similar to: Issue calling other PBX systems using VoIP with Polycom 501

I'm having some difficulty setting up a Polycom 501 with two different extensions. I've set up in the phone's xml file the proper reg.1withlineKeys = 2, and reg.3 with lineKeys = 1. I have also set up the sip.conf file on the Asterisk box correctly. However, the phone is not making any sort of registration request to the Asterisk server. Is there something else that needs to be

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

Hi List... I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing:

Hi, I've upgraded a few Polycom 501 to SIP 2.0.3b (can't get 2.1.0 because of Polycom's firmware policy, but this is the "latest publicly available" from Polycom's web site). I've noticed that some keys get "sticky" though. Soft buttons for example (i.e. "end call") need to be pressed 2-3 times for them to react. I've downgraded to

> -----Original Message----- > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steven > Sent: Thursday, January 12, 2006 4:37 AM > > This is supposedly fixed in 1.2.1. > The issue is that some companies using PRI are starting their > IVR as the ringback tone. > The problem is that asterisk considers the call not answered > yet, so will not send any DTMF

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No

I have one extension setup for each Polycom 501 I have, and when I try to call out on a conference call, I get "all circuits busy" for the second call. I have one sip trunk set up for each DID that I have through our VoIP provider. Each trunk is capable of having one call placed on it at one time. So, I'm thinking I need a way to tell Asterisk to have the second call go out

In R 3.5.0 using the `encoding' argument of source() prevents loading files from the internet; without the `encoding' argument files can be loaded from the internet, but if they contain non-ascii characters, these are not correctly displayed under MS-Windows (but they are correctly displayed under GNU/Linux). With R 3.4.{2,3,4} there is no such problem: using `encoding' the files are

>>>>> peter dalgaard >>>>> on Sun, 3 Jun 2018 23:51:24 +0200 writes: > Looks like this actually comes from readLines(), nothing > to do with source() as such: In current R-devel (still): >> f <- file("http://home.versanet.de/~s-berman/source2.R", encoding="UTF-8") >> readLines(f) > character(0)

Hi List; Can someone advise me if Polycom support H323 that work fine with Asterisk? And wether this H323 Polcyom devices more costly than SIP Polycom. Also, I am not able to know if new Polycom come with PoE adaptor so no need for PoE Switch (can use normal switch that does not support PoE)? Do I need any special cable for Polycom or normal Ethernet cable? Regards Bilal

Anyone know how (or if it's possible) to get a logo on the screen of a Polycom 501? I've been looking around for hints on how to do it but so far nothing would indicate Polycom supports doing it. -bill

Hi We've setup an Asterisk PBX recently and I encountered the following problem: When [mac address]-registration.cfg file includes the FQDN of the Asterisk PBX for the Polycom SoundPoint 501 phones it will not (even try to) register with the Asterisk PBX unless the DNS (it asks) successfully resolves the name: _sip._udp.[Asterisk FQDN]. Did this happen to anyone else? PS - The

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

I just received my first polycom 501, tried my best to follow all of the documentation and configs on the wiki ( though some conflicted in which case i tried both ways ), and at the end of the day can not connect my phone to asterisk. My Questsions: does the wiki information apply to the 501's? The images and config files at freedomphones, are those supposed to work with the 501? Has anyone

Hi: What's the trick for getting the Polycom IP 501 message light to go on when there is voicemail waiting? Any ideas? -Stephen-

We're looking to purchase new phones for Asterisk. There are a limited number of new Polycom 501's on the market, mostly refurbished available. Can you recommend a replacement phone? What ever model replaced the 501? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL:

The Polycom's need to have their registration time lowered. Set it to 60 seconds which will re-register every 30 seconds. The polycom doesn't have any sort of 'keep alive' feature to keep the NAT holes open. There is information on the wiki fruther describing this and how to set it up if you don't know where to look. p From: "Von L." <methodvon@gmail.com> To:

Hi I am looking for a good wirless headset to use with the Polycom Soundpoint 501 phone. I would greatly appreciate hearing from anyone with good experiences with a specific device. Thanks -- Jim Freeze

Hey All, This is more a polycom question, but involves how to make it specifically interact with asterisk... I have setup a special extension in asterisk which toggles whether our "Night Service" is turned on. However, I would like to have 2 things happen on our Polycom 501 (or 601)s. Where there are buttons which say "New Call" and "Forward", I would like to add a

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