Search results

  1. J

    Registration port showing differently

    That’s the source port, perfectly normal
  2. J

    Polycom VVX losing subscription status/indicator on call flows

    What version of FS? are you rebooting FS overnight? I notice with polycom the BLF lights will go out of sync if FS is rebooted, until you reboot the polycom phone also, have you tried shortening the subscribe time as well as the registration time under your polycom phone settings?
  3. J

    SQLite is BUSY often, switch to PostgreSQL?

    Sorry to hijack this thread but I found this interesting. For those who have experience, is it better to run the freeswitch database through Postgres, or is setting up SQLite in RAMdrive better? Specifically for high capacity.
  4. J

    What CRM is being widely used with FusionPBX?

    Sorry i disagree, Vtiger is super buggy these days, and looks like they want to push everyone to the paid version as there haven’t been any community updates in a while
  5. J

    Freeswitch service does not start

    My best guess is that the install script is not properly installing the components for you. Possibly a bunch of dead CentOS repos causing this. But that’s just a guess.
  6. J

    Freeswitch service does not start

    CentOS is dead. Use Debian 11
  7. J

    Safe to delete "Action set provider_prefix" in Outbound Routes?

    I havent had issues after erasing the line. Been few years already
  8. J

    Voicemail Transcription issue

    Try adding transcribe_alternate_language = es (or any other language you’d like). I believe that value needs to be filled in order for google to work
  9. J

    Consultant

    Feel free to PM me if you are still interested
  10. J

    Is there anyone who understands clear freeswitch?

    You can point fusion1 and fusion2’s databases to the Freeswitch server. Or something to that effect.
  11. J

    Looking to move to fusionPBX

    If I were you, I would create everyone under a single domain. Extension number will be the 10-digit phone number of the customer. You will have everyone under one domain, and automatically what will happen is “internal” calling will not leave your system. Hopefully this is a good thing and not a...
  12. J

    anyone get the transnexus scripts to work?

    That has nothing to do with inbound calls. It’s simply the name of your sip profile.
  13. J

    anyone get the transnexus scripts to work?

    where did i mention anything about inbound? if you follow exactly what i wrote you will get it to work
  14. J

    anyone get the transnexus scripts to work?

    I have it working - In each of your outbound routes that you want Transnexus to throw the Identity header, add the following, order it at 30: Action: set / clearipsti=1 Action: set / sip_h_Identity=${sip_i_identity} Action: export / sip_h_Identity=${sip_i_identity} - Bridge the call to the...
  15. J

    apt-update 401 Unauthorized signalwire

    As of 3/22/2022 they are requiring a token to download Freeswitch https://freeswitch.org/confluence/display/FREESWITCH/HOWTO+Create+a+SignalWire+Personal+Access+Token
  16. J

    Use FreeSwitch/FusionPBX as standalone voicemail server for existing IP PBX phone system

    If you dont care to have the MWI status on the phones, I have a very simple solution for you. Set up a SIP trunk between both servers. On your outside server, set up call forward no answer to dummy DID numbers in FusionPBX. In FusionPBX, route the dummy DIDs to their respective voice mail boxes.
  17. J

    For hire: HA cluster setup

    I’d like to hire someone, preferably within the US, to work with me on setting up an HA cluster with FusionPBX/freeswitch. I am also happy to discuss using Kamailio as well If you have the necessary experience and are interested please PM me and we can exchange information and chat more thank you
  18. J

    VitalPBX - More serious evaluation

    Personal advise. RUN from Bicom!! As far as possible. You’ll thank me later. Hopefully won’t be too late by that time.
  19. J

    Google Cloud Speech to Text Voicemail Transcription

    Please make sure there are no duplicates. Also I’m updating the directions now. Mistake on the URL. Use this URL instead: https://speech.googleapis.com/v1p1beta1/speech
  20. J

    VitalPBX - More serious evaluation

    Both started off with chan_sip in 2009 and migrated to PJSIP in 2017 I believe. surprisingly the older the asterisk version the more stable. The more I upgraded the more unstable it became your mileage may vary depending on the load you have. We have 200 tenants and easily 70-80 active calls...