[asterisk-users] Paging with Polycom 3.3.x

[Mike]

Polycom are at 3.3.1 now, so 3.3.0 should be fair game.

 

It has nothing to do with paying or not, the company that sold you the phone
should be able to give you the latest version no?  Unless you bought from a
guy who found a box that fell off a truck.or some third-rate reseller.

 

Mike

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of William
Stillwell
Sent: Thursday, February 24, 2011 3:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x

 

Is 3.3.x downloadable for non-paying people yet?

 

 

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mike
Sent: Thursday, February 24, 2011 2:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Paging with Polycom 3.3.x

 

Thank you Terry and Ryan, I will try those things and see if I can find my
problem. Will definitely come back with my solution in case it helps
somebody else.

 

Mike

 

 

Hi Terry,

 

I did that, and did find a autoRingAnswer and Ringanswer modes that I tried
to use, but somewhere I am missing something that breaks it. If ever you
find what you did, I`d appreciate if you'd share with me.

 

Mike

 

 

 <alertInfo voIpProt.SIP.alertInfo.1.value="Ring Answer"
voIpProt.SIP.alertInfo.1.class="4"/>

and

<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer"
se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6"
se.rt.4.mod="1"/>

where the timeout is the ampount of time on milliseconds before it goes to
speaker.

 

These values are in the sip.cfg, so in your server it may be sip_316.cfg.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110224/b5a4e493/attachment.htm>