Broken SipSettings module
describe "more serious"... because fwconsole chown didnt fix the issue [root@voip ~]# fwconsole chownSetting Permissions... 24743/24743 [============================] 100%Finished setting permissions
View ArticleBroken SipSettings module
Are the following files present on your system in the folder /etc/asterisk: sip_nat.conf sip_general_custom.conf sip_custom.conf Did you file a bug? edit: provide the output from: ls -l...
View ArticlePhonebook issue
When I enter the phonebook tab I see this and I can not save an entry. Whoops\Exception\ErrorExceptionUndefined index: nameFile:/var/www/html/admin/modules/phonebook/Phonebook.class.php:161 Asterisk...
View ArticleHow to handle very high number of inbound routes?
So forgetting the number of DIDs, whats the best way to sift through a giant list of CIDs to route the calls appropriately?
View ArticleHow to handle very high number of inbound routes?
A custom AGI script would probably benefit you the best
View ArticleStrange issue when being put on hold
Hi all, I have a very strange issue. When put on hold by some called parties (does not affect all called parties, but can be reproduced with certain called PBXs) out local FreePBX play music on hold...
View ArticlePJSIP hold and pickup
I am planning on moving over from freepbx 12 to freepbx 13 this weekend. I currently use Devices/Users mode. We do this because we have extensions that appear on multiple devices. So from what I...
View ArticlePJSIP hold and pickup
mcg1103: if a person is on a call for an extension that is on two devices, can the put the call on hold on one extension and pick it up on another? No. mcg1103: If an extension is on a call if a...
View ArticleHow to handle very high number of inbound routes?
Pardon my ignorance but does this script need to be called in the additional.conf file or how do I grab the call before it gets processed normally?
View ArticleExtension Module not being displayed
My freepbx system shows that the extension routing module is activated and installed. But in the outbound routes I do not see it?
View ArticlePJSIP Trunk settings in Freepbx 12
I am having difficulty in that no caller id information is being sent as Remote Party ID on a pjsip trunk. I have the situation where a sip trunk to my voip provider carries several different numbers....
View ArticlePJSIP Trunk settings in Freepbx 12
@stonetLike I have already mentioned: Create a trunk in the pjsip_custom.conf settings with a context someting like mycustomtrunk and after in the freepbx gui: connectivity -> trunks -> add...
View ArticleDigium Addons issues, requires Asterisk 1.8.4+?
Hi, I have 4 licenses for G-729 digium addon. I updated FreePBX to the 12.0.76.2 framework running with Asterisk 1.8.25.0, Centos 5.11, and PHP5.3. When I go to the Digium site to get the digium...
View ArticleDigium Addons issues, requires Asterisk 1.8.4+?
Please file a bug report. The function they use for version checking only understands 1.x versions so it will fail on 10,11,12,13 etc...
View ArticleAfter module updates (10/2), constant red Error in right corner
After updating the the 3 modules tonight (10/2), every time I click to go to any other page, I get a re error in the upper right corner of the web gui. Never had it prior to the updates tonight. It...
View ArticleAfter module updates (10/2), constant red Error in right corner
Your error report is confusing so I hope you can provide some screenshots. You say every page. Then you say only when you go to dashboard.
View ArticleRecent update provides fax destination error?
So after the most recent update to the free PBX system I have this strange critical error that shows up. Inbound faxes now use User Manager users. Therefore you will need to re-assign all of your...
View ArticleAfter module updates (10/2), constant red Error in right corner
Sorry... Sometimes I do have a hard time trying to explain a visualization. Attached is a current UCP error. And... I cannot now replicate the little red error that came up on every page! ARGH! It...
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