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Thank you very much for your help @hfoster. I asked the freeswitch community in Slack. If I have an answer I will post it here.

@dcitelecom, already checked and everything is handled by FusionPBX/FreeSWITCH

@dcitelecom, Indeed transfer features in Zoiper are only available in the PRO edition. However, I don't believe that Zoiper is acting as a PBX. I did a test where both B and C are mobile phones and everything worked fine. Also, while B and C were connected I closed Zoiper and the connection...

@dcitelecom thank you for your feedback. I though that the flow I described above is a standard way for doing attended transfers. I am using Zoiper and this is something they recommend: https://www.zoiper.com/en/support/home/article/199/Call%20Transfer%20with%20Zoiper%205 I also tried to use...

@hfoster is this possible to make Att Transfer to Cust 2 (step 2 as you described above) with the current dialplans ? Are you referring to feature code *4?

@hfoster did you have any chance to run sngrep? Thank you in advance!

Additionally to what @hfoster suggested above I would recommend to consider the following: 1. If you are going to have transcoding you should also incude this in your tests. Transcoding will increase the CPU load. 2. In a real environment, if your PBX is opened (your SIP ports are accessible by...

Another interesting thing is that I tried to use the uuid_display command to update display information of a local call (using Zoiper and 3CX as clients) and nothing changed. I didn't notice any re-invite in sngrep. For this test I set ignore_display_updates=false in the diaplan manager (diaplan...

Thank you @hfoster ! The plan is to go out via a sip trunk. However, if I do my tests using local extensions (with different outbound caller id numbers) I don't see the change either. Does this also depend on the sip client ?

Hello, I have the following scenario: 1. Agent (1001, CLI=123456) makes an outbound call to Customer 1 (destination number=777777). Customer 1 sees 123456 as the incoming number. 2. The same agent makes another outbound call to Customer 2 (destination number=888888). Customer 2 sees 123456 as...

Thank you very much @hfoster !

@hfoster thank you very much for your reply. If I want to implement this feature on my PBX shall I use the limit command in a dialplan? I also tried to set the 'Limit Max' variable in Extensions to 1 but it does not work.

Hello, I would like to ask if it is possible when an extension is busy (already talking to the phone) to transfer its incoming calls to its voicemail. Currently, I observed that when an incoming call is received it is directed to the voicemail only when I reject the call (hit the hang-up...

If you want to go with a commercial solution you can also try Kolmisoft MOR. I am using it for the last couple of years and everything is working well.

Following the comment of @alaamrim I did the following: I created a SIP extension (1000) at domain x.com I edited the "domains" ACL entry and added the domain x.com with type=allow I edited the new extension and I set its CIDR field to 3CX's IP address (e.g., 2.2.2.2/32) I executed the...

Thank you very much @alaamrim ! I will try to do the same and test on my system too.

Hello, I would like to connect 3CX with FreeSWITCH (FS) in order to route calls to/from the following directions: 1. 3CX -> FS -> SIP TRUNK 2. SIP TRUNK -> FS -> 3CX The reason I want FS in the middle it is because I have some special features for processing CDRs and recordings which are not...

Yes that's correct!

You could use heplify agent (https://github.com/sipcapture/heplify.git) to send packets from FusionPBX to Homer without any configuration on FusionPBX. I am using it for a long time now without any problems.

I am not sure about this. In the most of the cases, if I am not wrong, the \d is used (e.g., ^\+(\d{6,17})$). Of course I can modify the expression and accept whitespaces at the beginning or end of the string but for this I have to modify hundreds of outbound routes :)