Sangoma Trunking Module
Joshua C. Colp
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Sangoma provides two SIP trunking services which are available to customers, SIPStation which is a great solution for the everyday user and VoIP Innovations which
Sangoma provides two SIP trunking services which are available to customers, SIPStation which is a great solution for the everyday user and VoIP Innovations which
Did you know that sipp can play back RTP streams? A lot of long time users may not! Did you know that you can use
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
As mentioned in this post, Asterisk now supports the use of RFC4733 digits in common bitrates beyond 8kHz. At the end of the post, we
The ability to send SIP NOTIFY messages to endpoints and arbitrary uri’s via AMI and CLI has existed for some time within Asterisk. Changes from
In a previous blog post we talked about using Asterisk’s uni-cast functionality as a bridge between the PSTN and an external service. This post entails
A recurring theme I’m seeing lately is people deploying VoIP, running into issues, and not approaching their issues from the perspective of taking all components
Note: This new implementation is available as of Asterisk 18.22.0, 20.7.0, and 21.2.0. It’s been almost 4 years since STIR/SHAKEN support was first added to
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
This is a bit of a strange blog post but recent responses to people reminded me of the Colp Hard Facts webinar I did recently.
