Public Change Log

PhreakNet has been responsible for finding bugs, suggesting feature improvements, and adding features to some open source projects, mostly to Asterisk. Below is a list of some of these bug fixes and features.

Asterisk

Feature development is generally driven by user and organizational needs. You may contact us if you would like us to develop a feature for you, in exchange for a small donation to keep our work going. All patches are submitted to Sangoma for consideration for inclusion in its Asterisk code.

Features We Have Created or Added To (or Bugs We Have Fixed In) Asterisk, merged into official Asterisk

• Patched SIP channel driver to recognize application/hook-flash as a hook flash event — patched March/April 2021, merged May 2021
• Patched file.c to not throw errors on hook flash events — patched March/April 2021, merged May 2021
• Added hook flash support by exposing flash events through AMI (Asterisk Manager Interface) — patched April 2021, merged May 2021 — download patches
• Added custom tone option (ability to play custom tone or none at all) to VoiceMail() application — patched April 15, 2021, merged May 2021 — download patch
• Added MIN, MAX, and ABS dialplan functions — May 2021, merged May 2021
• Added read capabilities to VOLUME function — May 2021, merged May 2021
• Added configurable answer supervision option to ConfBridge() application — May 2021, merged June 2021
• Missing config from SIP to PJSIP conversion script — May 2021, merged June 2021
• Patched PJSIP to recognize application/hook-flash as a hook flash event — May 2021, merged June 2021

Features We Have Created or Added To (or Bugs We Have Fixed In) Asterisk, not yet merged into official Asterisk

• Added OTHER_CHANNEL dialplan function — May 2021
• Expanded A option in Dial application, allowing for caller announcement instead of or in addition to called announcement — May 2021
• Added WaitForCondition() application, allowing for any arbitrary expression to evaluate to true, with configurable timeout and polling interval — May 2021
• Added ConfKick() application, allowing for a specific channel, all participants, or all non-admin participants to be kicked from a specified conference bridge — May 2021
• Added encryption to RSA authentication in IAX2 — reported 2012, partially patched 2014, addressed 2021 + bug fix
• Added option to set channel variables using Originate application — May 2021
• Added Reload() application — May 2021

Bugs We Have Reported and Worked On

• slin192 incorrectly picks gsm over ulaw if both available and costs are identical — May 2021
• ANI2/Originating Line Information (OLI) support missing from PJSIP — June 2021

Bugs We Have Reported and Sangoma/Digium Has Fixed

• Fatal Asterisk crashing error where unauthenticated IAX2 calls are attempted and authentication is required — patched February 2021
• DTMF events duplicated on progress events (12 year old bug) — patched March 2021
• Prevent duplication of flash events in res_rtp_asterisk (AVT/RFC2833 DTMF Flash Events) — patched March 2021

Bugs In Which We Have Participated

• Audiohooks require constant media flow for whispering — reported 2014, no resolution

Bugs We Have Reported That Have Not Yet Been Fixed

• VoiceMail() does not cancel an in-progress recording when it should (e.g. hangup on rerecord option) — reported April 2021
• Somewhat bizarre issues with MixMonitor() - autofallthrough error and inaudible tone with the B() option — reported April 2021
• DTMF on progress results in infinite loop if progress followed by hangup received — reported May 2021
• Improper same-provider multi-registration parsing in SIP to PJSIP conversion script — June 2021

In Progress Development

• New NOTCH_FILTER function
• Improvements to third-party SoftModem() application
• Digit collection callback using framehooks

Non-Asterisk

Features We Have Suggested Which Someone Else Has Added

• SameSite Strict option in phpMyAdmin (released in phpMyAdmin 5.1.0) — patched October 2020, released February 2021