WebCall and Digital Recepcionist Function not Working 03.10.2008 23:01

Forum: Bugs

Hi PBXes developers,

I am just writing because the function "webcall" is not working only in my PBXes account.

I made the same configuration for me and for my costumer, but in my PBXes account the WebCall function doesn't work (in my costumer PBXes account it works fine with the same configuration).


My user name is: mesa ---- > webcall not working

My costumer user name is: gcosin ---- > webcall working fine


I registered the extension "mesa-500" in my Linksys ATA and I can't receive calls from "webcall" function (other calls I can receive normally in this extension).


Can I ask you to fix it for me!


Other problem in my account is with digital recepcionist configuration.

When I dial *77 and say something in the microfone, PBxes doesn't record anything. I confirmed it because I dialed *99 and nothing was recorded.

Can you please fix this two issues of my PBXes account?

Please, take a look in my account!


Can it be solved if I delete my PBXes account and creat a new one?

thanks

RE: IAX Protocol support for trunks 03.10.2008 22:47

Forum: Feature Requests

Actually, there are many Voip's with IAX support, but with both support (IAX and SIP).

In my PBXes account I have Voip with support for both protocolls (SIP and IAX) but I have to use SIP because PBXes don't have an alternative to configure IAX trunk.

I think that is not dificult for PBXes developers to put an alternative to add IAX trunk's.

IAX Trunk 01.09.2008 09:46

Forum: Feature Requests

I would like to ask PBXes to give a way to users add IAX Trunks.

We know that IAX has more quality voice than SIP protocol.

The IAX protocol is used by the Asterisk as an alternative to SIP and H.323.

IAX were updated to IAX² that uses a single UDP port 4569, and thus works well in NAT environments.

It means that IAX² uses ONLY one UDP port for both control and data traffic, and it provides more quality voice.

IAX² supports PKI-style authentication and trunking, and IAX trunking allows multiple voice streams to share a single "trunk" to another server, reducing overhead created by IP packets.

IAX always sends DTMF outline (RFC2833), eliminating the confusion often found with SIP.

One more advantage of IAX² over SIP is the IAX² jitterbuffer. This tends to improve the audio quality noticeably over non-ideal links like ADSL.

Nowadays, more and more Voip's providers are using the IAX² protocol, as alternative to SIP, and that's why I give this idea to PBXes to allow us to configure IAX² trunks.