similar to: Polycom 600 problem

I have noticed the same behavior. I have several different type of phones, and all my Cisco 7960's do not register anymore with the latest CVS release. I have not made any config changes, only the binaries have changed. -brian On Tue, 04 Jan 2005 at 16:38 Andrei (MPI) (asterisk@markovprocesses.com) wrote: > Hello All, > > Sorry if it is known problem. > > I have tried to

I heard Sipura had really awesome documentation on the SPA-2000 and SPA-3000, but you have to email them for it. When I did, they said I had to get it from a reseller. It's been a while since I bought my units, I don't even remember where or who they were bought from. Can somebody email me the documentation for these devices? I'm quite interested in knowing what every one of their 200+

Greetings, Recently there has been talk of the presence/buddy feature with asterisk and Polycom phones. I have it setup, and working as expected, however I can only get 7 buddies to appear on the screen at any given time. Has anyone gotten more than 7 buddies to appear? I'm just trying to find out if this is some polycom limitation, bug, or my error. Thanks, Matt -- Matt Gibson VOIP

Hello everyone, I'm sure someone had an experience arranging hunt-group setup for incoming calls on T1 PRI channels of Digium TE110P card. For instance, I have main DID channel associated with number (555) 222 0001. And I have whole bunch of other DID channels on same T1 card like (555) 222 0090, (555) 222 0091, (555) 222 0093. My goal is when a call comes to the main number which is

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, We have a scenario where we receive calls from 2 different places: 1- Avaya IP Office 2- CIC Interactive Intelligence PBX and the calls are transfer automatically to an Asterisk Box. The problem we are experiencing is that more that half of those calls come with Echo and Jitter. For outbounds calls everything works perfect. I'm on RH9 Stable v1.4 Version, using an quadspad TMD400P digium

Hi, I have was testing some of the different ring types with my polycom 500, and the ALERT_INFO settings. Now when my phone receives a call it won't ring. All the other phones ring fine, and my phone wasn't the only one I rebooted with the changed sip.conf and impd.conf. I have reverted back to a standard sip.conf and impd.conf and I still can not get my phone to ring for any incoming

Got my first round of IP500s in today. Anybody have any example sip.cfg files they'd like to share? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/a923e094/attachment.htm

I just set up a Polycom 500 on *. Every few calls I make, the call connects and the receiving party can hear me (thru Broadvoice), but I still get ringing on my end, as if they never picked up. * logs look just fine. Does any one have any suggestions? Thanks. ________________________________ Adam S. Robins Executive Vice President & CIO PHARMACENTRA, LLP 5901B Peachtree Dunwoody

Ok, so I emailed the list earlier about my polycom phone not ringing when anyone called in. Well, polycom support said that is impossible that this could happen because of a change in a configuration file. However the new phone arrived today (a refurb.) and it also would not ring. So I obviously got rather frustrated and blasted away all of my configuration files from the FTP server. I then copied

I'm still trying to work this out. I've got this in my sip.conf [1003polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.215 context=default mailbox=1003 subscribecontext=phonestatus [1004polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.214 context=default mailbox=1004 subscribecontext=phonestatus And this in my extensions.conf [phonestatus] exten =>

Hi, We are in the process of moving from an Avaya Definity to Asterisk for our institution's phone system. I got one feature that the Avaya had, which I have not been able to reproduce with Polycom phones and Asterisk; since this feature seemed so small and useless to me when testing, I kind of ignored it. Now I am getting more "I miss that" requests than I expected. =) On the

> What is a `"logically"-widened alwaysinline wrapper for the vector function`? Can you provide an example? Also, what is the `tricky processing` you are referring to that the vectorizer should care about? For the case mentioned earlier: float MyAdd(float* a, int b) { return *a + b; } __declspec(vector_variant(implements(MyAdd(float *a, int b)),

I have a .csv table named mailing.csv as below. It consist a receiver, subject and sender. Receiver subject sender 1 Adrian Cole RE: [WHIRR-117] Composable services Tom White 2 Adrian Cole RE: [WHIRR-117] Composable services Tom White 3 Adrian Cole RE: [WHIRR-117] Composable services Adrian Cole 4 Adrian Cole RE: [WHIRR-117]

Hi Francesco, > I am crafting the attribute so that it makes it explicit that we are using OpenMP and we are expecting a Vector Function ABI. I just thought that another option would be to force FE to always emit "logically"-widened alwaysinline wrapper for the vector function that does the arguments processing according to ABI inside (we need that info in the FE anyway). That way

I have my * and polycom system setup to do Auto-Answer for internal SIP/Staff calls, and I am running into an issue with this and the polycom call transfer feature. * is seeing a new call come through from the polycom and is then transferring the call over. I need to know if there is some way I can grab a message from the SIP header or something to determine if I should not set the ALERT_INFO tag

Hi Aki, Thanks. I was especially interested in documentation related to dovecot and it's users permissions, the way in which dovecot uses users. Till now I found only spread information on different articles from dovecot's website. Thanks, Mura Andrei On Sat, Apr 11, 2020 at 9:49 AM Aki Tuomi <aki.tuomi at open-xchange.com> wrote: > Hi, > > >

I have looked at the wiki and the mailing list. But I need to find how do we setup the external IP address and the rtp ports for the Polycom IP-500 and IP-600. There web interface has a nat setting but can't find instructions on how to set this up. I would like to set this up via there ftp file setup instead of via there web setting. Also There QoS settings are set to 5 and 2 but there it

Maybe you need to subscribe to that folders? On 4/25/20 3:07 PM, Andrei Petru Mura wrote: > Rectification: For example, created a_folder would be found on server > under Maildir/cur/.b_dir. > would be: > For example, created a_folder would be found on server under > Maildir/cur/.a_folder. > > On Sat, Apr 25, 2020 at 2:45 PM Andrei Petru Mura <mapandrei at gmail.com >

Could someone please direct me (via personal email) to a provider with good prices on Polycom Soundpoint IP 500's with POE cables? I need 14 of them. Thanks, Adam ________________________________ Adam S. Robins Executive Vice President & CIO PHARMACENTRA, LLP 5901B Peachtree Dunwoody Road, Suite 380 Atlanta, GA 30328 Office: 770-395-0088 x34 Fax: 770-395-0989 Mobile: