a 488 message if no telephone-event payload (RFC 4733) is received. When the 488 message is received, the Service Provider equipment should re-send the Invite with a telephone-event payload and the transcode digits received in-band into telephone events using RFC 4733. You must enable this option only with Service Providers that support this feature. Otherwise, this might result in dropped calls to customers. | No
Renegotiate SDP To Enforce Symmetric Codec | Select 'Yes' to force the use of the same codec--for example, G.729--in both incoming and outgoing directions. When an SDP Answer is received with a list of codecs with a different ordering, it can be unclear which codec will be used in either direction. If MiVoice Business detects this situation, this option will send a Re-invite with a single codec to ensure that there is no misunderstanding. | No
Repeat SDP Answer If Duplicate Offer Is Received | Select 'Yes' to treat identical SDP offers received in the same session as a session refresh instead of renegotiating media. This is the RFC recommendation. Devices should increase the session-version when modification is made to SDP session data. The default is still 'No' because not all vendors adhere to this recommendation and usually this is the safer option. The interop process should detect whether the device increments the session-version as expected or not, and set this option accordingly. The drawbacks to renegotiating on a duplicate offer is the restart of RTP streaming and possible change of the media attributes, which may not be expected by the peer. | No
Restrict Audio Codec | This option controls which codec/s can be negotiated over SIP trunks. Calls that use any other codec will be rejected. The recommended setting is 'No Restriction' (default), which means any codec is allowed and none removed. NOTE: This option is ignored for Emergency calls, which are always allowed to go through with the preferred codec. | No Restriction
RTP Packetization Rate Override | Enable this option if the Service Provider to which this peer is connected requires a packet rate other than the standard 20 ms rate in both the transmit and receive media streams. Most Mitel devices and applications now support variable packetization rates. NOTE: Disabling this option allows the system to operate at ANY packetization rate. It does not force it to operate at the greyed-out rate in the RTP Packetization Rate list box. | No
RTP Packetization Rate | If the "RTP Packetization Rate Override" option has been set to 'Yes', the SIP Module will force all calls involving this trunk to use the packetization rate specified in the "RTP Packetization Rate" option. The calls will be forced to use the specified rate in both the transmit and receive streams. Rejected calls or audio problems resulting from an unsuccessful packet rate negotiation will generate Media Negotiation maintenance logs. For more information, see Maintaining the SIP Interface. | 20 ms
Special handling of Offers in 2XX responses (INVITE) | Set to 'Yes' to have MiVoice Business treat 2XX " sendonly" offers as " sendrecv" offers, causing a data stream to be opened to the IP address and port specified in the sendonly offer. For this option to work, the incoming messages must have a valid IP address and port number (not zeroes). This option addresses an interoperability problem with some SIP gateways, whereby unending message negotiation causes an infinite loop. NOTE: Because this option violates the specification, use it with care. | No
Suppress Use of SDP Inactive Media Streams | Select 'Yes' to allow MiVoice Business to minimize the use of inactive SDP messaging if it is not fully supported by the service provider. While media connections are transitioning (hold/transfer/conference, etc.), or in some cases at call setup, the media may be temporarily unavailable causing MiVoice Business to send an inactive SDP. If this option is set to 'Yes' MiVoice Business will instead attempt to use the IP 0.0.0.0 or mark streams as sendonly/recvonly to avoid the use of inactive which is not supported by some SIP Peers. | No
Signaling and Header Manipulation
Trunk Group Label | Enter the trunk group label to insert into the Contact header of SIP URIs. The "tgrp" tag is defined in RFC 4904. | Blank
Allow Display Update | Select 'Yes' to allow the system to signal display updates containing connected party name and number information during transfers, conferences, and call forwarding scenarios. The information is sent in Update messages either using the method as defined in RFC 4916 or in the P-Asserted-Identity header. The option "Use P-Asserted Identity Header" must be set to 'Yes' if the second method is being used. NOTE: A display update can update the display on any type of device (not just SIP phones). | No
