Joshua C. Colp

Blog Author

PJSIP Transport Reload Fun

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 PJSIP and are what allow the flow of SIP signaling. PJSIP provides UDP, TCP, and TLS transports and we provide one for Websockets for WebRTC. Naturally we needed to allow

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PJSIP Invite Session Lifetime

In the past month I’ve been fixing an issue with Asterisk and PJSIP that I thought would be fun to share in a blog post. The originally filed issue was for a crash experienced when Asterisk was manipulating the reference count of a PJSIP invite session. For those who may be unaware the INVITE session

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Asterisk 13 and 17 Security Fix Only

While this has been mentioned in a few places I thought it prudent to also create a blog post. Asterisk 13 and Asterisk 17 have entered security fix only status. What does this mean, though? When bug fixes and changes are put up for review they will no longer be done against Asterisk 13 and

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Asterisk 18.0.0 Released!

Kia Ora! (Or Be Healthy) (Don’t ask – I like greetings from various languages) If you haven’t noticed from the various emails and posts Asterisk 18.0.0 has now been released and is available for download here! As previously mentioned in our blog post for 18.0.0-rc1 this is an LTS release, meaning it will be supported

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A Video Update

Video has been a continued theme of Asterisk for some years now. We put into place the foundation to allow us to do video better, and have over time taken advantage of this and advanced things further. I thought I would take a little bit of time to reflect back on what has been done.

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Asterisk-18.0.0-rc1 Released!

Greetings Everyone! Another year has flown by and while we now live in interesting times this has not stopped the development and progress of Asterisk or the Astricon conference[1], so much so that the first release candidate of Asterisk 18 is here[2]. The first step in creating the 18.0.0 release of Asterisk is tagging a

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Common SIP URI Issues

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 within PJSIP. This is because in chan_sip these are generated on your behalf based on different configuration options while in chan_pjsip we leave this up to the user. Let’s take

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Local Channel Multistream and Re-Negotiation Support

When stream support was initially added to Asterisk we did it in the most backwards compatible way possible to ensure that we did not have to modify the entirety of Asterisk. This has allowed us to gradually improve parts of Asterisk as we’ve expanded our stream and video support. To that end the next part

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ARI Create Channel With Variables

In a blog post long ago we talked about the addition of the create and dial ARI functionality for allowing channels to exist within ARI applications before they have been answered. This has seen use by various people and it came to light that it presented a slight difference in API definition in comparison to

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