"Initials" by "Florian Körner", licensed under "CC0 1.0". / Remix of the original. - Created with dicebear.comInitialsFlorian Körnerhttps://github.com/dicebear/dicebearVO
VoIP 11mo ago
Jump
SkySwitch implementations, traversing firewalls
  • "Initials" by "Florian Körner", licensed under "CC0 1.0". / Remix of the original. - Created with dicebear.comInitialsFlorian Körnerhttps://github.com/dicebear/dicebearTH
    TheRealNalaLockspur
    11mo ago 0%

    Skyswitch. Gross. That MSA is a disaster.

    TLS + a good SBC, you’ll never think about firewalls again.

    I haven’t touched a customers firewall in over two years. Albeit, I am writing my own platform lol.

    0
  • "Initials" by "Florian Körner", licensed under "CC0 1.0". / Remix of the original. - Created with dicebear.comInitialsFlorian Körnerhttps://github.com/dicebear/dicebearVO
    VoIP TheRealNalaLockspur 11mo ago 100%
    [Dev Only] Freeswitch Click To Dial - USER_NOT_REGISTERED

    I am in the process of doing my Zoho PhoneBridge integration and I am trying to get Click to Dial working. Well, I do have it working, but only for physical phones and the mobile app. WebRTC (Sip.js) is having an issue. I am calling the originate command through ESL using node. Click to dial works great if I have a Yealink or my mobile app registered, but if I only have the web app registered (Sip.js), then Freeswitch console throws: 2023-11-28 14:04:42.951275 97.17% [NOTICE] switch_ivr_originate.c:3049 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2023-11-28 14:04:42.951275 97.17% [DEBUG] switch_ivr_originate.c:4045 Originate Resulted in Error Cause: 806 [USER_NOT_REGISTERED] The funny thing is. The webapp *is* registered. Makes and receives phone calls just fine and even sofia shows the extension is registered. I have reactive components that actively show the registration state in the web app too. I have no idea why Freeswitch is throwing this error. Anyone have any leads?

    1
    1
    "Initials" by "Florian Körner", licensed under "CC0 1.0". / Remix of the original. - Created with dicebear.comInitialsFlorian Körnerhttps://github.com/dicebear/dicebearVO
    VoIP 11mo ago
    Jump
    Securely Exposing FreePBX for Remote Access with a Focus on SIP and RTP Ports
    "Initials" by "Florian Körner", licensed under "CC0 1.0". / Remix of the original. - Created with dicebear.comInitialsFlorian Körnerhttps://github.com/dicebear/dicebearVO
    VoIP 11mo ago
    Jump
    WebRTC client can only make calls. Not receiving any.
    "Initials" by "Florian Körner", licensed under "CC0 1.0". / Remix of the original. - Created with dicebear.comInitialsFlorian Körnerhttps://github.com/dicebear/dicebearVO
    VoIP TheRealNalaLockspur 11mo ago 100%
    [Dev Only] Freeswitch ESL and SMS (SIP SIMPLE)

    Does anyone have experience with freeswitch esl? I am hooked into freeswitch esl through node. I do this to greatly extend the functionality of freeswitch. What I am trying to do is catch the event when freeswitch processes an SMS. I can see it on fs\_cli (console) but I am not finding ANY event header for it and trying to avoid writing a js script fs side in mod\_sms. I googled for a bit and found that the event header was 'CUSTOM' (Where sofia events fire) and the subclass was 'sms::receive-message'. But this subclass does not exist. I even looped through all object properties of the event::custom object. I searched for sms and SMS but turned up nothing. Does anyone have any experience with this?

    1
    1
    "Initials" by "Florian Körner", licensed under "CC0 1.0". / Remix of the original. - Created with dicebear.comInitialsFlorian Körnerhttps://github.com/dicebear/dicebearVO
    VoIP 11mo ago
    Jump
    Accessing audio media streams in SIP call in nodejs using sip.js
  • "Initials" by "Florian Körner", licensed under "CC0 1.0". / Remix of the original. - Created with dicebear.comInitialsFlorian Körnerhttps://github.com/dicebear/dicebearTH
    TheRealNalaLockspur
    11mo ago 0%

    I don't know where to begin with this... holy cow man.

    When you googled everything.. how did you magically land on Sip.js needs to be on Node.js (Backend)?

    0
  • "Initials" by "Florian Körner", licensed under "CC0 1.0". / Remix of the original. - Created with dicebear.comInitialsFlorian Körnerhttps://github.com/dicebear/dicebearVO
    VoIP 12mo ago
    Jump
    kubernetes and freepbx
  • "Initials" by "Florian Körner", licensed under "CC0 1.0". / Remix of the original. - Created with dicebear.comInitialsFlorian Körnerhttps://github.com/dicebear/dicebearTH
    TheRealNalaLockspur
    12mo ago 100%

    I setup K8's successfully using freeswitch and KAZOO/Kamailio as an SBC. Even have our webapp in Azure kubes.

    NAT was a bitch... but just keep pushing through!

    I highly doubt there is anything on the market that is turn key though... and this took a couple of months to program too.

    1
  • "Initials" by "Florian Körner", licensed under "CC0 1.0". / Remix of the original. - Created with dicebear.comInitialsFlorian Körnerhttps://github.com/dicebear/dicebearVO
    VoIP 12mo ago
    Jump
    Advice for setting up a SIP server
  • "Initials" by "Florian Körner", licensed under "CC0 1.0". / Remix of the original. - Created with dicebear.comInitialsFlorian Körnerhttps://github.com/dicebear/dicebearTH
    TheRealNalaLockspur
    12mo ago 100%

    Spin up a freepbx instance or fusionpbx instance. You'll have to hook it up to a trunk provider to place calls through the PSTN though. Can't say any trunking providers names... cause we get banned now. But you have google :)

    1