Share SIP trunks
@mcg1103 wrote: Is it possible for a FreePBX install to share inbound and outbound trunks on a different FreePBX system? If it is possible, is there a tutorial on how to do this? Can they do extension...
View ArticleSimultaneous calls out on specific route
@claloano wrote: simultaneous calls out on specific route I saw that you can restrict calls for simultaneous trunk But if you want to limit outgoing call on a specific route that files you have to...
View ArticleCallForward Ringtimer setting doesn't work from UCP
@avayax wrote: Setting the call forwarding ring timer in UCP to something other than the default 15 seconds before it goes to voice mail. Unfortunately, my settings don't get applied, whatever I set...
View ArticleUCP Node is not running
@avayax wrote: My dashboard says my UCP node is not running.How can I start the service without an amportal restart? FreePBX distro 6.12.65-29 Posts: 4 Participants: 2 Read full topic
View ArticleSpecifying Outbound Route
@mohammed_kakuji wrote: Hi, We currently have 3 SIP Lines within 3 1 PBX system. Currently all outbound calls go through the first outbound route. Is there any way to specify which extensions go...
View ArticleUser Control Pannel: cannot change Voicemail greetings
@arnoutvdk wrote: Hi, A while ago I posted a bug record in the bug tracker: 1 - I created a user and gave him permissions to set custom voicemail messages2 - I logged in as the user in UCP3 - I...
View ArticleReload failed because retrieve_conf encountered an error: -1
@dcorp wrote: Hello I have FreePBX 12.0.76.2 and asterisk 13 installed, this error would happen to me from time to time when I hit the Apply Config button, this is a full stack trace : eload failed...
View ArticleCisco 7960 Incoming Call Problem
@uberamd wrote: Hi All! I'm fairly new to FreePBX and am using this is purely a learning experiment. I'm using a trial of SIPStation and when using a pjsip softphone I'm able to make and receive calls...
View ArticleSeveral extensions showing UNKNOWN status in CHAN_SIP PEERS
@munozj wrote: Using FreePBX 13.0.26, I have about 5 particular extensions that are showing up as UNKNOWN in the show sip peers report. These users are complain that they are missing calls. Sometimes...
View ArticleCompare the estimated hold time and max wait time for going to failover...
@peter59 wrote: Hello, I want to know if it's possible to use the estimated hold time in order to compare it to the max wait time? If the estimated hold time is superior to the max wait time so we are...
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No audio on both inbound or outbound calls
@dcorp wrote: Hello I've installed 12.0.76.2 and Asterisk 13 on my centos 6.7 machine, and everything seemed to be in order as I followed the official install guide, with some small issues here and...
View ArticleIAX2 Extensions and auto answer header
@avayax wrote: I am using an IAX2 extension on a Zoiper softclient. I would like to send an auto answer header to my extension. Can I send auto answer headers in IAX2 at all? Posts: 1 Participants: 1...
View ArticleHelp Understanding Version Numbers
@munozj wrote: So i recently upgraded to version 13 using the upgrade module in the web gui. I'm still on Asterisk 11 which is causing some issues with CDR's and User Manager. I was going to try and...
View ArticleError upgrading ucpnode and voicemail modules on v.13
@adolfoc wrote: I'm using FreePBX 13 and for some reason I'm trying to upgrade the ucpnode and voicemail modules and get a similar error. Downloading and Installing ucpnodeDownloading ucpnode Error(s)...
View ArticleAzure freepbx behind nat doesn't register extension
@letal1609 wrote: Hello, I have a freepbx installed on a azure virtual machine. The server is behind a nat but it doesn't register sip extensions. It receive the request tough. I've checked that with...
View ArticleDegradation Functionality Asterisk
@korda wrote: Hello.We have a FreePBX Distro. Centos install on VMWare 6. At more than 150 simultaneous calls are seeing a problem with Asterisk. Calls for a long time are processed, do not always...
View ArticleSetup Asterisk and Turn server on the same system?
@dineshvg wrote: Hey,I'm pretty much new to Asterisk and also to SIP. I'm trying to setup Asterisk and connect to remote extensions. But my extensions are behind a symmetric NAT. So, I have a docker...
View ArticleVmX mobile transfer hangs up and long delay before IVR kicks in
@godinperson wrote: Two issues here. 1- I've setup a VmX locator to reach me on my mobile if 1 is pressed during voicemail. On one extension, it works while on the other, it hangs up [2015-12-08...
View ArticleWebrtc will not load
@eam12 wrote: I am running FreePBX 13.0.27 debian 8 installed the webrtc module and allowed under user manager however when i login using chrome to UCP no webrtc is loaded any ideas why ?? Posts: 7...
View ArticleDial through asterisk/freepbx
@gw22 wrote: Running a 5.211.65 distro and want to be able to pass calls through the Freepbx system to the Norstar, and if no answer then back for voicemail, then vmail to email. I dont care about the...
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