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Hello - Is there a way to extract the Call ID information in a sip invite. I want to use this information in a custom dial plan but I cannot seem to find a freeswitch variable to extract the CALL- ID. Any thoughts? 2021/11/9 14:27:05.420600 10.0.0.58:5060 -> 10.0.0.59:5060 SIP/2.0 407...

Advanced > module > set Opus to true true Check the status should say running. Then global codec, OPUS,G722,PCMU,PCMA Outbound codec, OPUS,G722,PCMU,PCMA save, flush cache, reload, xml, rescan your profile - Opus should be running

We experienced similar issue as well - same as you over 150 domains all working well and we had this database failure. We just simply restored back to the old back up as we had so many users on my system. Interested to also know what could have caused the issue you reported above.

great idea - let me try this. Exactly what i thought was a way to somehow just find the dial plan and modify it. Thanks

oh thanks for this. much appreciated A custom module is way above my capability. I was thinking of using allowing fusion to trigger my push notification script of i can use my mobile app. Thanks for your response

Hi everyone, Please does anyone know how to find the call center module in fusionpbx. i have tried - /usr/share/freeswitch. Also checked/ /var/lib/ I want to test a script so my script is trigger before the call centre queue attempts to ring an extension. Any ideas where the call centre...

Hi all does anyone please know if we this call whisper feature in fusion? I tried the pre fix either and it did not work

/var/lib/freeswitch/recordings/<DOMAIN>/<WAV FILE>

Thanks for your help Hope somebody else find this useful. Now my call centre can play the announce sound based on the frequency set

Finally sorted it. i was missing the domain name in the path. Full path should have been : " /var/lib/freeswitch/recordings/domainname/audio.wav"

Thanks. it doesnt work when i put " /var/lib/freeswitch/recordings/audio.wav" in the box it doesn't work? I shall check the permission Thanks

Yea - the file is in the system under recordings and i can see and play the file in fusionpbx recordings I used the var path: /var/lib/freeswitch/recordings/audio.wav There must be a way to get the file path and use it in announce sound

Thanks for your help. However, both options don't seem to work. The sound is not announced. I suspect the path is not location the correct file or something.

Thanks - the name of my file is 'audio.wav' What do i put inside ? find / -name 'audio.wav' I am doing somethign silly, it doesn't seem to work?

Hi Everyone - I need some help as I am having the same issue finding the wave path for the annouce sound. Any ideas please?

I want to change the default voicemail greeting globally for all domains. Currently the default intro is something like ......"the person at extension 1000 is not available, record your message after the tone, ..... Is there a simple way to replace this default message with another wave fiel...

@slayerphil We reduced our sip expiry time down from 3600 to 120 and the issue went away. Try that. It appears the expiry number should be between 120 and 180

I am having same issue. We use linphone app and the app create mutiple registration them self. We are seing 100 registration for same user even though I am the only one logged in. Any update on this please? Any way to stop this ? Any way to force freeswitch to loose/removed expired registrations?

t Thank you

root@failover:/usr/share/freeswitch/scripts# wget https://gist.githubusercontent.com/DigiDaz/0bb08b7f4b9f9845b7ad5ea0b1cb8ad1/raw/fb5d21b43ee26399ff3d0bbd9bfcbff170493450/call_forward_all.lua --2020-03-30 21:51:52--...