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  1. Adrian Fretwell

    Freeswitch debian repository asking for username/password

    It is asking for a password, no idea why.
  2. Adrian Fretwell

    What have you changed?

    We have not added very much to the GUI. We had to code our own call centre wallboard because the commercials did not work for us in paying the whatever colour membersip that was required at the time. We built a few desktop apps that did BLFs, chat, call stats and clipboard dial etc. and as a...
  3. Adrian Fretwell

    Copyright removal?

    grep is your friend.
  4. Adrian Fretwell

    Debian Buster Backports libcurl3-gnutls breaks ability to upgrade via the web gui and via git directly.

    Indeed. OS version is largely governed by the recommendations for FreeSWITCH.
  5. Adrian Fretwell

    Recorded Greeting & Remove Dynamic Greeting

    If you are bridging to voicemail, you may need to use "export" rather than "set". All the details for FreeSWITCH voicemail are given here: https://freeswitch.org/confluence/display/FREESWITCH/mod_voicemail You can see how the FusionPBX voicemal lua works by looking at...
  6. Adrian Fretwell

    Extract Call ID header in SIP invite

    Just came across this whilst researching something else, just adding for completeness... From the FreeSWITCH Documentation: https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files Call ID inbound-use-callid-as-uuid On inbound calls make the uuid of the session equal...
  7. Adrian Fretwell

    How to empty the accumulated errors in the folder?

    I'm not sure that I understand your question. If you are referring to the FreeSWITCH log file then you can start a new one with: fs_cli -x "fsctl send_sighup" And then delete the old one if you wish.
  8. Adrian Fretwell

    Can't send email from outbound dialplan route

    Sorry, should have said - I'm rushing a little this evening. 4.5.11 on Debian 10.
  9. Adrian Fretwell

    Can't send email from outbound dialplan route

    Hmm... Just tried your setup and it worked OK, screenshot below:
  10. Adrian Fretwell

    Can't send email from outbound dialplan route

    Just looking at the example at the bottom of the send_mail.lua file, it looks like the first argument needs to be headers not to@email.com... --local headers = { -- ["X-FusionPBX-Domain-UUID"] = '2d171c4c-b237-49ca-9d76-9cffc1618fa7'; -- ["X-FusionPBX-Domain-Name"] = 'domain.com'; --...
  11. Adrian Fretwell

    SOLVED Only local calls are being recorded

    In the case of outbound calls the dialplan user_record relies on the condition from_user_exists. How are you requesting recording? In the extension record?
  12. Adrian Fretwell

    API Guide

    API is a member feature that you pay for unless you write your own.
  13. Adrian Fretwell

    Whats your Biggest deployment?

    If you are serious about being a hosting business, then get your own servers in a reputable Data Centre. VMs are OK as long as you control the hypervisor. From the questions you are asking, it sounds like you have a steep learning curve in front of you. I will try to answer your questions...
  14. Adrian Fretwell

    Convert yesterdays wavs to mp3

    Is your script setting the shell variable $BNAME correctly? For what it's worth, I do this slightly differently without referencing a variable in the SQL statement. This method makes the assumption that the file conversion completed successfully as there is no connection between the conversion...
  15. Adrian Fretwell

    Extract Call ID header in SIP invite

    I'm sure it can be captured in the DB one way or another, I seem to remember reading something about [enabling] B-leg channel variables and XML_CDR somewhere on the FreeSWITCH website, but I can't remember off the top of my head. I'll post back if I get time to look, likewise let me know if you...
  16. Adrian Fretwell

    Extract Call ID header in SIP invite

    I believe what is shown in the CDR is the sip_call_id for the A-leg of the call. As FusionPBX (FreeSWITCH) is a back to back user agent (B2BUA) there will be a different sip_call_id on the B-leg. I don't believe it is possible to get the B-leg sip_call_id within the dial plan.
  17. Adrian Fretwell

    IVR prompt choice - not dialing in

    This is probably an issue with DTMF tone detection. It may help if I provide a little background information. There are three types of DTMF transmission within SIP: RFC 2833, Info, and Inband. RFC 2833 has become the method most commonly used although technically speaking RFC2833 has been...
  18. Adrian Fretwell

    Move to new server

    The function tells you if magic quotes is switched on in the php configuration. Having magic quotes was always a bad policy and I believe it has now been removed for security reasons. You are probably safe to just delete the whole IF statement in php.php.
  19. Adrian Fretwell

    Extract Call ID header in SIP invite

    The channel variable is called sip_call_id. You could also get it with sip_h_call-id