If you didn’t already know, both SIP and HTTP share the same digest authentication mechanism described all the way back in RFC-2069 “An Extension to HTTP
Overview Realtime has been around for a while now, but operating systems are constantly evolving. Because of this, guides can become outdated. I find myself
Most SIP endpoints will qualify each other by sending OPTIONS back and forth to make sure they are there and can respond. Asterisk uses this mechanism
Last month I wrote a blog post titled “Configuring an Asterisk build from the command line” which outlined how to use the menuselect command to
If you’re an Asterisk package maintainer, you already know this. If not, did you know you can run menuselect/menuselect from the command line to enable/disable modules, set
When PJSIP support in Asterisk was being developed one of the critical areas of development was transports. These are for the most part provided by
What is STIR/SHAKEN? STIR/SHAKEN (Secure Telephony Identity Revisited / Signature-based Handling of Asserted information using toKENs) is a new technology that the telecommunications industry is
When transitioning from the chan_sip channel driver to chan_pjsip one of the items that can catch people off guard is the use of SIP URIs
One of the most common issues I see when people deploy SIP is calls hanging up after approximately 30 seconds or traffic not going to
Asterisk offers its users great flexibility in most of its features. One of them is the choice between three different modules for different ways to
The Sangoma Blog