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
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
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