similar to: Re: Polycom IP501 with Asterisk - distinctive

CVS Head from 07/07/2005 I'm trying to make an IP-501 auto answer a call. exten => 301,1,SetVar(_ALERT_INFO="Ring_Ans") exten => 301,2,SetVar(ALERT_INFO="Ring_Ans") # Tried both combinations exten => 301,3,Dial(SIP/5001,15) exten => 301,4,Hangup Sip.cfg for Polycom phone <alertInfo voIpProt.SIP.alertInfo.2.value="Ring_Ans"

I am looking for clues on how to configure distinctive ring for a PolyCom SoundPoint 300. Does ALERT_INFO apply? If so, how? Thanks, David Koski david.nospham@kosmosisland.com

Hi, I'm having trouble setting the ringtone on my Polycom 501. The relevant entry in extensions.conf is: exten => 801,hint,SIP/creative1 exten => 801,1,SetVar(ALERT_INFO="Test") exten => 801,2,Dial(SIP/creative1,20,Ttr) In the sip.cfg: <alertInfo voIpProt.SIP.alertInfo.1.value="Test" voIpProt.SIP.alertInfo.1.class="13"/> and <TEST

I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call "22" and the phone rang it did not auto answer. Did I miss something? exten => 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten => 22,n,SipAddHeader(Alert-Info: Ring Answer) exten => 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten =>

In ipmid.cfg I have: <G3INTERCOM se.rt.10.name="G3INTERCOM" se.rt.4.type="ring-answer" se.rt.4.timeout="1000" se.rt.10.ringer="7"/> In sip.cfg I have: <alertInfo voIpProt.SIP.alertInfo.1.value="G3INTERCOM" voIpProt.SIP.alertInfo.1.class="10"/> I set up a test extension: exten =>

Hi, I've been meaning to fix my non-working paging feature here for a while, and I've just spent the last 5 hours looking at many, many web pages that all say the same thing. I am using Asterisk 1.6.2.18 and Polycom phones, both older (501 with "latest" legacy 3.1.7 firmware) and newer (335 and 650 with latest 3.3.1f). I have changed the correct values in sip.cfg like

Hi Guys This has been discussed a little in the list before so my apologies for sendig it again but I have done what others have done in the list but to no avail. I have configured Asterisk to send the callerID of extension phones as "firstname lastname" and that seems to work well and extensions show calls originating on other extensions in this format. I set the following in

I'm stumped on this one and any help would be greatly appreciated. I'm just trying to get my Polycom 601 to have multiple extensions on it. For example, on line 1 I want extension 21, on line 2 I want extension 22, and on line 3 I want extension 23. Ideally I'd actually have each extension appear on 2 lines and therefore filling up all 6. I should be able to do that with the

hi there I'm setting up asterisk@home and I'm using Polycom IP 500 phones. When I call a number that has a digital receptionist (i.e. "dial 1 or such and such, dial 2 for this and that...") the Polycom doesn't seem to send the extra digits. When I try it with X-Lite things appear to work fine, so I think the problem is with the Polycom configuration. Here's some

(new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone?

Hi All - I'm trying to implement the auto-answer config from the wiki, but the result for me is that the phone just rings as normal. I'm running firmware version 1.4.1 on an IP500. I've added the following to my sip.cfg: <alertInfo voIpProt.SIP.alertInfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4"/> and this to my ipmid.cfg

Hi, I'm using asterisk 1.2.1 on a debian sarge distro. I've followed notes in http://www.voip-info.org/wiki/view/Polycom+auto-answer+config and http://www.voip-info.org/wiki/index.php?page=OptiPoint+600+SIP+-+Distictive+ring+using+ALERT_INFO but I still cannot change ring style via asterisk using exten => 666,1,SipAddHeader(ALERT_INFO="ring3") in extensions.conf . Is it

Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time

I am using SugarCRM together with the asterisk plugin, which allows me to click a number, SugarCRM calls my extension then places the call when I pickup. I would like to have that extension auto-answer. I set it up as line 3 on my phone so normal calls do not get auto-answered. However, I have not been able to get this to work. Has anyone implented this? This is what I put in the config file

I was hoping that someone may be able to shed some light on some issues I'm having on trying to get an Asterisk test server up and running. At the moment I have the basics, two Polycom hard phones (301 & 601 with expansion unit (which oddly will not power up)) that can call each other, log into voicemail (one touch) and have custom directories & buddy lists. Unfortunately some of the

Hi all - This isn't really directly Asterisk related, but has anyone successfully set up a Polycom phone to register two lines on two different Asterisk boxes? I can get the first line to register, but the second one does not. I can still place calls from that second line, which indicates to me the server, user, and secret are correct. I'm running the newest 2.6 series firmware with the

Hi, My phones stopped auto-answering when being paged, since I moved on to Polycom firmware 3.3.0 (3.3.1 is the same, I tried). That is with Asterisk 1.6.2.16. I looked at the wiki but nothing I try there works, even if I cut and paste the same setup. Any one has any idea of what I should change from my old 3.2.3 setup? My older phone (501) still using 3.1.6 still auto-answer

This is my first foray into SIP telephony, so be gentle. :-) The Polycom SoundPoint IP 501 phones have been fantastic so far. I still have a lot to learn when it comes to them, but the manual seems pretty extensive and so far Asterisk has been playing well with them. I have a need to be able to identify incoming calls based on some factor (could be time of day, caller ID, dialed number, it

The answer is yes, I think, but I don't recall precisely how off the top of my head, and I'm walking out the door in a moment. The phone will hold more than a dozen distinct ring tones which you can create for yourself, and you can have asterisk direct it to use a ring tone independently of line appearance. The most direct way would be with the SIP Alert-Info field, but the phone itself

BOFH told me he uses it to listen to his co-workers > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith > Sent: Thursday, February 09, 2006 12:27 PM > To: asterisk-users@lists.digium.com > Subject: SOLVED: Re: [Asterisk-Users] Polycom IP501 with Asterisk - > distinctive