Sorting Contact Manager Groups and Entries
@Bradbpw wrote: This may be more of a feature request, but I wanted to make sure I wasn't missing how to do this. In Contact Manager, I can't sort contact groups or entries. I would like to...
View ArticleCaller doesn't hear rings or person called when Fax detection is turned off...
@jtshows wrote: This one is a strange one, be warned.I have inbound route, based on DID. The route sends the caller directly to an extension (private phone number for each user).When the caller calls,...
View ArticleGrandstream display not showing Lines
@skip189 wrote: Hi everyone, this seems like a fairly simple issue but I cannot find a way to show the lines on the phone. Registered or unregistered, used to be in Green and/or red, and I could have...
View ArticleCaller can hear me, but can't hear caller
@veratronics wrote: Hi, I managed to setup the trunk. And for the inbound route I used the default setting and set one of the extension as the destination. The calls come through but, I can't hear the...
View ArticleCan't start UCP Node
@avayax wrote: UCP node not running and I can't start it on an HA system via fwconsole start ucpnode.Starting UCP Server...[--------------->------------] 10 secsThe process did not stay running....
View ArticleCustom Context for sets of extensions
@Atechhito wrote: We are trying to semi-isolate sets of extensions on our server using contexts. So lets say we have the following sets of extensions: 100-199 (context=alpha)200-299...
View ArticleHA question with regards to upgrades/asterisk changes
@mvogel4949 wrote: I am about to deploy my first HA installation and I had a question. I have looked at the wiki but am looking for some clarification. It appears that with HA if I go to the floating...
View Article"Follow Me" not honoring trunk sequence?
@jrothschild wrote: Asterisk certified/13.13-cert1 + FreePBX 13.0.190.19 + CentOS 6.8 (fully updated). Server is configured with 2 trunks: Trunk #1 ("Wiline") is the primary SIP provider Trunk #2...
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Dialling a PJSIP extension with 2 registered contacts using TAPI
@Gerald wrote: I have a FreePBX Installation (on Distro v7 but I don't think this matters) and am trying to do TAPI dialing with PJSIP extensions. I've created PJSIP extension 12 and increased the max...
View ArticleMulticast Paging Issues with Digium Handsets
@jdupuis wrote: I am having a very difficult time getting multicast paging working properly in freepbx13 with Digium D60 handsets. I tried creating a custom extension using exten =>...
View ArticlePhonebook module
@UnknowPerson wrote: Hi, I am looking information is there way to make many different phonebooks? I want to create one phonebook which include 5 person and these extension and second where is only few...
View ArticleSip trunk issue
@busster8 wrote: Just wanted to mention something that happened yesterday on an installation.Installed latest stable, 10.13.66Created a sip trunk to a known service provider. Outbound calls would not...
View ArticleQueue & IVR Behavior
@FN_GM wrote: Hello, I have setup a Queue in FreePBX. I have the retry option set 1 second, so if someone is in a call, they put the phone down and their phone rings with the next person in the queue....
View ArticleDo Not Distrub no red light and error of no ext or hint
@derickmain wrote: FreePBX 13.0.190.19Asterisk 13.13.1Do-Not-Disturb (DND) 13.0.3 I am currently having an issue with DND not showing up on my polycom vvx 310 phones. I have disabled the hardcoded DND...
View ArticleMitel 6869i 50 BLF Maximum
@chrischevy wrote: Hi, I am setting up a Mitel 6869i phone with 3 M685i expansion modules.It seems like I cant get the phone to monitor more than 50 extensions with BLFs. The first 50 BLFs are fine...
View ArticleBlank cdr; cel is fine
@braden wrote: Hi - for some reason calls stopped being logged in the CDR on Jan 21 (i checked the cdr table) but information is still being logged in the CEL. Any recommendations on what to check?...
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Dial pattern matches problem
@newtask wrote: I'm running the FreePBX 13.0.190.11'VoIP Server'. I created one trunk and uses the SIP provider for international calls. I am able to call to USA and India's region but unable to call...
View ArticleAsterisk under Double NAT
@Codrut94 wrote: Hello, I am having a difficultysetting up the Asterisk NOW server for use over the internet.To be more specific , My server is behind a router (which gives a 192.168.1.x /24 address)...
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Inbound calls not working on Grandstream GXW4108
@alexvivas wrote: Hi I'm trying to configure a grandstream gxw4108 version 1.4.1.5 with Freepbx version I'm settting FXO port 1 in gxw4108, I followed these guides for doing the configuration:...
View ArticleSnome Phone Setup
@iceget wrote: dear community, have one question:if i have setup my asterisk correctly with extensions now.how i should now setup my snom phones directly, without auto provisioning? i mean which "most...
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