search for: hielk

Hi All, I was wondering, in my effort to show how Asterisk can replace Call Manager, if there is support for call transfers/forwarding from the users Cisco 7940 SIP phone to either another SIP client or through the AS5300 on to the PSTN. I do see some stuff in the docs but seems to be specific to a local PRI board in the PC of which I don't have. Any experiences/comments most appreciated.

On Tue, 2002-10-29 at 05:01, Hielke Christian Braun wrote: > i am trying to use dovecot with pam and radius. My users have names > in the format joe at somedomain.com. When i have pam configured to use > the normal passwd/shadow files it works fine. With radius it does not. > I see at the radius server that the domain pa...

Hello, hope somebody can help. I have setup a queue which maps to some Budgetone SIP phones. When a call is answered, the # key to transfer a call does not work. Everything else regarding the queue works fine. Is there a way to activate it? Maybe something like the t option in the Dial application. Thanks in advance, Christian.

...t; > Christian. > > > On Wed, Sep 10, 2003 at 09:04:41AM -0500, Brian West wrote: > > On the grandstreams if I recall the docs are incorrect on how the transfer > > feature works. Transfer + EXT + Transfer > > > > bkw > > > > On Tue, 9 Sep 2003, Hielke Christian Braun wrote: > > > > > Hello, > > > > > > hope somebody can help. I have setup a queue which maps to some > > > Budgetone SIP phones. When a call is answered, the # key to transfer > > > a call does not work. Everything else regarding t...

I have a * setup with Grandstream SIP phones dialing out through Nikotel via SIP. When i dial out and the other side picks up, the Grandstream keeps ringing for another seconds and two and the sound coming from the other side is lost. After these two seconds the call is connected find and works flawless. Calling the same path the other way to the Grandstream phone works fine without the cut-off.