similar to: Polycom 501 + Asterisk +Edit buttons

Hi, all. I want to have a button on my receptionist's 601 that, when pressed, will forward her current call to a given extension. Is there any way to do that? I've tried to RTFM, but I'm coming up empty. Thanks, -Ken D'Ambrosio

Hi - Has anybody been able to successfully remap SpeedDials on Polycom phones? The manual seems to indicate that you can, and I followed the advice in this list message: http://lists.digium.com/pipermail/asterisk-users/2005-October/129142.html The result I get is that the remapped buttons act as the VolumeUp button, rather than as a speed dial. My configs look like this: <Mac

Anyone else have problems with soft buttons not being responsive at all? 2 of the 4 soft buttons do not respond, no matter how hard you push. It is an IP500. Well over 1 year old. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061208/b3fa0b42/attachment.htm

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, Just started using an asterisk-based PBX with Polycom IP501 phones. Am Fairly satisfied and am starting to get into FTP setup of the phones. Have figured out most things except for how button remapping works. In sip.cfg, I have this entry: <keys key.IP_500.31.function.prim="DoNotDisturb"></keys> This works as expected but if I try to change the remapping to any

I am doing some experimenting with Asterisk 1.0.9 and Polycom IP501's. I have the extensions setup, and everything is working well up to this point. Now, I want to setup my system so that a user at an extension can start a recording on demand. I have tried various Google searches, but am coming up empty. Could someone point me in the right direction, or have some sample Polycom/Asterisk

I think Polycom has added another feature that nobody wants. With MWI configured, and a phonexxx.cfg that has this: <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="XXX" msg.mwi.2.subscribe=...> </msg> Under 1.3.4 and earlier, the phone would immediately

Hi - > I am not sure what I did but blind transfers do not work. The Polycom does > not allow me to dial the extension of the person I want to transfer to after > I hit: > > transfer -> blind I would strongly suggest getting the latest firmware, and using the sample configuration files with that firmware to set up your phone. This SHOULD work. If it still does not work

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Noah Miller > Sent: Wednesday, April 25, 2007 9:52 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] My Polycom IP 501 is formatted its file > systemitself > > Hi Chandra - > >

Hi, This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the display screen. How can I set the "New York" time? What value I have to give to GMT offset value? Look forward to your response. Thank you. Regards, Chandra. --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

This is a dialpaln issue. I solved the same problem recently. For 4 digit extensions you need to append the dialplan statement in the sip.cfg configuration file as follows <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/> Michael > I am not sure what I did but blind transfers do not

Howdy - > The transer button on the polycom phone does not seem to transfer/park the > call properly. I have to use the # -> 700 to park the call. If I recall, using the Polycom transfer, you have to make sure it is done as a blind transfer. The Polycom attended transfer (default) option does not work. > Furthermore the # -> 700 only works on incomming calls. If I

Hi All - I noticed that the Polycom IP501's are now shipping. Has anyone gotten one yet, and if so, what's different about the phone? Any UI improvements, or is it just better hardware? Thanks, Noah

Hi, We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. Look forward to your response. Thank you. Regards, Chandra. --------------------------------- Ahhh...imagining

Hi Andrew - > On Monday 13 March 2006 10:20, Noah Miller wrote: >>> The transer button on the polycom phone does not seem to transfer/park >>> the call properly. I have to use the # -> 700 to park the call. >> >> If I recall, using the Polycom transfer, you have to make sure it is done >> as a blind transfer. The Polycom attended transfer

I'm attempting to set up my SoundPoint 501 with my Asterisk server. I've configured DHCP and TFTP and successfully updated both the BootRom and SIP application. I've also created a custom cfg file for this phone's MAC address and the settings seem to be taking just fine. I can see that the phone registers with my Asterisk server but 'sip show peers' reports that the phone

We have two small business that run out of our office. One business has 3 phone lines, and the other has only one. In a perfect world, Asterisk would indicate WHICH line (or group) the outsider caller called, so that we would know which way to answer the phone. The incoming calls would go through a voice mail menu first - different for each company. (Press one for sales, two for

Hi All - I've got a load of Polycom phones, and for the most part, I think they're great, but one thing that is bugging the heck out of me (and my users) is the "on-hold" feature. When you're on a call, and another one comes in, it doesn't ring the second line appearance on the phone, even though I have it registered separately, and I've tried to make my

I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system

Hi Andres - > I have a small setup with 2 SPA3000 1 SPA2001 and 1 Polycom 301 > > The Polycom misses 1 out of 2 dialout calls, this is the full log > from a > call which didn't go through. > > 303094 Sep 14 10:45:15 DEBUG[15073]: Stopping retransmission on > 'f4e376c3-7531ff39-c86f6812@192.168.1.18' of Response 2: Found > 303095 Sep 14 10:45:15