Microsip Bad Gateway

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Amelie Robertos

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Aug 3, 2024, 10:36:23 AM8/3/24

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Make sure you have entered valid "SIP server", optional "SIP proxy". Your SIP provider should provide you with this information.
It is also possible that your SIP operator must add your IP address to the server's whitelist.
Try changing the transport: UDP, TCP or TLS.
You can also try spoofing the "User-Agent" string in the microsip.ini file.

Check your SIP server, domain, username, password. The proxy and login are often empty, but you must specify them if required by your SIP provider. Some SIP providers require that you enable the STUN server if your PC does not have a public IP address.

This is a response from your PBX. Your PBX may not support the requested audio/video codec, encryption, or other requested feature that you have enabled in microsip. Refer to the PBX logs for details. You also should test with a clean installation of microsip, where all additional features are disabled by default.

If you can't change PBX configuration, you can try to enable "Allow IP rewrite" feature, that will do that work on the softphone side and if possible disable "SIP ALG" in the router/routers settings. "SIP ALG" may interfere with the correct rewriting of IP.

This article explains how to configure SIP Gateway so that your organization can use compatible SIP devices with Microsoft Teams. To find out what SIP Gateway can do for your organization and what hardware, software, and licenses your organization needs for it, read Plan for SIP Gateway.

Open your firewall to Microsoft 365 and Teams. Open your network's firewall to Microsoft 365 and Teams traffic as described in Office 365 URLs and IP address ranges. Firewall rules are needed for outbound traffic only.

You can also enable SIP Gateway by using the PowerShell Set-CsTeamsCallingPolicy cmdlet. To enable users for SIP devices, select a policy, and set the -AllowSIPDevicesCalling attribute to True. The default value is False, so users will not be able to use their SIP devices unless you enable them.

Add SIP devices to your Teams organization by configuring the above SIP Gateway provisioning server URL in your DHCP server. To learn more about DHCP server, see Deploy and manage DHCP. Also, you can use DHCP option 42 to specify the Network Time Protocol (NTP) server, and DHCP option 2 to specify the offset from Coordinated Universal Time (UTC) in seconds. The devices in your organization will be routed to the SIP Gateway provisioning server. Successfully provisioned SIP phones will display the Teams logo and a soft button for sign-in.

Ensure SIP devices are on the minimum supported firmware version for onboarding. During onboarding, SIP Gateway will push the default configuration and authentication user interface to the device. To find out the required firmware version for SIP devices, see Plan for SIP Gateway.

Conditional Access is a Microsoft Entra feature that helps ensure that devices that access your Microsoft 365 resources are properly managed and secure. SIP devices are not managed by Intune hence stricter conditional access checks are applied to them. SIP Gateway authenticates SIP devices with Microsoft Entra ID, so if your organization uses Conditional Access for devices in the corporate network, you should do one of the following:

On the Provision devices pane, under Waiting on activation, select a device and then select Generate verification code to generate a one-time verification code for each provisioned device. Note the verification code for each SIP device.

On the SIP device, dial the enrollment feature code followed by the verification code. On the SIP device, dial the enrollment feature code *55* (used by SIP Gateway for enrollment one-time-verification code validation), followed by the verification code that is generated in Teams Admin Center for this particular device. For example, if the verification code is 123456, dial *55*123456 to enroll the device.

Enter the pairing code displayed in the Sign in a user dialog into the web authentication app to pair the SIP phone with the user's account. On a successful sign-in, which might take a while, the SIP phone will display the phone number and username, if the device supports it.

Press Sign-in on the SIP phone to display the authentication URL and pairing code. The pairing code is time-sensitive. If it expires, the user must press Back on the phone and start the sign-in process again.

Enter the pairing code displayed on the SIP phone into the web authentication app to pair the SIP phone with the user's account. On a successful sign-in, which might take a while, the SIP phone will display the phone number and username, if the device supports it.

Username column: Put in the list of Microsoft Entra user names or user principal names (UPNs) to use to associate with the device's MAC address found in the HardwareId column.

