Tap OPENVPN Tunnel to FreePBX Server with NAT
@mani9876 wrote: Hello, I am trying to connect some phones which are outside my company network.I've set up an openvpn tunnel ( bridged ), if I open the tunnel on a notebook, I get an IP Adress inside...
View ArticleProper firewall rules for security 5060 and 5061
@ray123 wrote: I know this has been asked before many times but all posts say close port 5060 to the external world. Would that not also block the trunk as well? I was having a problem with SIP...
View ArticleFactory resetting Polycom 331 no luck
@sentinelace wrote: I got some polycom 331 donated for a church project. I need to be able to logon to them to point to the freepbx box. We do not have a DHCP server except the local router. When I...
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PJSIP Trunk one-way audio problem
@zirophyz wrote: Hi all, Not what you think this is - usually, one way audio doesn't defeat me but this looks a little non standard. The advantage is that I am also the SIP Provider, and PBX...
View ArticleCall disconnected if queue full
@dennislittle wrote: Hi, First hello to everyone on the forum i am just new here. We are experiencing the following problem. We have a queue with 5 agents. If all the agents are busy or not available...
View ArticlePlease help!
@nbhelms wrote: Since I upgraded to v13.0.131 all calls that come from my T1's just get a busy signal. Dahdi starts with no errors. Our sip trunks to vitelity are working. Please help we are unable to...
View ArticleFreePBX PJSIP RTP Issues with NAT endpoints
@kadarlevi wrote: Hello, I am currently running: PBX Firmware: 10.13.66-12 Asterisk 13.9.1 I have the PBX in a data center behind NAT. I have a soft phone in my house behind NAT as well. The...
View ArticleTime conditions?
@jjczon wrote: Tried to set up time conditions as I have a number in a +7 time zone and want to enjoy my sleep as well.But it seems to be that the rules destination if time matches and don't matches...
View ArticleIntrusion Detection Whitelist and iptables
@jos50 wrote: In my System Admin > Intrusion Detection > Whitelist I have the following IP127.0.0.1192.168.0.0/24192.168.44.0/24192.168.55.0/24192.168.33.0/24so that I can have SIP ATA in those...
View ArticleCisco-9971 Setup
@josephchrz wrote: Hello not sure where to put this at. But I got a new phone in today from a friend of mine a Cisco-9971 I'm trying to program it with my freepbx setup. I'm not sure how to get into...
View ArticleCaller ID not showing numbers
@modcar wrote: Hello; In CIDLookup, I set source type 'internal' which displays the name of the caller, if they are in the Asterisk Phonebook - which is great My issue is, if someone calls, whos not...
View ArticleReset FreePBX Admin Password
@itmonitor wrote: Hi! I had installed FreePBX 2.11.0.43 on Raspberry some time ago, but had scarce time to configure it due to other priorities. I am now stuck at the Admin login because I have...
View ArticlePolycom phones DHCP option 66 Won't register - will donate for help!
@sentinelace wrote: I am trying to provision some polycom ip 331 phones that were donated to us. Freepbx sees the phone and the phone gets an IP address. I have configured PFsense with with option 66...
View ArticleCisco-9971 XML
@josephchrz wrote: Hello I'm just learning how i setup the Xml file for the TFTP and one thing i found online is the XML file so when i searched online i found one but it is set for Europe And I'm in...
View ArticleHow to setup IVR
@nrkarthik wrote: Hi I am new to asterisk, I need to setup ivr and route all inbound calls to IVR. Could you please help me to to that. thanks in advance Posts: 7 Participants: 2 Read full topic
View ArticleRecovering root password
@cgallery wrote: A friend recently passed away, and we've been trying to put-together login information for his IT stuff (he ran a small services business). We have most everything we need, but the...
View ArticleCan Dialled DID be sent to device?
@voipmuch wrote: We have customers with multiple DIDs, all associated to the same device. When a call comes to the device, the invite shows up as DEVICE@URLThere is nothing in the invite or any other...
View ArticleUse "Display Name" from linked user as extension's CallerID?
@jakal wrote: Hi. Can I use "Display Name" from linked user as extension's CallerID? And how?Thanks. Posts: 1 Participants: 1 Read full topic
View ArticleSuddenly, call flow routing doesn't work thus no incoming call
@digiteltlc wrote: My inbound route points to call flow control 0 (there is a second one configured)This morning , suddenly, no incoming call are possible, with voice playing ss-noservice message.To...
View ArticleOutbound SIP response 480 "Temporarily Unavailable" (only certain carriers)
@jimvman wrote: Hello again everybody,I'm running FreePBX with asterisk 11.13.1. Over the past several weeks I've been having periodic outbound call issues. After diagnosing for quite some time, it...
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