Re:
silentfun:
Thanks for your reply.
My example with the "Failed to find a transport to host 0.0.0.0" is a response from your previous post where you recomend me try sip.sipnumber.net.
I know sip.sipnumber.net not exist, that the way i show you. The correct address is sipnumber.net. But last one was problem with version 5.
I know we have another forum dedicated only to voip providers, but in this case i reporting also other problems with extentions wich remain busy after disconnect.
But, you have right, i can post only the voip provider problem in that forum.
Thanks
Nick:
Thanks you too for your response.
I know, your software is really good, and i mencioned that, it's free. Really how i told you, i'm a big fan of 3cx. I think it's really great, an exceptional as you told me. I not doing a negative feedback to 3cx, otherwise, some responses give to me a negative feedback and i only ask what happened. Please, don't think as a negative feedback.
In my first post of the problem, i only want announce to you the problem, and maybe i can get an solution from your team or other user. i don't be waiting and enterprise level support, only a little help.
I hope we can continuing collaboration with you and your team improve your software, is great know your remain your policies unchaged. And now is good also know your team is focus right now in other type improvement, and after january we can get more luck.
My principal reason to try version 5 was the caller id name feature. But, now i'm back to version 3, and using direct sip domain calls from my pstn gateway to an extension, i can get the incoming caller id without problem. So, i can continuing testing the version 5 in another machine and keep the version 3 running in home.
Thanks,
JOSE