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I have created one IVR (IVR350 ) and assigned it to ext 350. Created an inbound rule from the Dial plan. See attached settings. I can call ext 350 internally and I am using an inbound route to forward calls from a DID to the IVR. I can select 1 or 2 and the calls and they transfer...

Looks a complicated method for what I need. In the app I filter on the extension ( eg ext 1000 ) for the connected tenant. I then pull the call history from the database. I then want to drill down using 'call_recording_uuid' I combine the path and filenames using call_recordingPath (...

Got that working now. I had to restart freeswitch. Thanks for the help. Worldwide domination here we come.........

And in the profile: rtp-ip: autonat: $${local_ip_v4} sip-ip: $${local_ip_v4} Is the the internal sip profile ? Is the setting for rtp-ip = 'autonat: $${local_ip_v4}'

Hi guys, anyone know how to do this. I presume I would need to password protect the folders so only my app can load the wav files. I've found various methods to do this but am unsure what to do and don't want to screw my folder permissions up......

I have no changed the port setting in rules.v4. Still no audio ngrep shows from one app one way # U 192.168.1.2:20021 -> 2.99.151.76:4013 #19 ..........................................9ac88cd5.%FreeSWITCH.org -- Come to ClueCon.com... # U 192.168.1.2:20021 -> 2.99.151.76:4013 #20...

Thanks for the info. I have changed the /etc/freeswitch/autoload_configs/switch.conf.xml file and saved. restarted freeswitch Added the port forwarding. I'm still not getting any audio. I have the following port forwarding. 5432 (PostgreSql) Sip 10060 TCP&UDP Sip 5080 TCP&UDP 20000 - 20500...

Hi, I have my fusionPBX running on our office server. We have changed the sip port to 10060 ( for security ) and our TalkTalk router does not allow port forwarding of 5060. So we port forward on the office router 10060 and can make calls between extensions outside of the domain but no sound...

Update..... I tried a couple more installs and then I realised I was installing the 32bit version not the 64bit version. Just done a fresh install and can connect to the sip trunk. Cheers to Adrian Fretwell for all the help ......

For anyone else looking at the database side of things here is the Table Structure. Hope it helps. The structure looks fairly straight forward. If you need any help PM me.

I need to be able to play the messages within my app. I have got the paths from the database table v_call_recordings eg var/lib/freeswitch/recordings/my.domain.com/archive/ so if use my.domain.com/var/lib/freeswitch/recordings/my.domain.com/archive/2020/may/mywav.wav So how do I expose...

I am using the following ( from various searches ) originate sofia/internal/1000@test.toolfolks.com &bridge(sofia/internal/1053@test.toolfolks.com) This should initiate a call from ext 1000 to ext 1053 but nothing happens. What am I missing here guys ?

Posted in wrong category so re-posted here. I need a notification when a an inbound or outbound call is made along with the caller ID. I then need to pass this into our mobile app ( I'm looking at firebase at the moment but open to suggestions) I'm looking through the database and see the...

Update.... I'm looking through the database and see the v_xml_cdr table stores the call logs. I can create a trigger on the insert record but unfortunately this is not inserted until the call is ended. Is there any temp tables created anywhere when a call is made or received ? So I can create...

Hi guys, plodding along with all this Linux typing stuff... ( difficult being Dyslexic as well as old.... ). I have puttyed ( correct wording ? ) into the server as root. As an attempt to understand this I will attempt to connect from 2 mobiles one with a correct password and one incorrect to...

What are the ‘ optimum ‘ settings. It’s a minefield as there are 6 settings on the 6128 so how does one test the various combinations. Any documentation out there.

Cheers. I’ve changed them. test in the morning

Cheers, as usual as a newbie I go to check the settings and there is a few different options. Which one is the registration timers

Been looking at lua ( doesn't look too tricky). Where do I add these magical scripts within the interface ? Any tutorials ?

I have a Grandstream GXP1628 and HT108 ATA. I have our mobile software running on IOS & Android and also GSWave on IOS & Android. All can call each other put on hold etc etc. I have a DialPlan that checks for the caller ID and if it matches transfers the call direct to ext 1053. However after...