pb13.2 has just been released fixing a couple of issues with pb13.1:
- missed call notifications are no longer sent for successfully completed calls
- incoming calls from SIP or IAX providers landing in voicemail will now be read back the account’s username if it is numeric instead of the internal extension
- incoming calls from providers will now be accepted in more cases (previously only numeric and ’s’ extensions would be matched, now all extensions containing alphanumeric, # and * characters will be matched)
In addition to these bug fixes, a couple of new features were added:
- transmit and receive gains can now be set for analog interfaces (working patch provided by devon in the forums, modified for code consistency)
- improved documentation on the analog and isdn interfaces pages
- manual attributes can now be defined for analog interfaces
- an authentication method can now be selected for the SMTP server used in “Services -> Voicemail”
- * and # characters may now be used in application extensions
Thanks for the continued feedback and to those starting to submit code to the project!
Download: http://askozia.com/pbx
Changelog: http://askozia.com/pbx/changelog
pb13.1 has just been released fixing a couple of issues with pb13:
- having multiple SIP Provider accounts on the same host no longer results in unpredictable incoming routing
- outgoing caller id overrides in providers are now functional
In addition to these bug fixes, a couple of new features were added:
- phones and call groups now have selectable ring lengths
- incoming caller id from providers may be prepended or replaced by a user defined string
- SIP accounts now have a configurable NAT parameter
Please note, the default NAT settings for SIP phones and providers were changed in this release and there is a possibility that phones or accounts which were working correctly before may no longer register. To cure this, adjust the new NAT option for the provider or phone to match your situation. A consensus was reached where defaulting NAT to “yes” would work in the most cases. Please report any cases where this new default breaks a previously working account.
Thanks to everyone testing development snapshots and helping troubleshoot bugs!
Download: http://askozia.com/pbx
Changelog: http://askozia.com/pbx/changelog
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
Public beta 13 is ready for download!
As the changelog shows, a ton of changes have been implemented since pb12.2. Some of the major additions:
- Extensions can now be exposed to public networks and therefore be directly dialed (i.e. myextension@myhost.com).
- Phones can also directly dial SIP uri’s instead of requiring a provider.
- Custom applications can now be added to the dialplan.
- Incoming extensions from providers has been made more flexible allowing for multiple parallel destinations.
Many bugs were eliminated for this release. A huge thanks to everyone testing the new development snapshots! Through their feedback, we’re able to make such large changes with better assurance their release form will be stable.
There are, however, still some issues with pb13:
- ISDN provider routing logic needs improvement.
- Multiple accounts with the same SIP provider causes abnormal incoming call routing.
- Call notifications are sent even if a call is answered.
- Caller ID (outgoing and incoming) is broken in certain cases
We will attempt to make the public beta release cycle shorter in the future. The project’s growth requires us to keep the infrastructure up-to-date and does sometimes distract us from keeping the updates coming as quickly as we would like. For the next few weeks, we’re back in 100% development mode. Also, a bit of documentation as to how releases are going to be managed is now available here (http://askozia.com/pbx/development/).
Once again, thanks to everyone for testing and providing feedback!
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
We’re having some problems with the mailing lists at the moment. Please be patient, we’ll update this entry as soon as they’re back online.
Update: The DNS problem has been fixed but will take some time to propagate.
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
