FreePBX repo cert issue
@psdk wrote: Hi, When I want update some modules, I receive this error : Warning: Cannot connect to online repository(s) (http://mirror1.freepbx.org,http://mirror2.freepbx.org). Online modules are not...
View ArticleSIP Registration Timeout Not Changing on the Web Interface
@ynagy wrote: Hi there,I have updated to FreePBX 13 recently. For some reason I couldn't register couple of my trunks any more with a specific provider as they restrict the SIP Registration Timeout to...
View ArticleBlacklist Power
@BARDINET wrote: I should be interested in FreePBX Blacklist module, to use it to redirect calls for a small company . They just want an unknomn caller to be routed to normal incoming route, but a...
View ArticleMake an extension only receive calls from another extension
@andrewd wrote: I'm using the latest FreePBX and I have a situation here. I have an outdoor siren at Extension 3523 on a system I am putting in and I have an emergency notification system at Extension...
View ArticleRing Group after Upgrade to 6.12.65-32 Outside caller hears no ring
@RalphD wrote: After upgrade to 6.12.65-32 when a ring group is pointed in the incoming route the outside caller hears no ring we set it to music and works. We did see that if you activate Signal...
View ArticleUsed to have the Name in Queues - Now I just have the Extension
@GSnover wrote: Starting to see this when users log into the Queues: Local/682@from-queue/nOmar Llanos (Local/612@from-queue/n from SIP/612) So some logins show the name properly and some show just...
View ArticleRemove Congession Tones from macro-outisbusy
@dawho9 wrote: Just looking for clarification to the best way to accomplish something. I have an Asterisk/FreePBX that is sitting between two SIP Trunks doing some B2B termination. Inside of the...
View ArticleCrosstalk/Packet Loss
@dave69s wrote: I wonder if anyone can provide suggestions for me. I'm running on Freepbx 12.0.76.2, with Asterisk 11.18.0 and am having some audio issues. Everything was fine until about a week ago....
View ArticleIssues with AsteriskJava
@choiceit wrote: Hi guys! Please let me know if this is not the correct place to ask for help with this! I am trying to get an application called Zoho Phonebridge working. Currently, making outgoing...
View ArticlePaging groups
@ekohler wrote: We would like to set up emergency paging. I noticed that the suggested page group number is 40 extensions. Is there a module , free or commercial, that will page a group of 40 phones...
View ArticleEvent to trigger the type of DTMF code
@claloano wrote: I am facing a new need I have to catch a code typed in handset and trigger an event as soon as you type I think I need to switch this...
View ArticlePanic button solution
@mcdouglas wrote: Hi, I'm looking for an in house panic button solution with IP phones. I was thinking about assigning a fast dial key on the phones to a designated extension, but I ran into some...
View ArticleError on adding Device?
@radicall wrote: Can you please advise what the possible cause is of this error and how to fix it. With the PBX in User/Device mode the system throws an error to say the Mailbox is not properly set up...
View ArticleInclude SIP Trunk headers into CDR
@afallison wrote: Hi everyone. I'm attempting to save two headers that are sent from Twilio's SIP trunk to our CDR. X-Twilio-CallSid X-Twilio-RecordingSid Incoming Route[from-pstn-custom]exten =>...
View ArticleApply Config causes IAX2 trunk to become unreachable when using Asterisk 13...
@jehrich wrote: Background I'm setting up a new (virtual) FreePBX system with the following Latest version of FreePBX (PBX Firmware: 10.13.66-9) Asterisk 13.7 IAX2 trunk to VoIP.ms account Every time...
View ArticleIs it possible to have followme pass DTMF?
@FreeSeb wrote: I am using followme, and once the takecall digit is pressend to accept the call, the channel is bridged and both parties can communicate. However DTMF are not going through. Is there a...
View ArticleOne way Audio on second call - draytek vigor 2860 router
@al2lo wrote: HI, I have noticed that when i call in from VPN the call works okay. If another call is in progress then there is no audio on second call. All calls ring and then no answer so i assume...
View ArticleHow To Connect SIP Phone To FXO Line
@kbocek wrote: I've got two HT-503s with their FXO ports configured and have the trunks, inbound and outbound routes setup. There are lots of pages out there talking about setting up trunks and I/O...
View ArticleFreePBX repo cert issue
@psdk wrote: Hi, When I want update some modules, I receive this error : Warning: Cannot connect to online repository(s) (http://mirror1.freepbx.org,http://mirror2.freepbx.org). Online modules are not...
View ArticleSIP Registration Timeout Not Changing on the Web Interface
@ynagy wrote: Hi there,I have updated to FreePBX 13 recently. For some reason I couldn't register couple of my trunks any more with a specific provider as they restrict the SIP Registration Timeout to...
View Article