An interesting issue was recently filed where dialing a Local channel resulted in the following: [2021-03-28 22:41:35] ERROR[15773][C-00000030]: translate.c:1396 ast_translator_best_choice: Cannot determine best translation path
In a past blog post I talked about how you can’t reload transports without enabling explicit support, but alluded to changes coming in the future.
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
When stream support was added to Asterisk it was initially done with the focus being for SFU with a single video stream from each participant
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
Over the past few years we’ve been working to improve the video support in Asterisk. We initially started with adding stream support[1] in a backwards
If you download Asterisk 17 and start it up, you might be one of the people that notices the following messages: If you are using
A comment that I see frequently when helping people with PJSIP is the lack of a general section (with global options) and how this causes
When DTLS support in Asterisk was added the information about how to use DTLS support in OpenSSL was not as flushed out as it is
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