search for: freedomphones

...de works and we have been using the same basic client for the last month here at my company and it is working just fine. I'm eager to hear what you all have to say about it, so send me those comments. Here are the screen shots of the same application running on Linux and Windows: http://www.freedomphones.net/astguiclient_linux.gif http://www.freedomphones.net/astguiclient_windows.gif MATT---

...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 gotten one of these updated phones to work with asterisk, and if so can you share your experience. Thanks,

...atest Polycom firmware SIP 1.6.5 and BootROM 3.1.3as per http://www.polycom.com/resource_center/1,,pw-492,00.html Can someone help? We have legitimately obtained these phones but even our official distributor can't get their hands on updated firmware. The only thing we have found is http://www.freedomphones.net/polycom/files/?M=A which has only old versions. Are there any kind Polycom authorized dealers who can help me? -- Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060327/3454efef/attachment.htm

I'm guessing this may be a question for dev list, but wanted to try my luck here first. I'm trying to compile app_amd (Answering Machine Detection) against 1.2.7.1 and am getting some errors. I should point out that I simply snarfed app_amd.c from http://svn.digium.com/view/asterisk/trunk/apps/app_amd.c?rev=14714 ...so if there are other includes and such that are required, that would

I just got a Polycom soundpoint and I set it up using the phone and web based admin. I cant seem to figure out the config files and they are confusing me greatly and I dont have time for it :) Some things are odd, like on every reboot it seems the volume I set is reset? is there any way to fix that. And the ringer seems low. - Even all the way up Anyone willing to point out a good asterisk

I have a couple of Polycom phones, bootrom 2.5.0, SIP 1.3.1.0056. Works great with Asterisk when I power on the phone. However, after some time, say an hour, I cannot receive calls on this phone. On Asterisk, when I do "database show", it does show the phone in there, but it cannot reach the phone. Is there a way to keep alive the connection with Polycom? Any suggestions? Thanks, --

...I have made some changes/improvements/bug fixes to the Asterisk GUI client I have written in Perl/TK and have released a second beta version on sourceforge: http://sourceforge.net/projects/astguiclient/ Here are the screen shots of the client application running on Linux and Windows: http://www.freedomphones.net/astguiclient_linux_0.8.gif http://www.freedomphones.net/astguiclient_windows_0.8.gif Here are some of the changes I've made: - Several bug fixes - New button for monitoring live extensions on Zap channels - Changed the method that the live channels/phone were populated on the listboxes o...

...seem to want to support it at all, even when you do try to go through the proper reseller channels. I think the margins on these phones must be VERY low. To answer your question, a nice gentleman on the list sent them to me a while back. I don't know how he got them. The person who runs freedomphones.net has now posted these files on his site, so everyone can feel free to download them: http://www.freedomphones.net/polycom/files/

...but I think it would greatly help the community in general as well as promote Asterisk in the telecom industry. We may even be able to charge manufacturers for the "privalege" to be Asterisk Certified or sell them stickers and have all the money go to Digium for development. http://www.freedomphones.net/asterisk_certified.gif Let me know what you think. MATT---

I am getting to my wits end with these phones (and so is my boss). I am getting an random echo on these phones and I have an issue opened with Polycom and its been in their research and development department for almost a month with no results. I have noticed that I get a message "RFC3389 support incomplete. Turn off on client if possible" in asterisk. I have researched this and made

Has anyone been able to get the minibrowser on the Polycom SoundPoint IP 500/600 phones working? If so could you share the relevant sections of your config with me?

...I'm stuck. Has anybody some idea for me where I can start looking for further solutions? Any help would be appreciated!! :) Thanks in advance!!! Best regards and greetings from Austria! johnny P.S.: Some detail infos: I'm using Asterisk 1.2.9.1. I installed app_amd from http://www.freedomphones.net/files/app_amd2.c My dialplan context for generating the outgoing calls looks like this (I use AEL): context 1000_amd_tests { _0043[1-9]. => { // be verbose.. :) Verbose(0,InfoInfo ${CONTEXT} - ${EXTEN} called on ${CHANNEL}.); // set our callerId to the correct id Set(CALLERID(numb...

...lan.removeEndOfDial="1"> <digitmap dialplan.digitmap="9[2-9]xxxxxx|9[1]xxxxxxxxxx|[0-8]xx" dialplan.digitmap.timeOut="10"/> </dialplan> This standard dialplan and files for the ftp server was grabbed off this page http://www.freedomphones.net/polycom/files/ Hopefully this message will help someone down the road. -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor grunky@rockriver.net Rockford, IL 61101 815-968-9888 x102

I have these phones working well with Asterisk, thanks to http://www.freedomphones.net/polycom/files The code there is not the most current (and my Polycom Extranet login hasn't gotten updated yet to allow downloads). Been waiting a while for their support folks to update the access. If you know where I can get the 2.5.0 BootROM and 1.3.1 SIP code? Thanks, L.

I've got a couple Shoreline IP phones, their Shoreline model number is Shoreline IP 100. I believe this is actually a Polycom SoundPoint IP 300 phone. I believe the phone is using a MGCP stack. I want to use it for testing with Asterisk. 1. I suspect I need to re-image the phone to make it work with *. 2. How can I preserve the current image on the phone? 3. What is preferred image to use

...ect: [Asterisk-Users] Polycom IP600 stuck at "Running App = sip.ld"(was: Re: Polycom 1.4.1 firmware for IP500/IP600) On Tue, Jan 25, 2005 at 10:00:42PM -0500, Robert Augustyn wrote: > If you have it, can I get a copy please, or possibly can you send it to the > keeper of http://www.freedomphones.net/polycom/files/ > I am looking for the latest boot image too. 1) I have the 1.4.1 firmware. To whom should I send the files? There is no contact info in this web site. 2) Now I am having a problem with my IP600 test unit: While performing tests on the Polycom IP600 I changed a configurat...

I see that the regular site for the Asterisk Perl AGI module is down (or something), so I've made an alternate download from my site: http://www.nathanpralle.com/downloads/asterisk-perl-0.08.tar.gz (just in case somebody was looking for it) Nathan -- ------------------------------------- Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com

I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if

Hello, I'm having problem with a wildcard TE110P. As soon as I load the module (wcte11xp for kernel 2.6.10), it spawns a yellow error with or without an E1 plugged-in. Any one managed to set it up in France? Here are my files: zaptel.conf: span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 zapata.conf: [channels] language=fr context=default switchtype=euroisdn pridialplan=unknown

...esource_center/1,,pw-492,00.html) and I found a link to the polycom resource center (http://extranet.polycom.com/csnprod/signon.html) but I hadn't a username and a password so I had to give up this way. I searched on internet and I found SoundPoint-IP_SIP_1.2.0.zip at this location: http://www.freedomphones.net/polycom/files/ Got the files I put them on my server ,I turned on the phone and then I set its boot parameters (server IP, username and password and a static IP for the phone) to point to my linux account. I connected the phone to my LAN (PC slot) and to my server test (which is not connected...