DAHDI audio problem with analog OpenVOX A400E FXO card
@grekasa wrote: Hello,I have a FreePBX Distro v10 (Asterisk v13) 64bit and an OpenVOX A400E with one FXO & one FXS cards.I am using the FXO card to interface my IPPBX with my PSTN phone number...
View ArticleDead air calling voice mail from phone to freepbx
@lortech wrote: Phone is registered and using very basic switch/routers like a linksys or d-link and when calling voicemail get dead air. Get this error on the screen. Setting change in freepbx?...
View Articlesip_poke_noanswer: Peer 'XXX' is now UNREACHABLE!
@rsarceno wrote: Asterisk 11.21.2Freepbx 13.0.79 I have three chan_sip extension. It was working until I reboot (VPS) server When I tried to reboot the phone, It will register and stay online for...
View ArticleCan't stop call recording
@JSBecker wrote: SETUP:Currently we have Extensions set to "Yes" for Call Recording and certain Outbound Routes set to "Never" for Call Recording. GOAL:Out goal is that if an Agent warm transfers a...
View ArticleInbound route for Withheld numbers
@RichE wrote: I want to create an inbound route that catches withheld numbers and sends them to an annocement, on the inbound route CallerID Number there are options for unkown, private, blocked but...
View ArticlePhonebook/CID Superfecta Problem with international phone numbers
@jst68 wrote: I have got CID Superfecta working fine for local and national phone numbers, but I am struggling with the setup for international phone numbers. Basically, the incoming number comes in...
View ArticleCall forwarding in custom extensions
@nikak27 wrote: Hello everyone, I have freepbx server with custom contexts for every extension and I want to turn on call forwarding (*72) but when I type *72 from sip phone asterisk says that this...
View ArticleNeed To Send Call Records to Call Accounting software with Proper E.164 Format
@RalphD wrote: I have calls going to a Class 4 switch then to a call accounting server how can I set trunks to send the format requested "the international destination number in proper E.164 format...
View ArticleYealink warm transfer
@gwntc wrote: We can't seem to warm transfer with the T48G yealink phones. Incoming call is answered, we press transfer, press the extension of the person (wait), talk to the person at the extension...
View ArticleChange outbound CID on the fly
@Bradbpw wrote: Am I able to change my outbound CID on a per-phone-call basis? Maybe by dialing a PIN at the beginning of the call? Posts: 3 Participants: 2 Read full topic
View ArticlePBX and Active Directory
@jonos wrote: Hi Everyone, I'm using non-distro freePBX 13 on Centos 7. I don't have option to auth with Active Directory. Is that feature available only for Distro or I'm missing something? If so, is...
View ArticleAsterisk behind Fritzbox
@TheNetStriker wrote: I'am new to Asterisk and I' am trying to use it behind my Fritzbox. I found several threads regarding this, but nothing helped so far. I've installed AsteriskNow and I'am using...
View ArticleSIP INFO Ignored by Asterisk 13 - PJSIP
@shauberg wrote: When configuring a PJSIP trunk with a provider that uses SIP INFO requests for keepalive purposes the call gets torn down by the provider after a number of SIP INFO requests which are...
View ArticleCan't Apply config after upgrading modules, error 255
@Lumute wrote: Hi, After being away for a month or so connected our FreePBX 13 appliance and noticed 43 modules available for online upgrade. Went to the Module Admin and did the Upgraded All /...
View ArticleBulk Extensions And BLF
@Pandera wrote: Hi Everyone, I have uploaded more then 100 extensions using bulk extensions.the extensions can register without any issue, but for some reason the BLF is not working.i have deleted one...
View ArticleSwitch to Asterisk 13 before switching to FreePBX 13 or after?
@mvogel4949 wrote: Is it recommended to switch asterisk versions prior to a freepbx upgrade to 13 or after? Thanks Posts: 1 Participants: 1 Read full topic
View ArticleFreePBX Usermanager and Bulk Handler questions
@WB3FFV wrote: OK, I have upgraded some systems from FPBX 11.x to 13.x, and that now means I have usermanager to handle access to things like UCP and so on. After the conversion I ran into issues with...
View ArticleCall Waiting on Sim Card in Portech
@stee89 wrote: Hello everyone, I configured a PBX with a FreePBX Portech MV-378. In sim cards have activated call waiting, you can handle it from the exchange? thank you! Posts: 1 Participants: 1 Read...
View ArticleHow To Load Asterisk 13 as asterisk instead of root on CentOS 6.7 & FreePBX 13
@kdwycha wrote: I have followed the following FreePBX how-to on installing FreePBX 13 on CentOS 6.X. (Link omitted since I am a new user and will not allow me to post a link) All is well and...
View ArticleSymlink from modules failed
@assos40 wrote: PBX Firmware: 10.13.66-9Asterisk 11.21.2 Hi,After updated about 15 modules i get this on the dashboard "retrieve_conf failed to sym link: /var/lib/asterisk/bin/check_portal.php from...
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