Build Contact Using Request URI Address | Enable this option to construct the contact address in 180 and 200 messages using the Request URI Address received in the initial invite. This option should only be enabled in those situations where the Contact needs to be based on the address received in the Request URI. | No
De-register Using Contact Address not "*" | Select 'Yes' to de-register with a precise contact value rather than *. See RFC 3261 for details. | Yes
Disable Reliable Provisional Responses | Select 'Yes' to disable the use of reliable provisional responses (PRACK) on outgoing and incoming calls, unless the Required Header is received on incoming calls. An incoming call that uses the Require header may negotiate PRACK despite this option being enabled. Most Peers now support PRACK and this can be useful in interoperability scenarios with the PSTN (see RFC 3262). If the SIP Peer also supports PRACK, it is recommended that this option be set to 'No'. | No
Disable Use of User-Agent and Server Headers | Select 'Yes' to disable the insertion of "User-Agent" headers in Request and "Server" headers in Response messages. This option should be enabled only if the product has known deficiencies which place the system at risk of attack. | No
Discard Received P-Asserted-Identity Headers | Select 'Yes' to prevent the use of the information received in P-Asserted-Identity header as display updates. Service Providers may use the P-Asserted-Identity header for different purposes. | No
Domain for Trunk Context | If Trunk Group Label is used in the Contact header of SIP URI messages, you may enter an alternate domain name of the terminating switch to be included in the trunk-context field. For more information, see RFC 4904. | Blank
Emergency Call Headers | Select the required option from the drop down to configure emergency call scenarios. Available options are: CESID in From, [and PAI] :- Includes the CESID in the P-Asserted-Identity. Allow Privacy :- The P-Asserted-Identity header must be sent and this includes the CESID. CESID in PAI, Use E911 Headers :- Includes additional custom E911 headers and optional Geolocation headers used by the service providers for handling the emergency calls. CESID in From, Callback in PAI :- The P-Asserted-Identity header includes a Callback number and the "From" is usually used to convey the Location or CESID information. | CESID in From, [and PAI]
E.164: Enable sending '+' | Select 'Yes' to enable adding '+' to the Called and Calling Party Numbers generated by MiVoice Business. | No
E.164: Add '+' if digit length > N digits | Adds '+' only when digit string length exceeds the value specified. For example, to omit the '+' on any digit string less than or equal to 10 digits, enter 10. Can only be changed when "E.164: Enable sending '+'" is set to 'Yes'. | 0
E.164: Do not add '+' to Emergency Called Party | Select 'Yes' to omit the '+' from emergency numbers regardless of the length of the called party number. Can only be changed when "E.164: Enable sending '+'" is set to 'Yes'. | No
E.164: Do not add '+' to Called Party | Select 'Yes' to omit the '+' on all called party numbers, including emergency calls and thereby overriding the preceding option. Can only be changed when "E.164: Enable sending '+'" is set to 'Yes'. | No
Force Max-Forward: 70 on Outgoing Calls | Select 'Yes' to force a value of 70 in the Max-Forward header of outgoing calls. Select 'No' to enable the system to set the Max-Forward header value, which will be 26 or less depending on whether the call has gone through previous hops. Setting this to ‘Yes’ opens the possibility of call loops and should be used only when required. | No
If TLS use 'sips:' Scheme | Select Yes to use the "sips:" (Secure SIP) scheme to signal secure (TLS) SIP messaging. Select No to use the "sip:" signaling scheme, where security is indicated by the "transport=TLS" parameter. | No
Ignore Incoming Loose Routing Indication | Select 'Yes' for MiVoice Business to ignore the loose routing indicator and use strict routing instead. Enabling this option is not recommended unless required by the SIP Peer. See RFC 3261 for more information on Loose Routing. | No
Multilingual Name Display | Select 'Yes' to enable the Multilingual Name Display feature for outgoing SIP trunk calls. This option takes effect only if the 'Multilingual Name Display' System Option is enabled. Selecting 'No' for this option does not disable the feature for the whole system - only for outbound SIP trunk calls. | No
