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    Incoming calls are disconnected after a few seconds

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    If you are experiencing issues with incoming calls, follow the steps below:

    1. Enable the NAT keep alive setting on the devices experiencing the issue.

    2. Change your device’s local SIP port from 5060 to an alternative one (e.g. 49000), ensuring that this port is not being used by other devices.

    3. Try using port 6050 instead of 5060 for the SIP Server. To locate this setting, refer to your device’s configuration instructions.

    4. Disable SIP ALG on your router. For detailed instructions on how to disable SIP ALG, please refer to the relevant Help Center article.

    Frequently asked questions

    The most common cause is NAT-related issues or incorrect handling of SIP traffic by the router. In many cases, features such as SIP ALG negatively affect communication between the VoIP device and the SIP server.

    Calls that disconnect after approximately 30 seconds are often associated with NAT traversal issues, SIP ALG, or incorrect SIP port settings. Enabling NAT Keep Alive and disabling SIP ALG can help resolve the problem.

    NAT Keep Alive is a feature that maintains an active connection between the VoIP device and the SIP server, preventing the router or firewall from terminating the communication.

    SIP ALG (Session Initiation Protocol Application Layer Gateway) is a feature found in many routers that attempts to manage SIP traffic. However, it often causes issues such as dropped calls, audio loss, or SIP registration failures.

    Yes. NAT settings, firewall rules, or SIP ALG functionality on the router can affect the proper operation of VoIP services and lead to call disconnections.

    The most common symptoms include:

    • Calls that disconnect unexpectedly

    • No audio during calls

    • One-way audio

    • SIP registration failures

    • Problems with incoming or outgoing calls

     

    Does changing the SIP port affect my phone number?

    No. Changing the local SIP port only affects how the device communicates with the SIP server. It does not change your phone number or any of your telephony services.

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