PBXes (http://www1.pbxes.com/forum/index.php)
- English (http://www1.pbxes.com/forum/board.php?boardid=16)
-- Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups (http://www1.pbxes.com/forum/board.php?boardid=20)
--- RE: Intercom settings (http://www1.pbxes.com/forum/threadid.php?threadid=1412376379)

Intercom settings

Hi there,

I have 5 Aastra 67571 SIP phones and I want to enable an intercom feature between 2 extensions. Everything is configured on the phone side, but I cannot get it to work. Is there a special prefix to use for PBXes for intercom calls (I've tried ** but that didn't work)?

Peter

RE: Intercom settings

Hi,
are you saying that you cannot directly dial any extension numbers from your sip extensions, but other calls work ok ?
What appears in your call monitor ?
Sounds like your sip device isn't configured properly

RE: Intercom settings

I can dial any extension and the phone rings and can be answered, but one of the features of an intercom is to have the call be anto answered (e.g. by a secretary or v.v.).

Everything that has to be setup for intercom is configured on the phone side: () only for explanation

sip intercom type: 2 (1=phone side | 2=server side) **
sip intercom prefix code: (??)
sip intercom line: 1
sip allow auto answer: 1 (1=true | 0=false)
sip intercom mute mic: 0 (1=true | 0=false)
sip intercom warning tone: 1 (1=true | 0=false)
sip intercom allow barge in: 1 (1=true | 0=false)
sip early media mute mic: 0 (1=true | 0=false)

** according to the admin manual of the phone: "This parameter is required for all server-side Intercom calls."

Hope you are able to help me with this.

Thx.

Peter

RE: Intercom settings

Hi,
Sorry, I have no experience of the intercom function.
There are however a couple of references in pbxes forums which may be relevant to you although they are not recent:

http://www1.pbxes.com/forum/thread.php?postid=1209719128

http://www1.pbxes.com/forum/thread.php?threadid=1694&sid=

Regards

RE: Intercom settings

Hmm, I've read these threads but it isn't clear to me what exactly should be changed. The author refers to an Asterisk server, which is (if I'm correct) a reference to PBXes. But these configurations cannot be changed by me (SoHo).

Maybe a moderator (i-p?) can be of assistance on this matter?

Peter