similar to: How to send a call to a Polycom SIP phone with NO callerid whatsoever

I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is

We are running 2.1.1 with 3 expansion modules. Presence in turned on in sip.cfg, hints are set up on the extensions and buddy watch is enabled on about 40 extensions mapped to the expansion modules. We have 3 paging groups: "Building A" with 20 phones, "Building B" with 20 phones and "Both with all 40 phones. When Building A or B is paged separately, the 601 handles it

Hi, I know having Asterisk aware of Polycom "Do No Disturb" state wasn't working before (1.4), but is this working in any recent version? Is there any "custom" way of doing this? Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Contact me off list, I have a sample extensions.conf file that I can share. It has Paging (one to one and One to Many) Ivr includes, time of da routing and it is geared towards Polycoms. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Thomas Johnson > Sent: Friday, January 20, 2006

Hi, I know that if you have Polycom phones properly configured, you can use "sip notify polycom-check-cfg SIP_REGISTRATION_ID" to have the phones download the new configuration from the provisioning server and reboot. Is there anyway to send the same command to all peers (let's say I had 50 polycom phones that I wanted to reboot)? Thanks, Mike -------------- next part

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 everyone, I have been dealing with a certain issue with a particular customer site for months now. The problem occurs when there is an error with caller id as shown in the following: WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' When this happens, it appears that the call still goes through as I can see the caller still navigating

I'm having a problem with the callerID that the polycom IP600 phones are displaying. I would like to modify the CIDName and leave CIDNumber as exactly what the phone call came in as(provided they aren't hiding callerID). Most of the calls will be going to the queue, but a few will go directly to the SIP phones. I've done a various combinations of using SetCallerID(),

Hi all, Has anyone made up custom ring tones for the Polycom SIP phones? We use different rings for different lines, but the ones it comes with are all very similar. In the interesting of sharing, here's one I made up for paging: <PAGE_BEEP se.pat.ringer.13.name="Page Beep" se.pat.ringer.13.inst.1.type="chord" se.pat.ringer.13.inst.1.value="12"

Hello everyone, Please forgive the exclamation points but I have been battling this one off and on for about four days now. Sorry for the cross post. It all started with a box of IP 501s. I contacted my reseller and obtained the latest BootRom and SIP firmware. Unzipped, configured, copied over to my FTP server (running AstLinux, of course). The phone booted, so far so good. Updated

Greetings- As many of your are Polycom "experienced", I was hoping some kind soul could provide direction on a specific issue. On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an instance where, using intercom/paging functionality of FreePBX, I need to override an end user's 'Do Not Disturb' selection on the handset. By default, DND simply

Somebody was helpful enough to give me the very latest release of Polycom's firmware (2.1.0). Unfortunately, I still get that issue. So I'm stuck asking again: Anybody ever got that? Mike _____ From: Mike [mailto:list@virtutel.ca] Sent: Wednesday, April 11, 2007 13:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Polycom 501 issue with latest

Hi, This used to work fine in 1.4: exten => 2131/,1,NoOp(reject3: ${CALLERID(num)}) exten => 2131/,n,Playback(no_unknow_callerid_here) exten => 2131/,n,Hangup And now, after upgrading to 1.6.1.x it matches every callerid. Did something change? Thanks,

Hi, Has anyone used polycom phone IP500/600's conference feature? I ran into some issues. Polycom is not behind NAT. Polycom calls pstn number A, (gets two way voice), presses conference, (A is put on hold), calls pstn number B, (gets two way voice), presses conference again, suppose to have all three parties in the call, polycom and B have two way voice, but polycom doesn't have two

I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 "Internal Server

Hi I am struggling with hardware choices to get started with. My options are narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G. of importance is: - functionality / integration with asterisk - headset functionality and use - voice quality - build quality Is there much of a difference between Polycom and Cisco? Scanning the group it looks like there may be slightly more

Hi, Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound?

Docs? Polycom has docs? Where would one find this fabled land of... err I mean Polycom does stating what ftp servers are supported? Doug. -----Original Message----- From: Ken D'Ambrosio [mailto:ken@jots.org] Sent: Tuesday, March 07, 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hi, Where should I go to get the Polycom`s latest official (non-beta) version? I am registered on the Polycom customer website but that doesn't seem accessible. Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061102/284b67a7/attachment.htm

Hello all, We have an Asterisk server that has worked without issue for a while. Before, only Sipura and Polycom 500 series phones were used. Recently, we've added a few POE switches and 20 or so Polycom 330's. The 330's seem to lock up often. One easy way to do this is by hitting the * key. I looked thru the XML files for dialplan and what not, nothing stood out.