Add ConnecttoTeams to your White List (trusted traffic) Whitelist all core-infrastructure IP addresses. Core Infrastructure ITEM IP ADDRESS PROTOCOL API Gateway 52.184.185.243 HTTPS (Reverse Proxy) Edge Proxy (SIP) 13.68.254.158 TCP/UDP/TLS Edge Proxy (SIP) 20.115.6.243 TCP/UDP/TLS Edge Proxy (SIP) 52.255.184.52 TCP/UDP/TLS Edge Proxy (SIP) 13.68.233.171 TCP/UDP/TLS Edge Proxy (SIP) 20.59.120.227 TCP/UDP/TLS Edge Proxy (SIP) 20.59.124.13 TCP/UDP/TLS Edge Proxy (SIP) 20.124.236.139 TCP/UDP/TLS Edge Proxy (SIP) 40.117.126.112 TCP/UDP/TLS Edge Proxy (SIP) 40.117.126.110 TCP/UDP/TLS Edge Proxy (SIP) 23.29.127.82 TCP/UDP/TLS Port Ranges RTP: 16000-16999. SIP Signalling: 5060.

Overview ConnecttoTeams provides you various options for values to use for Caller ID, Caller Name and PAI on your outbound calls and forwarded calls. The ability to have this granular control for CallerID and Caller Name fields initially was made available on May 21, 2023 as part of ConnecttoTeams portals version 2.7.0. The ability to control the PAI field was added later on February 3, 2024 as part of ConnecttoTeams portals version 2.12. The table below presents a definition of each control and their applicability. Attribute Definition Configurable in Outbound PBX Calls Forwarded PBX Calls Outbound Trunk Calls Forwarded Trunk Calls Caller Name The caller’s display name for a SIP call. (This is not to be confused with the CNAM service for PSTN calls which is a telco provider’s service.) Caller ID The caller’s phone number.     P-Asserted-Identity (PAI) The “From” header does

workspace ID, Slack users’ ID, and profile information (name and email) and keeps them in its database. Teamslink only uses this information to route messages between different platforms and provide external contacts with their connected Slack users’ contact details. Log Data The Teamslink servers automatically record a log entry for each message they process. The log entry contains only the metadata without the message content. The metadata consists of the following fields: Sender address (e.g., [email protected]) Receiver address (e.g., [email protected]) Message type (IM, Presence, typing, error) Time and date of the message Chat session ID Management Using the ConnecttoTeams Provisioning Portal, you can seamlessly connect different collaboration platforms within your company, or partners such as customers, partners, or suppliers outside your company. The ConnecttoTeams Portals provides customers with trailing 12 months of charts and graphs depicting the number of unique users, the number of

accessible from the drop-down on the upper left portion of the page. The numbers are prefixed with Personal and Group to identify the phone number type. The chosen SMS CallerID Number is the FROM number for outgoing SMS/MMS messages. When you are assigned several SMS numbers, be sure to choose the right CallerID to send messages to your intended recipients. Recent Conversations This lists the parties you have conversed with in the last 120 days. These could be individual phone numbers, corresponding contact names, or text group names. Main Chat Dialog Screen The Main Chat screen displays the recipient's numbers or saved contact names, the SMS/MMS conversation (including inbound and outbound messages), and a message box for composing a new message. Note You will only see the messages of the selected SMS CallerID Number in the drop-down menu. Sending SMS/MMS To send an SMS

to attach the encryption key. If negotiation of the encryption data is successful, the call is established. Typically, 'a=crypto' is included in secured media (RTP/SAVP). However, there is also support for including 'a=crypto' in non-secured media (RTP/AVP). In such cases, the media is handled as if the device received two identical media: one secured and one not. SRTP supports the following cipher suites (all other suites are ignored): AES_CM_128_HMAC_SHA1_32 AES_CM_128_HMAC_SHA1_80 AES_256_CM_HMAC_SHA1_32 (RFC 6188) AES_256_CM_HMAC_SHA1_80 (RFC 6188) Note: due to the additional cipher-suite data added to the SDP, enabling SRTP may cause SIP messages to exceed the 1500-byte UDP packet size, resulting in packet fragmentation and failed calls. It may be necessary to change the PBX transport to TCP to ensure successful calling. Note

the did_list, provide value 1 if choosing the second number as the Default DID. The other 2 numbers will be considered as Forwarded DIDs. * If there is only 1 number in the did_list, then provide 0 value in this field as the single DID is to become the Default DID and there will be no Forwarded DIDs. did_list : Could be a single phone number or comma-separated list of DIDs (such as 12125551234,12125551235,12125551236 ) which is used as Default DID or Forwarded DIDs. See description of above-mentioned field default_did_index_number to know how to indicate which is the Default DID. caller_name : The caller’s display name for a SIP call. (This is not to be confused with the CNAM service for PSTN calls which is a telco provider’s service.) caller_name_on_forwarded_calls caller_id : The caller’s phone number

sent by Teams as CallerID, without manipulation. If “Transparent” is chosen, the P-Asserted-Identity (PAI) of the forwarded call will contain the SIP User ID of the Teams User that is forwarding the call. The PBX must be able to identify and authenticate the call based on the PAI header. User’s Caller ID - This is the phone number assigned to the user in O365. (This was the behavior prior to the ConnecttoTeams   2.12 release.) Select the Assign Default DialPlan Policy checkbox to assign the Reseller-defined DialPlan policy to the user. SIP Authentication ID This is sometimes the same as the beginning of the SIP User, but not always. Input the SIP Password (twice to confirm). When you have set up an end-user portal that would be embedded into Microsoft Teams, enter the Portal Login Name and Portal Password for the User. This is optional . Click