This scenario was broken:
A ring group member forwards their phone to a destination. The destination is an external number and the outbound route had a toll_allow condition.
This error would be generated: [ERR] switch_cpp.cpp:1376 [route_to_bridge] Unsupportded condition: ${toll_allow}
This PR will get the toll_allow values from the RG member that is forwarding their phone. Then it will pass it to 'route_to_bridge.lua'.
* Add. Use `route_to_bridge` module to build routes fro ring groups.
This commit has several improvements
1. Select only needed fields. (do not select quite big XML text strings)
2. Filter routes also by context name
3. Filter dialplans also by hostname
4. Handle conditions based not only `destination_number`
5. Handle `break` and `continue` attributes for extensions
6. Escape vars inside dial-string
7. Add log messages similar as FS dialplan do
* Add. `route_to_bridge` set inline vars so it possible use then in next conditions.
Add. `route_to_bridge` can execute basic api commands from allowed lists.
`route_to_bridge` expand all known vars. If var is unknown then it pass as is.
Fix. `export nolocal:` action.
* Fix. Short variable names
* Add. some comments
* Fix. Do not try execute empty string
This produce error messages `[ERR] switch_cpp.cpp:759 No application specified`
* Fix. Export nolocal values.
Break the ring group query into two parts. One for getting the ring group info. Another for getting the ring group user.
If a user was not assigned to a ring group, then the query will fail to return a result. This will cause forward, prefix, distinctive ring, etc to not work properly.
* Merge (#4)
* Added variables to disable call waiting and t.38 faxing
* HT702 config file variables: grandstream_disable_call_waiting, grandstream_disable_fax_t38
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update {$mac}.cfg
* Update app_config.php
* Corrected bug with expansion board and call park value. Was 19, should be 16.
* MAC detection for some Grandstream phones (#2486)
Some GS models send the Mac address in the user agent
* Add/Update German and Austrian translations (#2483)
Updates to the following apps:
call_broadcast
call_flows
devices
dialplan_inbound
dialplan_outbound
edit
emails
exec
extensions
fax
fifo
fifo_list
gateways
ivr_menus
* Update destinations.php
* Update xml_cdr_inc.php
* Update status_registrations.php
* Update y000000000028.cfg
* Update y000000000066.cfg
* Update y000000000035.cfg
* Update y000000000065.cfg
* Update y000000000051.cfg
* Update y000000000023.cfg
* Update y000000000025.cfg
* Update y000000000029.cfg
* Update y000000000036.cfg
* Update y000000000038.cfg
* Update y000000000032.cfg
* Update y000000000046.cfg
* Update y000000000000.cfg
* Update y000000000054.cfg
* Update y000000000045.cfg
* Update y000000000069.cfg
* Update y000000000004.cfg
* Update y000000000044.cfg
* Update y000000000044.cfg
* Update y000000000005.cfg
* Update y000000000052.cfg
* Update y000000000007.cfg
* Update y000000000037.cfg
* BugFix for VM Transcription (#2491)
Records as MP3 and skips transcription steps if a user has transcription set to false.
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Yet another ip phone reporting MAC in the UserAgent (#2492)
* Update app_config.php
Begin adding uuid's for key functions.
* Update app_config.php
* Update app_config.php
* Update app_config.php
* Update app_config.php
* Update app_config.php
* Update app_config.php
* Create 560_extension_queue.xml
* Update y000000000037.cfg
* Update y000000000007.cfg
* Update app_config.php
* Update y000000000052.cfg
* Update y000000000028.cfg
* Update y000000000005.cfg
* Update y000000000044.cfg
* Update y000000000044.cfg
* Update y000000000004.cfg
* Update y000000000069.cfg
* Update y000000000045.cfg
* Update y000000000000.cfg
* Update y000000000046.cfg
* Update y000000000032.cfg
* Update y000000000038.cfg
* Update y000000000054.cfg
* Update y000000000036.cfg
* Update y000000000029.cfg
* Update y000000000066.cfg
* Update y000000000035.cfg
* Update y000000000065.cfg
* Update y000000000051.cfg
* Update y000000000023.cfg
* Update y000000000025.cfg
* Update app_config.php
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update {$mac}.xml
* Update app_config.php
* Fix. Add new sip profile setting. (#2500)
* BugFix [master] system information git (#2499)
fix for if the .git folder is present but corrupt
* Fix. Redirect back to correct profile after delete setting. (#2501)
Fix. Add setting (seems need also set sip_profile_uuid)
Change. Order settings by name when output.
