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Call Detail Records for your calls can be found in the Call Records section beneath the Elastic SIP Trunking category in the navigation menu.     Here you can search through records of the calls made across your Elastic SIP Trunks. Search filters are available to let you narrow down your search by From, To, Direction, Date and time, as well as the call Result. Additionally clicking into the record will give you the ability to download a packet-capture (PCAP) of the call.  Currently PCAPs are not available for Registered calls, or calls using encryption (TLS).  After you have located the records you are searching for, they can be easily exported into a .csv file for further use.   Notable Call Record Fields TIMESTAMP – The time the Invite was first received by our system FROM – The number or SIP URI of the Calling Party TO – The number or SIP URI of the Called Party TRUNKNAME – The friendly name of the SIP Trunk involved in the call DIRECTION – The direction of the call Inbound – An Inbound call is one that the Elastic SIP Trunking platform has sent to the customer Outbound – An Outbound call is a call which the customer has sent to us for termination to the Public Switched Telephone Network DURATION – The length of time the call was connected for voice conversation. RESULT – The outcome of the call. Completed – The call completed successfully without any problem. The call was connected through with duration. Busy – The call was routed to the end party, but we received back a SIP 486 Busy Here response. Failed – The call was unable to be completed. This could be due to system error, network congestion, or simply because the number was misdialed. Cancel – The call was routed to the user but was rejected with something other than a User Busy. No Answer – The caller hung up before the call was answered. COST – The estimated charges for the call. Note: This is for informational purposes only. The value here does not include any taxes or discounts, or additional service add-ons such as CNAM.   You can also export all records to a .csv file by clicking the Export button.   The /calls REST API endpoint can also be used access call history. ... View more

By default, the EST service allows calls to the United States and its territories as well as Canada. The ability to add (or remove) calling permissions for other countries can be found under Country Permissions beneath the Elastic SIP Trunking category in the navigation menu.   Modifying your calling permissions Upon accessing the Country Permissions page, you will find a list of all the countries currently available for calling. Select the checkbox next to each one you would like to enable. Take care to deselect any countries you do not need.       Once done, click SAVE & APPLY PERMISSIONS. Your Elastic SIP Trunks will now only be able to dial out to locations in the countries you have selected. ... View more

You must register a complete and accurate address associated with a phone number you're using with Elastic SIP Trunking. If you have not added an emergency address to the number, there is a banner directly above the Voice Configuration options for the telephone number warning that the number does not have an emergency address registered to it.     The Emergency Address is a physical address location that Sinch will send along with any 911 calls made from this number. If the 911 operators are unable to communicate with the caller, they will dispatch emergency personnel to this address.      Note: If a call is made to 911 from a number without an address registered to it, a $75 fee will be assessed per call.   Clicking the Add Emergency Address button will take you to the Emergency Services Registration page. Alternatively, it can be found under Emergency Services beneath the Numbers section of the navigation menu.   Validating your address Before we can register an address to your number, we need to validate the address to ensure there are no problems locating it. On the Address Validation screen for the number, enter the physical address associated with that number.     Our system will attempt to validate the address. If there is a problem, we will give you the option to edit the address and try again. Additionally, we will make our best “guess” as to the correct address and present it as an option for selection. This usually consists of small formatting changes. However, please scrutinize the correction we provide closely before accepting it.       After you have validated your address, you will be presented with a final screen to confirm your selections.       If everything looks correct press the Submit Request button to finish registering your address.     Your number is now registered with an address if anyone needs to dial 911 from it.   Important note! It’s important to confirm and test that the emergency call is provisioned with the correct address, but do not dial 911 to test that it is working properly! Instead, place a call to 933 from your SIP Device. This is treated exactly like a real emergency call except instead of being sent to a PSAP (Public Safety Answering Point), the call is routed to an automated system which will verbally read back the emergency address to you for your verification.   Congratulations! You now have a fully functioning Elastic SIP Trunk. Give yourself a hand! ... View more

New telephone numbers can be ordered by selecting “Get Virtual Numbers” underneath the “Numbers” section in the navigation menu.     Purchasing a new number After accessing the Get Virtual Numbers page, you will be presented with the opportunity to search for numbers in specific areas. Select the Voice feature option to ensure you are searching through numbers which can be used for voice communications.     Once you have found a number you would like to purchase, select the Get link for that number.     Enabling numbers for Elasting SIP Trunking After purchasing your new number, you will be taken automatically to the number’s configuration page to enable the new number to use Sinch Elastic SIP Trunking.     Note: If you clicked away, you can find the configuration page under “Your Virtual Numbers” beneath the “Numbers” section of the navigation menu.   Toward the bottom of the number configuration page, select the Elastic SIP Trunking button and find the SIP Trunk you created earlier in the SIP Trunk drop-down menu. Please note that default Voice configuration is for Programmable Voice and you must change it to Elastic SIP Trunking to use the number with your SIP Trunk.     Once done, press Save.   Next steps Now you must register an emergency address for the phone number you purchased.     ... View more

