Enabling direct RTP streams between SIP phones in Asterisk

By default, asterisk installations will instruct SIP phones to pass their media streams (RTP streams) through the asterisk server itself. That is to say, the RTP stream would look something like this: (phone 1) <-----> (asterisk) <------> (phone 2) However, for performance reasons, especially in non-NAT environments, it is preferable to have the RTP streams … Continue reading Enabling direct RTP streams between SIP phones in Asterisk

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Asterisk VoIP : Getting your outbound CallerID to show properly

We lately ran into a scenario where our asterisk server is connected via SIP trunk to our DID provider (mydivert.com). Our PABX server opens a single SIP trunk to the provider, however we have multiple DIDs running over this trunk. Having multiple DIDs means we can use multiple phone numbers, in different countries, benefiting from … Continue reading Asterisk VoIP : Getting your outbound CallerID to show properly

Web MeetMe : Conferencing in FreePBX

The company I work for currently uses TrixBox as their VoIP server. It's an excellent piece of software but being no longer supported, the decision was taken to upgrade to the later and more active FreePBX. Since they are both built around the asterisk core, I figured this upgrade wouldn't be too much of a … Continue reading Web MeetMe : Conferencing in FreePBX