* Restore Button + Audiocodes preliminary support (#2502)
* Audiocodes phone support + restore default for vendors
* Restore button
* Restore script
* Update device_vendors.php
* Some FR & ES translation fixes (#2503)
* Update app_config.php
* Update app_config.php
* Create {$mac}.cfg
* Create directory.xml
* Create favorite_setting.xml
* Create y000000000025.cfg
* Fix. Add extension with non numeric extension number. (#2508)
* Update ring_group_edit.php
* Update call_edit.php
* Create 080_default_caller_id
* Update 080_default_caller_id
* Rename 080_default_caller_id to 080_default_caller_id.xml
* Fix. Create needed number of extensions (#2509)
* Update dialplan_edit.php
* Update switch.php
* Update index.php
* Update switch.php
* Update voicemail_edit.php
* Update app_config.php (#2515)
Add the necessary permissions in order to use the database save function (fusion 4.3) when coding for voicemail option adds/deletes/updates.
I realize these voicemail permissions overall will probably get cleaned up even more once the whole app is updated to use the database function, but this is a stop gap measure. I am working on adding voicemail options to the Bulk Account Settings app and I'm stuck without these permissions. The function is kicking a out 403 Forbidden.
* Update app_config.php
* Really use configured transcribe_language for transcription (#2513)
* Fix renaming domains (#2512)
* Make presence for conferences work out of the box (#2514)
Use '@' instead of '-' for separating conference name and domain, which
is what FreeSWITCH mod_conference uses.
* Add. Speed dial respects contacts user. (#2249)
* Add. Speed dial respects contacts user.
One user can not use speed dial numbers from contacts
associated with another user
* Make SQL query more efficient
* Add. Support find contacts by user groups as well
If contact has set any `user` or `group` then only this users can use speed dial numbers
in other case speed dial numbers are global for domain.
* Fixup for renaming domains (#2517)
The previous fix was incomplete, this one should do it.
* Create app_defaults.php
* Update page.lua
* CC Key Support for Call Center (#2518)
* Exit Keys support
* Multilanguage support
* CC Key Support
* Update call_center.php
* Update code to iterate over numbers. (#1727)
* Update code to iterate over numbers.
This code fixes some problems
* range like `009-010` current code produce numbers `009` and `0010`
* range like `200-100` now raise error so it will be easy to debug
* range like `010-20` now generate error.
* `destination` is string so condition `destination == tonumber(sip_from_user)` is always false so change it `destination == sip_from_user`
* Fix. handle ranges like `100-100`
* Update app_config.php
* Update app_languages.php
* Update ring_group_edit.php
* Update index.lua
I really don't know if this is an IP phone issue (tested on grandstream) or a new behavior on FreeSWITCH, but in order to add the prefix to the caller id (name or number) correctly, you need to export instead of set.
* Add. attr_xfer analog based on conference.
Control DTMF sequence
`*0` transfer `self` to `enter number` state and `peer` leg to conference room
`##` transfer `self` to `enter number` state and hangup `peer` leg
`*#` transfer `self` to conference room and hangup `peer` leg
* Remove spaces.
* Add. Commented action to ring group.
FusionPBX
konradSC
Alexey Melnichuk
Len
Luis Daniel Lucio Quiroz
mafoo