Only use SDP to decide 180 or 183 | Enable this option to force MiVoice Business when alerting over SIP to send a 183 message if an SDP body is available. Otherwise, MiVoice Business sends a 180 message without SDP. Select ‘No’ to use a more complex algorithm based on call signaling to determine whether to send a 180 or 183 message. | Yes
Include Diversion Header for EHDU | This option applies to external-to-external calls, for example an external call to an EHDU. If enabled (set to 'Yes'), the diverting/forwarding party's DID number is inserted in the Diversion header when the call leaves MiVoice Business via a SIP trunk. NOTE: If Include Diversion Header for EHDU is enabled, see the associated note for that option. | No
Mode for Out-of-Band DTMF | Enables delivering Out-of-Band DTMF from the MiVoice Business system to a SIP service without interrupting audio playback. Select RFC 4733 DTMF to deliver Out-of-Band DTMF in the audio playback, using RFC 4733 formatted digits. This mode causes interruptions in the audio playback. Select SIP INFO dtmf-relay to deliver Out-of-Band DTMF through SIP INFO messages in the application/dtmf-relay content-body. This mode does not cause interruptions in the audio playback. | RFC 4733 DTMF
Prefer From Header for Caller ID | Select 'Yes' to give preference to the From header over the P-Asserted-Identity and the P-Preferred-Identity headers for providing the calling party ID for incoming calls. With this option enabled, the user portion of the From header will be used to identify the calling party. If the user portion is anonymous or missing from the header, the system will attempt to extract the calling party ID from either the P-Asserted-Identity header (1st choice) and then from the P-Preferred-Identity header (2nd choice). | No
Q.850 Reason Headers | Select Yes to include the Q.850 cause code in the Reason Header field of the CANCEL, BYE, 3xx, 4xx, 5xx, and 6xx SIP messages. See RFC 3326 and RFC 6432. | No
Require Reliable Provisional Responses on Outgoing Calls | The use of reliable provisional responses (PRACK) on outgoing calls is recommended for early-media negotiations. For more information, see RFC 3262. NOTE: Select 'No' for SIP peers that do not support the PRACK method. This option has no effect if it is enabled and the "Disable Reliable Provisional Responses" is also enabled. | Yes
Signal Privacy (if enabled) on Emergency Calls | Select 'Yes' if you wish the CESID number to be included only in the P-Asserted-Identity header and not in the From header when an emergency call is made from an extension with privacy enabled. With this option set to 'Yes' and privacy enabled, the From: header will have "Private" name and "anonymous" user instead of the users name and CESID number. ATTENTION: If your system is unable to extract CESID data from the P-Asserted Identity header, DO NOT enable this option. Otherwise, emergency response teams will not receive the necessary information. | No
Suppress Incoming Name | Select 'All' to suppress all names provided by the service provider. If you select "E.164", then names containing digits and + character are suppressed. If you select "No", then all names provided by the service provider are displayed. | No
Suppress Redirection Headers | Provides inter-operability with service providers that do not want the redirection header data for calls forwarded or transferred to them by MiVoice Business. Select one of the following: All - Omits redirection headers (Diversion and History-Info Headers for Call Forwarding and Referred-By Header for Transfer) in call setup messages. History-Info - Omits History-Info Header in call setup messages. Diversion - Omits Diversion Header in call setup messages. History-Info/Diversion - Omits Diversion and History-Info Headers in call setup messages. Referred-By - Omits Referred-By Header for Transfer in call setup messages. | No
Use Fixed Retry Time for 491 | Enable this option if you are experiencing audio delays due to the default retry time in a ReInvite collision 491 response. If set to 'Yes', this option replaces the default retry times (0-2 or 2-4 seconds, as specified in RFC 3261) with a fixed retry time of 100 ms, after the 491 response. | No
Use Privacy: none | Select 'Yes' to include the Privacy: none header in outgoing Invite messages. This option may be used by service providers to override an existing privacy setting and allow display information to be presented to the remote party. See RFC 3323 for more information on privacy settings. | No
