I am a VoIP novice. I thought I was doing good by making a bunch of VoIP phones talk to each other via a software SIP server. I even have analog phones connected via analog telephone adapters. Throw in a few soft phones and I thought I was a VoIP master.
BUT, I am trying something as simple as extensions and I have hit a HUGE WALL.
Here’s my scenario:
Totally standalone test lab; no outside network or POTS connectivity whatsoever.
I need to know if Asterisk can do what miniSipServer cannot.
I am currently running miniSipServer as my SIP server, and have numerous AAstra VoIP phones, Snom 320 VoIP phones, and some analog phones as well.
I am trying to set up extensions on a few different phones so that if a number is dialed (example: 333-1234), all the phones having the same number (333-1234) will ring. If one phone answers, I want the light associated with that number (333-1234) to stay lit on the other phones. Also, I want the other phones to be able to join the conversation by pressing the lit button.
I am able to use call groups to allow all phones to ring (on different phone numbers) when one is dialed but as soon as the call is established, all other lines are released. There is no indication on the other phones of a call in progress and more importantly, another phone can not pick up on the call and join the conversation.
This seems so incredibly simple. Maybe I am thinking too analog. I have been on the forum for my SIP Server and the forum for my Snom 320 and nobody seems to know how to do it.
I’m willing to change to Asterisk and/or different phones if necessary.
Can it be done? How?