Recent content by HeisenbergQin

@Adrian Fretwell Thank you so much, through your example, I think I already know how to do it.

Thanks, I have a lot of clarity through your explanation~

It may be that I did not express it clearly enough. I do not consider whether C is an external call. In other words, we do not consider the conference mode now, but use eavesdrop as an example: ABC all communicate with SIP extensions, A calls B, and C can pass through Dial *33 to monitor the...

In other words, does this feature need device support?

OK, Thanks ~

I can now let C participate in the conversation through conference mode or monitoring mode, but the premise requires C to call A or the conference number. Actually, this is not allowed, because we don't know when A and B are talking, so we must You have to invite C to join when A and B are talking.

Or what do I need to do before I can actively invite a user to listen to the call between A and B?

@Adrian Fretwell Thank you for your reply. I have tried to use the conference, but I need C to actively dial the conference number to join. What I want is that A dials C to invite him to join.

@hfoster Thank you for your reply. I have tried the conference mode, but C needs to actively enter the conference. I don't know how to use A to invite C to join the call.

Hello everyone, I want to implement a function: A dials B. After B is connected, A dials C to let them talk to 3 people. How do I implement it?

I am a newbie and now encounter a problem, asking for help: When I use the Freeswitch extension to call a number through the operator's line: Gateway (chinamobile), NO_ANSWER will appear (this is the case in most cases, but occasionally the call will succeed). If I directly use microSip to log...

Hi bro! Using the method you provided, I can indeed get the initial recording! I owe you a bottle of beer~

Okay, thank you. I will get back to you after I tried it.:)

thanks for the reply. Yes, I want to record before the call is connected because I want to get whether the customer number is off or busy...

I checked the log and found this hint: (switch_core_media_bug.c:1298 Removing BUG from sofia/internal/)Does it have anything to do with it?