Use P-Asserted Identity Header | Select 'Yes' to enable sending the P-Asserted Identity header within SIP messages. This option is used to convey identity information both at call setup and during a call. When the 200 OK registration message is sent, this header will include the Directory Name (as configured in the User and Services Configuration form) associated with a set, which can be displayed on SIP devices. When privacy is enabled, or CPN is restricted, this field will still contain calling/called party information. MiVoice Business relies on the use of the “Privacy: id” header to inform other SIP Peers that this information is to be kept private. If the peer is not trusted, select 'No' for this option. NOTE: If 'Signal Privacy (if enabled) on Emergency Calls' is set to Yes, the P-Asserted Identity header will be sent even if this field is set to No. | Yes
Use P-Asserted Identity for Billing | Select 'Yes' to enable the P-Asserted-Identity (PAI) header within SIP messages to contain the correct billing number, based on the algorithm described in Call Billing for SIP Gateway. This option overrides the 'Use P-Asserted Identity Header' and in addition to billing information, continues to use PAI header for sending display updates. NOTE: If 'Signal Privacy (if enabled) on Emergency Calls' is set to Yes, the P-Asserted Identity header will be sent even if this field is set to No. | No
Use P-Call-Leg-ID Header | Select "Yes" to include the Call Leg Call ID within the SIP trunk messages exchanged between MiVoice Business and third-party applications. This ID is always included in MiVoice Business-based MiTAI events, so enabling this field allows for SIP sessions to be aligned with MiTAI events. A new ID will be generated for each call. The header will be in the Invite message - for outgoing calls. For incoming calls, it will be in the 18x message and, if answered, in the 200 OK message OR in a clearing message, such as 3xx, 4xx, 5xx, 6xx (not authentication 401/407). | No
Use P-Early-Media Header | Early Media is the ability of two user agents to exchange messages before a call is actually established. If connected to SIP trunks that use RFC 5009, you can enable this option to indicate which directions (send and receive or send only) are authorized for early-media on inbound calls to MiVoice Business. If the service provider wants the MiVoice Business to authorize media in both directions the ‘sendrecv’ value should be selected. The service provider could request ‘sendonly’ or ‘inactive’ as well. Normally, we expect service providers that use RFC 5009 to request ‘sendrecv’ authorization. Once the call is answered the early-media limitation is lifted and the call continues fully connected. | No
Use P-Preferred Identity Header | From the drop-down list, select one of the following options:No - this header will not be used. Peer Profile Default CPN - the system uses the Default CPN data (name and number) to build the P-Preferred-Identity header. One purpose of this selection is to provide a company or link/peer number. For example, the From header may include the DID of the person making the call but the P-Preferred-Identity header may contain the main company DID. User Associated Billing - for calls initiated by an extension in the cluster, the User Billing Number configured for that extension will be provided in this header. For public trunk incoming calls diverted by an extension for example, this header will include the original public trunk CPN. If these numbers are not available, the Default CPN programmed in this form will be used. No name is provided for this option. NOTE: Since the information included in this header is not marked private, it is up to the service provider to ensure that the User Associated Billing information is used for internal purposes only (such as billing) and not be displayed to end-users. Otherwise, the "User Associated Billing" option should not be selected. | No
Use Restricted Character Set for Authentication | Select 'Yes' to use the restricted character set "0123456789abcdef" for creating nonce and cnonce strings used in authentication. This setting is used for SIP Servers with limited ascii character support and reduced authentication security. | No
Use To Address in From Header on Outgoing Calls | When this option is enabled (set to 'Yes') the Address used in the SIP From header line will no longer be the physical MiVoice Business IP address or FQDN. Instead the address will be replaced by the address to which the outgoing call is sent. Some providers require this for authentication purposes as it makes it look like MiVoice Business is in the same domain as the SIP Server or SBC (Session Border Controller). NOTE: This option does not work with the 'Alternate Destination Domain FQDN or IP address' option. When the 'Alternate Destination Domain FQDN or IP address' option is enabled, the From Header continues to use the FQDN specified for this peer in the Network Elements form. | No
