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Asterisk Session Timers!
@senate014 wrote: Well... this is an interesting one I've been battling with all weekend that I thought I'd share with you.... I've got a couple of FreePBX customers that use to have their phone...
View ArticleFollow Me not working with RCFed numbers
@striker495 wrote: We have recently had our old numbers RCFed to the new native numbers that are on our SIP service. When I place a call to one of the new native numbers that is routed to an...
View ArticleDiversion Header, and From field
@iceget wrote: dear community, i have a problem:my provider only allows to send the original caller id (on call fowarding),if i set a correct diversion header like +49XXXXXXXXXXXXX. currently if i set...
View Article911 Routing
@kbocek wrote: Is setting the emergency route attribute enough to successfully pass 911 calls or do I need to add a dial pattern as well? Posts: 2 Participants: 2 Read full topic
View ArticleAnalog door buzzer options?
@sentinelace wrote: We are replacing a nortel system. There is a buzzer at the front door that dials an extension. The system was digital and used 4 wires. What can I put here? My thoughts were to use...
View ArticleAsterisk not ruuning
@RichE wrote: Came in yesterday to my phones not registered and my box saying Asterisk not running in red in the top corner. Nothing has been changed or altered with for a lone time, system does have...
View ArticleProblems after restore PBX settings backup on the new server
@Vik89 wrote: Hi there, We set up a FreePBX 14.0.1rc1.7 installation on virtual machine and pre customized it for a real server - extensions, settings, audio messages, gateways. We then back up data...
View ArticleFreePBX not Responding Intermittently
@palsandy wrote: Hello,I have two SIP Trunks and 100 DID numbers from DIDForSale.com. Recently I have been noticing intermittent call failures. I see the signal coming to my Freepbx, it send 100...
View ArticlePlay a recording over the paging system?
@tonyg wrote: hi, i have a customer that I am replacing an old phone system for with freepbx. One of the challanges I have is, on the old pbx, the customer has a phone on the dock for deliveries. the...
View ArticleMacro-hangupcall-custom macro-hangupcall problem
@lukass wrote: Hi, In my FreePBX it looks like that:/etc/asterisk/extensionextensions_additional.conf [macro-hangupcall] include => macro-hangupcall-custom exten =>...
View ArticleTrunk Failover when subscriber absent
@Keithc wrote: I have two trunks specified for an outbound route, if one returns "Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)" it doesn't failover to the next trunk? Is this...
View ArticleFreePBX switched IP Address?
@mike_b3 wrote: Yesterday I had a weird thing happen. It is running now, but I am wondering what could have happened. So: I have an Asterisk system with FreePBX The system sits behind an Xfinity cable...
View ArticleNo audio on inbound calls
@azook wrote: FreePBX server [v:13.0.191.11] sits behind a firewall.A public,static IP address [123.123.123.123] has been forwarded to the internal address of the server without NAT at the...
View ArticleAudio dropped after a random time
@piero wrote: Hi, the last month i'm having a big problem with the calls: after a random time, some RTP packets are dropped, losing seconds of the call.I checked on the SIP provider and the virtual...
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Aastra/Mitel Remote Phone Setup Help - This should be an easy one ;)
@Shandley wrote: Greetings! I have a Mitel 6867i that I am trying to remotely connect to my FreePBX server as extension 4010. It seems to register with Asterisk but all calls in or out fail. **PHONE...
View ArticleCan't hear incoming caller's voice
@jyost wrote: When people call from outside we cannot hear their voice. They can hear our voice but we cannot hear them. I have seen this issue before but the problem has usually been a firewall...
View ArticleInbound callers get "the number you have dialed is no longer in service"
@azook wrote: FreePBX server [v:13.0.191.11] sits behind a firewall.A public,static IP address [123.123.123.123] has been forwarded to the internal address of the FreePBX server at the firewall.All...
View ArticleUnable to register any SIP Phone
@pvogt wrote: I have successfully built my FreePBX server and set up my SIP trunk. I'm confirming it's working because when I dial the DID on a cell phone, the server goes into the IVR menu I set up...
View ArticleWhat is difference Between PSTN and VOIP in Asterisk
@kpvsprasad wrote: I am very confused over usage of PSTN and VOIP-SIP please clarify it. Posts: 2 Participants: 2 Read full topic
View ArticleException when running "apply config"
@dandrzejewski wrote: When I attempt to run "apply config," I get the following stack trace: Whoops\Exception\ErrorException: Non-static method connectdirs::create() should not be called statically in...
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