After configuring your SIP Endpoints, you will need to configure your outbound call settings.   Configuring Country Permissions is optional; US, Canada, Puerto Rico and USVI are allowed by default.  Either Access control lists (ACL) or Digest Authentication is mandatory if you intend to terminate calls to the PSTN. The ACL is used when an outbound call is initiated over the SIP trunk. If the IP that initiated the call is allowed in the ACL, then the call will go through. Otherwise, the call will be rejected with a SIP 403 Forbidden message.   Note: You can also use Digest Authentication alongside or instead of ACLs to authenticate your SIP calls.   Select the CREATE NEW ACL button to open the ACL creation wizard:     Enter a name for your ACL, as well as an IP Network Address and range using Classless Inter-Domain Routing (CIDR) notation. If you are unsure of the proper range, don’t worry. To help avoid mistakes we display the full range of IPs you are allowing at the bottom of the wizard window. Click CREATE when you have added your first range. When it is created you can add additional ranges to the ACL as needed.   After the ACL has been created, use the drop-down menu to select it and add it to the trunk.       Configuring Inbound Call Settings If you would like us to obtain and pass CNAM information for inbound calls to your numbers, you may enable it here.     Note: This service has an additional cost if enabled.   Congratulations! You have successfully set up your SIP Trunk. At this point you can configure your SIP PBX or Application to use the domain name you created and send outbound calls over your trunk. Give it a try!   Next steps Now you can purchase numbers to add to your SIP Trunks.   ... View more

To create a SIP Trunk, select the Trunks menu option under the Elastic SIP Trunking section of the lefthand navigation menu in the Sinch Build Dashboard. This will bring you to the main page where you can create, view, and edit your Elastic SIP Trunks.     Select the Create New SIP Trunk option from the upper right-hand side of the screen to begin configuring your new SIP Trunk.   Configuring your SIP Trunk   After selecting Create New SIP Trunk the SIP Trunk creation screen is displayed. Here you can name the Trunk as well as create the domain name used to send traffic over it.     In the SIP Trunk Name field, give your SIP Trunk a distinctive name to differentiate it from any other trunks. After naming your SIP Trunk, create the SIP Domain name you will use when sending calls to your Trunk. Whatever you enter will become your base domain name with .pstn.sinch.com appended to it. For example, if your domain was “mycompany” then your fully qualified domain would be mycompany.pstn.sinch.com.  Localized domain names are also available to send directly to your POP of choice.  If you do not use a localized domain name we will use geo-DNS to route your traffic to the nearest POP automatically.   Once you have completed these steps, press the Continue button, to move on to configure your SIP Endpoint.   Configuring your SIP Endpoint   A SIP Endpoint contains all the information necessary to send calls to your company IP-PBX or SIP Application. It also allows the IP Address of your device so it can send calls over the SIP Trunk. Select the Add an Endpoint option to continue. You can also select Next if you want to configure your endpoints later.     After you have selected Add SIP Endpoint the Endpoint configuration screen is displayed. Each endpoint will either be statically defined or use SIP registration.     Static (IP Auth) Endpoints   Give your SIP Endpoint a name which will help identify the device it is pointed to. This is often something like “San Diego Office.” Once done, also enter the IP Address, Port, and Transport protocol for your SIP Endpoint. If you require encryption, set your transport protocol to TLS.     Registered Endpoints   Select Registered from the Add New SIP Endpoint modal.  Give the endpoint a name, transport protocol, and priority for inbound call routing. You will then need to select a credential which will be used for determining the username and password when registering. If you have not yet created any credentials, then you can select Manage Credentials and create one now. If you require encryption, set your transport protocol to TLS.       Finishing Your Endpoints   Priority can be left at the default of “1.” The Priority is used to control inbound traffic if you create multiple SIP Devices. The lowest priority Endpoint will receive all inbound SIP Traffic, with the other SIP Endpoints only receiving calls if there is a failure to the primary. SIP Endpoints may also be configured to the same priority. This will cause the system to round-robin calls between them.   Once you are finished select the Create button to finish setting up your SIP Trunk.   If you are satisfied with the settings, select Next and if you wish to edit the endpoint you can select the hyperlinked endpoint name.       Next steps   After configuring your SIP Endpoints, you must configure your outbound call settings. ... View more

To get started, you must have an account set up on the Sinch Build Dashboard.   Note: The following guide is generic and does not cover all configurations necessary for setting up a specific device (PBX, SBC, UCaaS, etc.). For specific integration guides, view the following guides found here.   All Elastic SIP Trunking and Numbers operations can be found under their respective sections on the lefthand navigation menu of the Sinch Build Dashboard.      A Sinch Elastic SIP Trunk consists of two primary elements— the SIP Trunk itself, where the Trunk’s name and domain are configured, and one or more SIP Devices which will contain the IP/Port and Transport options to connect with your IP PBX or Application.   Configure your IP Allowlist The Sinch Elastic SIP Trunking platform will send your calls from the following IP Addresses. Please allow these IPs in your network’s IP Allowlist: SIP:  206.146.131.0/28 206.146.133.0/28 206.146.134.0/28 206.146.136.0/28 206.146.137.0/28 206.146.138.0/28 206.146.139.0/28 206.146.141.0/28 Media: 206.146.131.128/26 206.146.133.128/26 206.146.134.128/26 206.146.136.128/26 206.146.137.128/26 206.146.138.128/26 206.146.139.128/26 206.146.141.128/26 Non-encrypted SIP traffic from the Sinch Elastic SIP Trunking network will be sent from port 5060. SIP calls encrypted with TLS will be sent from port 5061. For media we will use UDP ports 10,000-60,000.   SIP Trunk technical specifications Within your IP-PBX or SIP Application make sure your SIP Trunk is configured with the following settings. CODECs: G.711 μ-law (aka G711u, G711MU or PCMU) & G.711 A-law (aka  PCMA or G711A) DTMF – RFC 2833/4733 with payload type 101 Privacy header – Sinch Elastic SIP Trunking is designed around the use of P-Asserted-ID for the passing of privacy and STIR/Shaken information. However, we will also send Remote-Party-ID on calls inbound to location.   Next steps Now you can learn how to create your first SIP trunk.   You can also check out Elastic SIP Trunking release notes to see new enhancements. ... View more