Use user=phone | Select 'Yes' to insert the "user=phone" parameter into SIP URIs. | No
Use user=phone for Diversion Header | Select 'Yes' to insert the "user=phone" parameter into SIP Diversion Header -- if option "Use user=phone" is set to 'No'. This option is available only if option "Use user=phone" is set 'No'. Otherwise, this option is disabled. | No
User-Defined Header Name | Enter the user-defined header name. Maximum characters allowed is up to 32 characters. | Blank
User-Defined Header Value | Enter the user defined header value. Maximum characters allowed is up to 128 characters. | Blank
Timers
Invite Ringing Response Timer | Enter a value between "1" and "30" seconds to define the time the system waits to receive a ringing response on a SIP Trunk before it cancels and redirects the call using alternate routing. Some trunk types may require several seconds before ringback is achieved so a timeout value of 4 to 10 seconds may be required. Enabling this timer allows calls to be rerouted at the time of network failures instead of waiting for the link to be considered totally out of service. Also, in times of severe congestion calls can be also rerouted. To determine what value should be used, test and look for worst case ringback scenarios. For example: How long do international calls take to receive ringback? The value needs to be higher than your worst case time so that all normal calls can be completed successfully on the SIP Trunk. | 0 (disabled)
Keep-Alive (OPTIONS) Period | Enter a value between "60" and "99999" seconds, in increments of 1 second, to define the keep-alive period. When this timer expires, the system sends an OPTIONS message to an idle link in order to maintain or establish the connection between MiVoice Business and the peer. To disable this timer, set "SIP Peer Status" as "Always Active" in the Network Elements form. | 120
Registration Period | Enter a value between "120" and "99999" seconds, in increments of 1 second, to define the initial period for SIP trunk registrations. | 3600
Registration Period Refresh (%) | Enter a value between "50" and "99" percent to define the registration refreshment interval. When this percentage of the Registration Period is reached, the registration is refreshed. | 50
Registration Maximum Timeout | Enter a value between "60" and "3600" seconds, in increments of 1 second, to define the maximum amount of time the system waits before attempting to re-register a failed connection. Re-registration is initially attempted after 30 seconds. This value is doubled each time re-registration fails until the Maximum Timeout value is reached. The system then attempts to re-register at the Maximum Timeout value. | 90
Session Timer | Set the SIP session timeout value (in seconds). If the peer does not respond within the allocated time, the session (call) will be torn down. The default is 90. The range is from 90-9999. Setting the Session Timer to 0 disables session timeout. It is recommended that this be set to a non-zero value unless there is a specific reason to disable it. The benefit of this option is that it can help clear calls that get stuck due to signaling errors or loss of connection. If the peer responds with a 422 “Session Interval too Small,” you may want to increase the session timer to the value indicated by the peer to minimize delays in call setup. The Session Timer will use the UPDATE message if it is supported by the peer. If not supported we will use ReINVITE messages as described in RFC 4028. The Session Timer refresh is sent at half the Session Timer value. For example, if the value is 120 seconds, then the messaging (UPDATE or ReINVITE) will occur every 60 seconds. To limit extra messaging a larger value (e.g. 3600 seconds) can be used for the Session Timer to eventually cleanup any possible stuck resources. | 90
Session Timer: Local as Refresher | Selecting 'Yes' sets MiVoice Business (local), instead of the Peer, as the Session Timer Refresher (in accordance with RFC 4028). Session timer refresh information will be included in Reinvite messages sent by MiVoice Business, as well as in Updates. | No
Subscription Period | Enter a value between "120" and "99999" seconds, in increments of 1 second, to define the initial period for outgoing SIP subscriptions. | 3600
