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[asterisk-users] Shared line appearance phones?


 
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bmcmanus at gmail.com
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PostPosted: Thu May 15, 2008 12:13 pm    Post subject: [asterisk-users] Shared line appearance phones? Reply with quote

I hate to bring up an old thread, however, I'm implementing SLA as well.

I've got SLA working, my tunk executes slatrunk(line1), and my polycom 650
phone rings on the SIP subscribed line (button 1)

I'm assuming slatrunk sends the calls to the SIP/station1 SIP device, so
the call will always appear on "button 1" on the end device?

My question then is how does the receptionist or answering party know which
line this came in on so they can, for example, overhead page and say "Fred,
you have a call on line 1?"

My next observation is on the execution of slatrunk() the call quits
ringing, so the calling party gets dead air. Is that normal?

I believe the call coming in on button 1 is hear to stay, part of the
'trickery' but if their is an easy way to know which line that call is from
that would help, and secondly if slatrunk played a ringing tone till it saw
a station join the bridge that could help.

And perhaps I have everything misconfigured as well Smile

Thank you for all the asterisky things you do!

Brian

On Fri, Nov 30, 2007 at 6:10 PM, Russell Bryant <russell at digium.com> wrote:

Quote:
Mark Wiater wrote:
Quote:
I fought with this in 1.4.5 with polycom phones. I was hoping to share a
DID from a PRI on several
Quote:
Polycom IP430's.

Might you be willing to share some specific configurations for such a
situation?

There are some basic examples in doc/sla.pdf in the 1.4 tree. However, I
have
on my to-do list to spend a week with an SLA test environment and coming up
with
an extensive set of examples of the different ways it can be used.

I will post something to this list when that is available.

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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joakimsen at gmail.com
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PostPosted: Thu May 15, 2008 11:36 pm    Post subject: [asterisk-users] Shared line appearance phones? Reply with quote

The docs as far as I can tell are not correct. E.g. Zaptel is required
(because it seems that it uses MeetMe) but none of that is documented.

So yes please do see if you can make the feature work and please post
a working example config for a Polycom phone.

On Fri, Nov 30, 2007 at 8:10 PM, Russell Bryant <russell at digium.com> wrote:
Quote:
Mark Wiater wrote:
Quote:
I fought with this in 1.4.5 with polycom phones. I was hoping to share a DID from a PRI on several
Polycom IP430's.

Might you be willing to share some specific configurations for such a situation?

There are some basic examples in doc/sla.pdf in the 1.4 tree. However, I have
on my to-do list to spend a week with an SLA test environment and coming up with
an extensive set of examples of the different ways it can be used.

I will post something to this list when that is available.

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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