A new snapshot is available. This will hopefully be the last before pb13.1 is released. The only change in this release compared to the last is a new NAT settings override for SIP accounts.
http://downloads.askozia.com/pbx/snapshots/r422
Also, this will be the last development snapshot to be announced on pbx-users, pbx-announce and the Askozia website. We’re attempting to limit development discussion to the pbx-dev mailing list and the Development section of the forum so normal users are not overwhelmed by the noise.
Forum Thread
A new snapshot is available which should resolve the outgoing caller id issues with pb13. Incoming caller id can now also be prepended or replaced with a user defined string. Finally, SIP phone accounts have “nat=yes” set by default. There was a consensus that this is a more logical default. Please report if it breaks previously working phones (do give them enough time to reregister).
http://downloads.askozia.com/pbx/snapshots/r420
Enjoy!
Forum Thread
A new snapshot is available which should fix the following bug present in pb13:
Multiple accounts with the same SIP provider causes abnormal incoming call routing.
http://downloads.askozia.com/pbx/snapshots/r417
Thanks to Sergio in the forums for help on this!
Forum Thread
A new development snapshot has just been uploaded to the downloads server.
http://downloads.askozia.com/pbx/snapshots/r408
Changes since last snapshot (r403):
- reverted to Asterisk 1.4.17 for testing purposes (SIP provider registrations were failing with 1.4.18, this is to see if it truly is 1.4.18’s fault)
- added a few extra sounds so DISA() and Authenticate() work (these were very small files, unfortunately not every application’s prompts can be added)
- added Danish language prompts and voicemail notification e-mail translation (provided by McM in the forums)
- fix phone extension gathering (reported by devon in the forums)
- dmtf tones are no longer played back after answering a ringing analog phone (fixed by David Lawrence)
Forum Thread
A new development snapshot has just been uploaded to the downloads server. A few bug fixes and improvements are here along with a new version of Asterisk.
If there are no major bugs reported with this snapshot, pb13 will soon be released.
http://downloads.askozia.com/pbx/snapshots/r403
Changes since last snapshot (r395):
- updated Asterisk to 1.4.18
- revamped the printable dialplan page with new formatting, fields and custom application listings
- updated the way external phones are listed on the phone accounts page
- removed the number only restriction on dialstrings for external phones
- added a description field to external phones
- added res_convert to the included Asterisk modules to help those interested in converting the audio prompts to other formats (unfortunately, g729 will not work with this but it is a start)
- hovering over memory usage now displays kbytes used
- bug: all patterns for analog providers are now displayed (reported by devon in the forums)
- clear up the parkinglot documentation
- zaptel channels are now properly restarted after changes
Forum Thread
I just put up a new snapshot on the downloads server. Thanks to everyone who is testing these in their free time!
http://downloads.askozia.com/pbx/snapshots/r395/
Changes since revision 384:
- small fix to incoming provider context generator
- fix a rather important copy paste error which prevented provider deletion
- Advanced settings section added to all applicable acount pages. Less commonly used settings now have a place.
- corrected colspan attributes in advanced settings table
- this logic was fine when only used to split up dial patterns: updated to work more generically for new fields
- manual attributes may now be defined for phones, providers and under “Advanced” for SIP and IAX technologies (i.e. hacks now survive reboots)
- generalize help text in display_qualify_options and use it for iax and sip phone pages
- factored out the common caller id field into a new display function
- factored out the common description field into a new display function
- stop labeling the MAIN partition to cut down on annoying messages
- fix up some copy paste errors in generating external phone extensions (reported by Brian in the Forums)
Forum Thread
A new development snapshot has just been uploaded to the downloads server.
http://downloads.askozia.com/pbx/snapshots/r354/
Changes since last snapshot (r350):
- incoming calls from SIP providers should be working again
- callgroups are usable again (an existing bug may still be present, this fixes a parse error)
- jQuery animation added to menu hide/show
Forum Thread
A new development snapshot has just been uploaded to the downloads server.
http://downloads.askozia.com/pbx/snapshots/r350/
Changes since last snapshot (r344):
- updated: Asterisk to 1.4.16.2
- added: ajax.cgi allows execution of Asterisk Manager Interface and shell commands
- changed: “Diagnostics -> Manager Interface” now uses AJAX to query the new ajax.cgi backend
- changed: /exec.php now uses AJAX to query the new ajax.cgi backend
I haven’t had the equipment necessary to work on some of the reported bugs. Apologies for that. This ajax.cgi backend will be used by the storage system and any testing you can do using the listed pages would be appreciated. The live graphs have also been ported over to use it.
Forum Thread
A new development snapshot of revision 344 has just been uploaded to the downloads server.
http://downloads.askozia.com/pbx/snapshots/r344
Changes since the last snapshot (r336)
- added: factory default reset button support for alix23x platform (merged from m0n0wall)
- updated: Asterisk to 1.4.16.1
- updated: timezone information (merged from m0n0wall)
- bug: nge network interfaces are no longer ignored (merged from m0n0wall)
- bug: call records are now sorted properly (reported by Jakob Strebel)
