Voxilla VoIP Forum - TalkSwitch Support Forum

Is it possible to prevent transfer to extension during Auto Attendant?

Johnny Cakes — Tue, 07 Jul 2009 01:25:43 GMT

For example, if the caller knows that the president is extension 111, the caller can just avoid the "correct" options in the auto attendant and ring the president directly from the auto attendant, even when that is not an auto attendant option. This can't be avoided?

Configuration File

boots963 — Wed, 24 Jun 2009 18:53:46 GMT

Hello,

I am kinda new to this system a client i am doing work for bought it and wants me to make it work. Well as you can see i am here for some help so here it goes. I have to talkswitch boxes when i connect to them they state that one is a 284vs and the other is a 48ca when the system was first connected they seemed to recognize each other but since then i have defaulted each unit and they do not seem to work together now. I have put together a configuration file with unit 1 being the 284 and unit 2 being the 48ca but when i try to load the file it says the configuration file cannot be imported from selected file.The Talkswitch system you are connected to does not support the configuration in this file. i am connected to the 284vs via usb i asm kinda at a loss i tried the instructions given by talkswitch to no avail?

ANY HELP WOULD BE GREATLY APPRECIATED THANK YOU

Can MagicJack be an analog line for TalkSwitch?

acentiallc — Fri, 19 Jun 2009 03:21:46 GMT

I use a MagicJack as an analog line on a Talkswitch VS288 system. It works flawlessly so far.

Incoming SIP VOIP Call not clearing properly.

mdm2812 — Mon, 15 Jun 2009 14:32:55 GMT

I have a Talkswitch problem that I thought the group might help me with.

Talkswitch 488VS. Firmware 6.11.047 (current).

VOIP supplier: voip.ms.

Talkswitch is registering fine with voip.ms.

Voip.ms sends me DID calls through SIP at: SIP/12341234567@X.XXX.com (X.XXX.com is my talkswithc.)

Call gets received fine, 2-way voice etc.

At end of call, when parties hang up after about 1 second, the Talkswitch line rings again. Obviously, when the line is answered, there is no one there.

Very bothersome of course!

Any ideas where I should be looking to solve this problem?

I suspect that by having voip.ms send me the calls by SIP, the challenge is that somewhere the SIP call isn;t then sending the Talkswitch a diconnect signal? SInce I am a SIP newbie, I don;t really know what to look for in logs etc.

Any help is appreciated!

Talkswitch and non 'approved' Service Providers

mdm2812 — Thu, 11 Jun 2009 13:45:18 GMT

Is there an easy way to determine which system a VOIP provider uses to pass DID information?

i.e. if it is in the Request URI field or not?