Subscription Period Minimum | Enter a value between "120" and "3600" seconds, in increments of 1 second, to define the minimum amount of time that the system will allow for incoming SIP subscriptions. | 300
Subscription Period Refresh (%) | Enter a value between "50" and "99" percent to define the outgoing subscription refreshment interval. When this percentage of the Subscription Period is reached, the subscription is refreshed. | 80
Key Press Event Options
Allow Inc Subscriptions for Local Digit Monitoring | Select 'Yes' to allow incoming KPML (RFC 4730) subscriptions. Disable this option to reject requests for KPML subscriptions. | No
Allow Out Subscriptions for Remote Digit Monitoring | Select ‘Yes’ if an outbound proxy such as the MiVoice Border Gateway or session border controller is expected to perform outgoing KPML digit collection. Disable this option if the endpoint does not support KPML digit collection Enabling this option activates the KPML Transport and Port fields (see below). | No
Force Out Subscriptions for Remote Digit Monitoring | Select ‘Yes’ to force MiVoice Business to request outgoing KPML digit collection on a connection from endpoints that may wish to inject mid-call digits. | No
Request Outbound Proxy to Handle Out Subscriptions | Use this option to specify which network element performs outgoing KPML digit collection. Select 'Yes' for the outbound proxy (i.e. the MiVoice Border Gateway). Select 'No' for the session border controller (SBC). | No
KPML Transport | If you are using an outbound proxy for digit collection, select the transport used for KPML subscriptions: Default (UDP)/TCP/TLS. NOTE: Because this method of digit collection requires a separate subscription dialog, this setting can differ from the Outbound Proxy Transport programmed for the outbound proxy on the Network Elements form. | Default (UDP)
KPML Port | If you are using an outbound proxy for digit collection, enter the port number used for KPML subscriptions. NOTE: Because this method of digit collection requires a separate subscription dialog, this setting can differ from the Outbound Proxy Port programmed for the outbound proxy on the Network Elements form. | 0
Outgoing DID Ranges
Add Member | To add an Outgoing DID range, first you must program substitutions in the DID Ranges for CPN Substitution form. Then, in the SIP Peer Profile form, select a link, select the Outgoing DID Ranges tab, click Add Member, select the Index number for the substitutions, and then click Save. If a DID number sent over this link matches the DID range programmed for this index number, then the system makes the substitution. NOTE: You can only add members to an existing link. |
Delete Member | To delete a member, select a link, select the Outgoing DID Ranges tab, select the member, click Delete Member, and then click OK. |
Index | Display-only, protected field. Each index number represents a unique inward dialing modification rule. The rules are applied in consecutive order; for example, rule 1 is applied before rule 2. |
DID Range | Display-only, protected field. Enter one or more DID numbers (maximum 60 characters). You can enter a mix of ranges and single numbers (for example, "4000-4400, 4500"). Use a comma (,) to separate DID numbers and ranges; use a dash (-) to indicate a range of DID numbers. The first and last characters in the field cannot be a comma or dash. | Blank
CPN Substitution | Display-only, protected field. Enter the CPN number you want to substitute for any DID in the DID Range field. The CPN Substitution field can contain up to 14 characters. You can use up to 7 'x's ('overlay' characters) to indicate digits that are not affected during substitution. The number of overlay characters in the field must not exceed the number of digits in the DID number. Overlay characters must be contiguous and trailing all digits. For example, to substitute "6135922122" for "5552122", enter "613592xxxx". | Blank
Profile Information
Creator | Enter the name of the person responsible for creating the SIP Peer Profile. For a new profile, this field defaults to "Local." | Local
Date Created | Read-only field. Displays the creation date of the SIP Peer Profile. For a new profile, this field defaults to the current date. | MM DD YY
Created with Version | Read-only field. Displays the MiVoice Business software version number of the SIP Peer Profile. For a new profile, this field defaults to the current software version. |
Service Provider | Enter the name of the service provider that is supplying the SIP trunk. | Blank
Vendor Notes | Enter additional information concerning the service provider. For example, enter the release date and version number of the vendor's software. | Blank