The DeviceDetailsFilePath parameter specifies the location of the CSV you created and saved. The Region parameter specifies the SIP gateway provisioning region where the devices are being deployed. The values are: APAC, EMEA, NOAM.

User details and policies will be fetched to SIP devices when users sign in. Any policy changes thereafter for signed-in users will be synced to the device within one hour. Devices must have their registration refreshed with the SIP Gateway periodically. SIP phones depend on Call Redirect, so the admin must set the AllowCallRedirect attribute in Set-CsTeamsCallingPolicy to Enabled.

A SIP device can usually display information in many languages. Setting its UI language affects its interface, including softkeys and sign-in/sign-out system messages. Setting the UI language is done in the provisioning server, using DHCP server, or manually by appending a code string in the URL as in the following examples.

If Sign in to make an emergency call text is not translated to other languages, an abbreviated version in English only will be presented on Press Sign In on the following IP phone models due to a screensize limitations:

SIP Gateway only supports IPv4. Microsoft Teams service and client support both IPv4 and IPv6. If you want to control communications to Microsoft Teams, use the IP address ranges in Microsoft 365 URLs and IP address ranges.

SIP Gateway supports dynamic emergency calling (dynamic E911) for compatible SIP devices that share network attributes over the wire. These attributes are provisioned in the Teams admin center and can be a combination of local IP and subnet length, or chassis ID and network port number. For devices that do not share location attributes, or if the location is not resolved dynamically for any reason, SIP Gateway will continue to support emergency calling based on registered addresses. Currently, registered addresses are not supported for Direct Routing scenarios. For more information about emergency calling, see Plan and manage emergency calling.

We are often asked "why I cannot hear peer side?" or "why we cannot hear each other?". In most scenarios, the root reason is firewall filtering audio stream or NAT (network address translation) blocking it. We can always find that some SIP devices, including SIP phones, SIP clients and VoIP gateways, are deployed behind a NAT and configured with private IP address.

If there is a firewall deployed in your VoIP network, for example Windows firewall, please try to shutdown it and make some test. If problem is resolved, that means you need ask administrator to configure firewall to enable miniSIPServer to be visited from public networks or private networks.

Please check your router whether it can support SIP-ALG (Application Level Gateway) function. If it has such function, please disable it since most routers are full of bugs in their SIP-ALG. Once you disable it, please remember to restart your router.

If your SIP phones/devices are deployed in a private network, mostly you need configure STUN (Simple Traversal of UDP through NATs) server to help your SIP devices to route packages, such as audio packages. Most SIP devices can support STUN protocol.

If your miniSIPServer is deployed with public address, you can use it as your STUN server too because it is embeded in miniSIPServer. Of course, you can also use our simple STUN server "stun.minisipserver.com".

If the router has UPnP feature and has enabled it, you can configure miniSIPServer to request port mapping (forwarding) automatically. In miniSIPServer system configuration, please enable item "Enable UPnP to ask router to map ports".

In some area, local ISPs could block or change SIP signals which can also cause one-way audio problem. To avoid that, you can try to change standard SIP port or use SIP over TCP/TLS. Please refer to following documents for more details.

Beginning in early 2023, SIP Gateway will support analog telephones, allowing you to use Teams calling functionality on the following compatible AudioCodes MediaPack analog telephone adaptor (ATA) models:

The MediaPack series supports static IP addressing. Therefore, a proper working static IP address/netmask/gateway/DNS needs to be set up for the device to establish IP network connectivity prior to applying provisioning settings for SIP Gateway. AudioCodes provides some comprehensive guidance on this in their Quick Start Guide.

Hi everyone, I am Ekim, a fresh Bootcamp graduate and an IT helper (I don't dare to call myself a programmer yet). Every Friday, I will share some of the work that I've done over the last week in a bid to get feedbacks from you guys and record my journey to become a programmer.

Over the last week, I've been familiarizing myself with Asterisk, the open-source communications toolkit, which powers IP PBX systems, VoIP gateways, and conference servers. It is such a bitter start for a coding newbie like me, in which I struggled a lot in the installation process and making a simple phone call under the same network. This article aims at sharing a laconic work flow of the asterisk set-up and how a video call is made.