Listing prosody groups in XMPP client such as Pidgin
@jlayton wrote: Hello. I have set up a FreePBX server and am in the process of setting up Prosody to use as our XMPP server. I have been able to use Pidgin to chat back and forth with a coworker and...
View ArticleVoice mail stop sending from system
@jchen wrote: Our FreePBX system stop send voicemail through email, all the setting is keep the same, anyone can help me troubleshoot this problem? Thanks! Posts: 2 Participants: 2 Read full topic
View ArticleIntermittent Incoming Call Failures
@Florg wrote: Hello all, We're using freepbxhosting.com and just recently added a new retail location to our server, adding approximately 13 new extensions effectively doubling our extensions to a...
View ArticleFirewall Rules corrupted - over and over
@ris2 wrote: I decided to re-visit an issue I had updating the firewall module. I "gave up" in the past by reverting to an older version, I but I really prefer to keep everything up to date,...
View ArticleHowto remove opencnam warning?
@bgroper wrote: Hi forumJust installed a fresh version 10.13.66-13.Applied all updates and starting to make a brand new configuration.No trunks, routes, or extensions created yet.BUT I get the...
View ArticleIssue using voicemail pound (#) return to IVR
@orb wrote: I am using FPB12 and Asterisk 11. I have an IVR that allows for direct extension dialing. Also return to IVR is checked. However, when a user gets into voicemail and hits # to return to...
View ArticleThe number you have dialed is no longer in service
@sentinelace wrote: I am getting this error every other day. Sip provider is Voip.MS. I have allow guest to on. I am able to make outbound calls but not incoming. logs show: 2016-08-04 08:44:03]...
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Phones dropping offline
@2024301 wrote: I've just upgraded to 13.10.0, running Asterisk 13. Running a small server at home with Grandstream phones. When the server is rebooted all phones connect and are shown as online in...
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Add DataBase For Automatic inbound Calls
@amdr wrote: hello evryone, I can attach a database to elastix,For automatic calls? tnk for all . Posts: 3 Participants: 2 Read full topic
View ArticleWith two T1 cards can Asterisk be "invisible" and pass T1 info in one and out...
@ghurty wrote: Is this doable? I have a an old T1 based phone system that I cant replace currently. The time conditions on it is broken and I need after hours the calls to be forwarded to a different...
View ArticleCan anyone help me
@dsbsaul wrote: Hello I'm new with this stuff of VoIP currently I had a freepbx installation working but I'm having issues I think due to a bad configuration. I have one freepbx working in America...
View ArticleAsterisk restarts issue
@eam12 wrote: Hi i am running latest version on Freepbx with asterisk 13.5.0. i experienced issue today when logged into freepbx asterisk up time was 2 hours from the full log file i see this before...
View ArticleMySQL Database in freePBX
@telili wrote: Hi, Is there a command line to check if the MySQL databse works well in freePBX . I think i have a problem. Thanks ! Posts: 3 Participants: 2 Read full topic
View ArticleHow to limit inbound calls per extension?
@madman91 wrote: Asterisk 11 and 13 - same problem. Queue config:...
View ArticlePaging Phones Restarts Asterisk?
@mvogel4949 wrote: I have a page group setup to page 19 phones (below the limit) It doesn't happen everytime but quite often an all page causes asterisk to restart and drop calls. FreePBX13, Asterisk...
View ArticleDefault group setting GUI
@telili wrote: Hi,Someone could point on that kind of issue that FreePBX for trunk configuration keeps its default group setting ? Group round robin acsending 0 even when i switch to Group round robin...
View ArticleSomeone Trying To Hack Me - Intrusion Detection Not Picking Them Up
@eagle wrote: Someone is trying to hack SIP registrations into my new FreePBX, shouldn't Intrusion Detection be blocking them? When I look in the Intrusion Detection I can see it blocked one IP...
View ArticlePJSIP trunk lost connectivity
@dbareiro wrote: Hi all! I am currently doing some testing with a PJSIP trunk on FreePBX 13/Asterisk 13. The connection is against a SIP provider. I have also configured the server of this provider in...
View ArticleApp_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber...
@mst wrote: Dear Experts, I have FreePbx 2.11 with accessone trunk From time to time I have problem with dialing out and noticed a lot of these:WARNING[4028][C-00005d05] app_dial.c: Unable to create...
View ArticleGrandstream GXW-4108
@bhfisher wrote: Have any of you deployed a Grandstream GXW-4108 on asterisk? Before I purchase one these, I have a question: I need to be able send a Flash() to get new dial tone after I've made one...
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