similar to: Polycom IP 500 - Quality Issues

Hello, Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP phone? I am getting failed registrations, and the Polycom documentation is not very precise. Their web interface isn't helping much either. Thanks in advance, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am having a problem with Red Alarms on X100P cards. The most frustrating thing is I can not duplicate the alarms, therefore am not sure how to solve it. I have read after searching posts and the web that you can try different signaling methods which may help alleviate the problem. There is fxsks (Which I am currently using), fxsgs, and fxsls While reading the Digium site about these different

Hello, About every 2 or 3 days I notice in the messages log file: Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 2: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 3: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 4: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 5: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on

I have upgraded the GXP-2000 to the newest firmware 1.0.1.12<http://1.0.1.12>and the phone is much more usable However, I still have two slight sound quality issues: 1) There is static on the line at all times. It is not that noticable to me, but when I make calls out the PSTN the person on the other end hears it. If I use a Cisco ATA with an analog phone and call the same person again

Hello, After reading the lists and taking reccomendations from TC, I have finally given up on the echo can built into asterisk. I am sick of hearing complaints from users, so the money spent on a hardware echo can will be worth its weight in gold. I am curious however, about some setup and component requirements. It seems as if every telecom place I call, either never calls back or doesn't

Hi Lately I've been having problem with the handsets on the polycom phones. The agents are complaining that the humming/buzzing can be heard on random days. When the agents touch their computer the buzzing may increase louder or reduced. Also when they grab the handset cable, the buzzing noise is less loud and is loud again when I let it go. I have tried changing the handsets, problem not

I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for "buzzing" on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

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 rsync people Today I was recovering data from a beginning-to-fail external USB hard disk. I started with my usual 'rsync -av --ignore-errors <source> <dest>', and that was fine until it got to the first I/O errors. It paused but continued after the first couple of errors, but then the disk started buzzing and rsync gave error messages for every file (I'm afraid

Thank you for your response. The point is that I have configured my system to shut down at the event "upsc Inform@localhost ups.status"!="OL", not at "LB". So there will be no /etc/killpower file created. At an ups.status != "OL" event, the following script named "notifyme" is run: echo `date` >> /root/upsLog echo " No power, shutting

We are using an MGCP configuration. There seems to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is how our Vendor sees it: Here's what I see. 1. The first call is initiated. (CRCX) The interesting thing here is that the CA (Call Agent) tells us to go directly into sendrecv mode which means that we start streaming audio immediately. All other CAs that

I've spent the last week or so troubleshooting echo problems at my Wife's business, and I've been able to clear up about 99% of the echo, but there is still a little residual echo that I can't seem to "tweak out". The users describe it as "buzzing or crackling", but what it sounds like to me is a slight echo, but just of one syllable of the word.

Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using

Hi All, I've been playing about with the RECORD FILE agi function and am finding two distinct problems with the resulting wav file when using a non zero sample offset. Specifically, I call the function with a zero offset and a given filename (the "original" recording), and then later call it with the same filename and a non-zero offset (the "overdub"). When I do this, I

I've got a Digium TE405P feeding three Carrier Access AB-II units located in three separate suites of the complex my company resides in. I've had one AB-II connected over about 15' of cable and running for several weeks now, and it seems to be perfectly happy. Over the weekend, I added two more AB-IIs, one in each of the other two suites. The second unit is about 150 cable feet away

On Sun, 2003-03-16 at 19:24, Ted Faber wrote: > On Fri, Mar 14, 2003 at 02:48:26AM -0500, Andrew J Caines wrote: > > Starting the server as normal with xinit gives normal startup messages, then fails > > with a signal 11: > > I have this problem, too. Same nv server. I'm happy to provide > details. One that might help is that this is an Athelon XP 1800. > >

Hi, I am attempting to create a intercom buzzer system using asterisk as a back end. Most is figured out except the actual action of buzzing the door. I need to detect whether a DTMF key was pressed by the the called party (the resident). Is this possible to do using just a dialplan? I can't see any options on the Dial command that would lead to this, am I looking in the wrong place? I looked

We are working on a new implementation of asterisk. We are using a fiber-served WorldWide Packet switch at the home that incorporates a VOIP T2 switch that feeds 2 POTS connections. We are told that the T2 is programmed with code that follows the PacketCable spec. This version has a problem with the 'congestion' message which is based on the MGCP RFC (2705) spec. This causes the T2 to

Hi list, I have a client with a strange requirement: putting a noise gate on the Asterisk channel. For those who are not familiar with them, noise gates are used in musical instruments to avoid entering low-level noise into the amp system. What they basically do is, they measure the volume of the channel, and when it's too low they just let the channel close, i.e send perfect silence,