1. B

    Ring Time for SIP Extensions

    Hello, I've read several on-line posts and forums on how to change the default ring time for SIP extensions. I've tried changing "Settings - Advanced settings - Ringtime default value to 5, but this didn't change the ringtime. I have also tried changing the " Extensions - my extension -...
  2. A

    QUESTION Ip phone problem with Elastix

    Hello everybody.. I have elastix server and nortel ip phone 1120e sip phone , I have created domain in ip phone :.. Server.. which is elastix server and it taked ip from dhcp server.. I made a ping from sever to phone and it replied... Also ping from phone to server and gateway and...
  3. shetu

    TIPS One way audio

    Hello I install chan dongle (Asterisk 13.23.1, Huweai e1550, firmware 11.609.20.03.356). Codec uLaw, g722, gsm I run traffic capture at Wireshark and it shows RTP send receive both way. But other caller dose not hear anything.
  4. ankyr

    FOOD FOR THOUGHT New SIP client for Android

    Hi! We're the developers of a new VoIP app, Sipnetic. If you use an Android SIP client, such as CSipSimple, Zoiper, or Linphone, you should check out Sipnetic. I'll be glad to provide any info about this app or SIP protocol in general. Google Play...
  5. W

    TIPS Securing Server and Removing knockd?

    I consider myself somewhat technologically abled but working with Incredible PBX, VPSs, and networking definitely is getting my hands dirtier than I thought it would. I have several endpoints that have dynamic IPs and it's not really feasible to have them connect through a VPN. (1) I am...
  6. S

    SOLVED Incredible PBX 13 Extension will not register

    Hi All, I just built a new Raspberry Pi 2 with Incredible PBX to replace an older version (PBX 11-3?). I did all of this on a second Pi so I could still have the old one up and running for reference. I gave the old Pi a new static IP address and the new Pi the static IP address of...
  7. wardmundy

    ALERT GV: The Sky Has Fallen... Really

    We've been down this Sky Is Falling Road several times before with Google Voice, but this time it looks to be real. Google has announced they are dropping XMPP support in mid-June. That's 5 weeks. All of the usual indicators that they may change their mind appear to be missing this time around...
  8. Charles Steiner

    TIPS DID provider w/ SMS for Mexico City

    I need a DID for Mexico City that also has SMS capability for the same number. Vitelity does not offer SMS for international DIDs. And other services had could not offer both on the same phone number. Anyone have any leads. If you are not 100% sure, go ahead and give me the name and I'll check...
  9. J

    TIPS How to refresh registration manually in 3CX PBX edition (Free)

    I recently had to make some changes to my internal DNS, which caused a small outage. During this window, it appears it knocked out the registration on the server. Now I can't seem to get the SIP registration to re-register. I've rebooted the server to try and force it, but it seems it just...
  10. S

    SOLVED Problems with both out and inbound calls

    Hi. Installed XIVO a few months ago, got it working with the mac Telephone softphone with Didn't use it much after basic testing, as I wanted to get a sip desk phone, plus figure out the automated attendant (that's still in the future). Today, went to install a sip phone and found that...
  11. S

    SOLVED AnveoDirect on Xivo - Incoming calls busy

    I have a recent install of Xivo that I setup with Google Voice (via Simonics) and AnveoDirect. GV works flawlessly as does outbound calling on Anveo. However, when I dial into the Anveo number I get a busy signal and this line in the logfile: NOTICE[1750][C-00000010] chan_sip.c: Call from...
  12. EcstaticMark

    QUESTION Caller Not Getting My 200 OK

    I'm losing sleep over this, so here goes. Short version: I can't answer inbound calls to my trunk's DIDs, but CAN answer calls on the same trunk to my ported DID. Asterisk Ver. 13.6.0, Incredible PBX 12.0.74. 1 Digium Trunk with 2 lines. 234-521-xxx1 and 234-521-xxx2. One 1 line, my home number...
  13. ChiefGyk

    NO JOY Avantfax not letting me log in

    So I set up a fresh CentOS VPS with IncrediblePBX, but Avantfax is not letting me log in on any browser with the new password I made. CentOS 6.7 VPS on Digital Ocean w/ IncrediblePBX 13
  14. P

    NO JOY Cannot make incoming calls from non-SIP devices

    Hello, I am able to make out-going calls to the real world(non-sip phones) and am able to call extensions with software SIP programs but I am not able to make incoming calls. I would say that devices that aren't registered to my PBX will not be able to have incoming calls For example, if I...
  15. J

    TUTORIAL Vitelity Trunk Setup

    Hi there, I'm setting up my PIAF Green and can't seem to setup my inbound trunk. Any help would be appreciated. Here are my current out and in settings. Thanks. OUTBound type=friend dtmfmode=auto username=XXXXXXXXXX secret=XXXXXXXXXX trustrpid=yes sendrpid=yes context=from-trunk=;XXXXXXXXXX...
  16. Jerrod Belvin

    GO HERE Cisco 7970 XML HELP

    Hello, New to converting phones over..... Bought Cisco 7970 SIP programmed phones. Does anyone have the Zip files to convert these phones? Or the correct XML files? I dont want to end up with paperwieghts.... We were going to purchase the FreePBX Endpoint manager but that shows unsupported...
  17. chris_c_

    NO JOY No audio 603 Decline Nortel 1535 SIP video phone

    Alert, this is probably a stupid question, but I'm tearing out my hair over this and can't seem to find the answer. Just got the famous Nortel 1535 SIP video phone !! No joy! Calls are dropping instantly with no audio AND 603 DECLINE sip message!! Details: The phone registers OK into the...
  18. chris_c_

    RECOMMENDATIONS Add support for

    Would be nice to add support in the Incredible GUI, for automatic free CA-signed TLS certificate for every incrediblepbx instance. These free certs are available from the new Certificate Authority,, are fully revokeable, and would work without complaint in the Incredible GUI web...
  19. B

    TRY THIS internal calling broken chan_sip

    Hello, Running Asterisk 13.6.0, Freepbx 12 on Incredible PBX ISO. I have a few phones registered to the system, which is running on a VPS. Using Pjsip, the phones can register, and internal calls pass successfully without issue. Using chan_sip, phones appear to register, and outgoing calls work...