How to bulk modify incoming routes
@byteguy wrote: I have recently installed and setup Superfecta and now need to enable it for all our DDI incoming routes. Since we have about 1000 incoming routes doing this via the GUI would be a...
View ArticleIncoming call not always in
@jackyzhb wrote: I have one small setup and incoming call is strange, and it only happen sometime.When I dial the DID, some time ( it only happen some time ) I will get "the call can not completed...
View ArticleHow to Remove SDP from 183 Progress
@Randy_F wrote: ATT Cell callers calling in do not hear a ring, Net provider tells me we need to remove the SDP profile from the 183 Session Progress , where and how do I do that, have not found the...
View ArticleCertain calls not recording
@RobertH_PCM wrote: Server EnvironmentFreePBX 13.0.169Asterisk 13.10.0Centos 6.6 I've been having an issue with call recording. Most calls are recorded, however, calls that are blind transferred are...
View ArticleUpdated FreePBX install instructions for Chan-SCCP-B
@cynjut wrote: I just updated the Chan-SCCP-B Wiki page for FreePBX. It is here - FreePBX Chan-SCCP-B Driver I updated it to clarify a couple of steps and to add some new insights - specifically:-...
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High CPU load spikes from mysql
@wooshell wrote: Hi,we're running FreePBX 11 for several years now, but over the last few months we're seeing growing spikes of CPU load coming from MySQLd, that recently started reaching 100%, and...
View ArticleArgument not passed when calling tts.php from an AGI
@jerryriggin wrote: I'm trying to do some dynamic TTS via an AGI, which involves calling tts.php included with the freePBX distro from within another AGI. The following dial plan works exactly as...
View Article/var/www/html/admin/views/config.php
@mjurcev1 wrote: Since of the last updates a new file is showing up, that should not be there: /var/www/html/admin/views/config.php It is Base64 encoded PHP, containing malicious code evaluating...
View ArticleEditing Conferences default PIN timeout
@mjurcev1 wrote: This is how FreePBX is generating the conference room in the extentions_additional.conf, especially line defining PIN input:...exten =>...
View ArticleBLF with FreePBX and Yealink T46G
@tlarrea wrote: I've recently installed FreePBX 13 and am having problems getting BLF to work. I'm trying to have extension 281 be able to monitor and pick up extension 237. I've added ext 237 to a...
View ArticleHow to reject calls over the first one?
@frankge973 wrote: Hello.I wish to reject all incoming calls with a busy tone if at least one trunk is in a call, this because the office is small and if a people is making a call, it is annoying that...
View ArticleRed Login cog on all servers
@cyberco wrote: We have noted today that all servers when logging in have the cog as red and an ERROR! red box pops up when using chrome Version 52.0.2743.116 (64-bit) have also updated to Version...
View ArticleRoute incoming calls to multiple servers with device and user mode
@gav wrote: I've not been able to find much information when Googling, and I'm not sure it this is even possible. I was wondering about the feasibility of having several Asterisk servers using device...
View ArticleFreePBX hacked voicemail?
@RKFAULHABER wrote: Hello,We’re running a FREEPBX system (12.0.76.2) for our company. Recently, one of our users, who has an 800 number that is linked to his extension, has started receiving calls...
View ArticleT.38 Fax with PJSIP fails
@mlaihk wrote: I am transitioning my VOIP trunks to my provides from SIP to PJSIP. For voice communications, they work very well. However, there is one trunk that we use for T.38 fax and when...
View ArticleAdmin user lost after modules upgrade
@esarant wrote: Hello, I am not sure if anyone is having the same problem but I have it in 3 of my FreePBX installations. The default user "admin" (the one created during installation) has been lost...
View ArticleCDR problem after update
@sinij wrote: Hello,After recent update FreePBX I can`t play recording calls. When I click the button play in CDR I getting error:"RuntimeExceptionsh: 1: asterisk: not found...
View ArticleVoicemail on request
@claloano wrote: voicemail on request I have a client that a particular bit .... : He wants that when you call an extension to another extension will continue to ring for a long time ....and until...
View ArticleRegistration fails with incomplete SIP command
@mynamewastolen wrote: I have a user connecting to our FreePBX but they are showing up in the log as sending an "Unknown command" This is the line in the log: NOTICE[29605] chan_sip.c: Unknown SIP...
View ArticleHow to upgrade from asterisk 11.22 to 11.23?
@jronald wrote: I am running FreePBX 13.0.188.2 and need to update asterisk from 11.22 to 11.23 do to "https://issues.asterisk.org/jira/browse/ASTERISK-25888". Do I update Asterisk from within FreePBX...
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