similar to: Re: Dial Out??

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

Hi - We have a couple of large spaces that we'd like to cover with dedicated conference units like the Polycom Soundstation IP3000. We're concerned about adequately covering the spaces, though, one of which is very long and narrow. I wanted to get external add-on microphones for the IP3000, but I've found that unlike some of their other conference products, it does not have this

Adam, Tor sent this one a little while ago that looks really promising for solving the problem. Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tor Setane Sent: Thursday, January 06, 2005 2:09 AM To: Noah Miller Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re:

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

I have these very phones and took me a while to figure this out myself. The phone considers each line registration to be a line with a second line. So, call line while someone is on a call and another instance will appear below. That means you only need one registered instance for the phones to get two incoming calls. If however you want to have a second registered extension rung if the first

Dean, I appreciate the suggestion. Is it really necessary. I've got slackware already installed on the box. (I consider myself a bit of a Linux guru.), all the drivers, and Asterisk. Everything seems to work fine, EXCEPT getting it to talk to the extension. wouldn't it be easier/faster to just adjust a few settings than to re-initialize my entire box. dean collins wrote: >If

Hi All - I'm trying to implement the auto-answer config from the wiki, but the result for me is that the phone just rings as normal. I'm running firmware version 1.4.1 on an IP500. I've added the following to my sip.cfg: <alertInfo voIpProt.SIP.alertInfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4"/> and this to my ipmid.cfg

Hi - I've just been testing out the latest CVS HEAD (as of about 10:00a EDT today). I'm getting some weird errors. Calls from one sip phone to another have OK audio in one direction and highly scrambled audio in the other direction. The console shows this error repeated ad nauseum during each call: Jul 18 16:08:03 ERROR[22941]: utils.c:509 tvfix: warning negative timestamp

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

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 all, was wondering if someone could assist with a slight problem i'm having. I have asterisk setup with extensions 101 to 109 and am using xlite, grandstream budgetone, polycom ip500 and a couple of other phones. the problem is: 1. only the xlite extension (107) can receive calls. 2. all extensions can dial into voicemail and get mwi when msgs are received. 3. when dialing a non-xlite

Hi - I've told lots of people about the Flash Operator Panel before, but I've never actually used it myself. I've got it all set up nicely, but I can't seem to authenticate to the asterisk manager (which is running on the same box). When I set the op_server.pl to give debug messages, it shows that it can reach the asterisk manager, but cannot authenticate: ** Asterisk

Hi - Thanks for the fast response! > Try changing your dial statment to use a group instead such as > Dial(Zap/g2/3055551212) Same result: Spawn extension (from-sip, 000, 1) exited non-zero on 'SIP/2000-8dca' -- Executing Dial("SIP/2000-a4f1", "Zap/g1/19142246402") in new stack Dec 14 22:11:09 NOTICE[1506]: app_dial.c:800 dial_exec: Unable to create

I have working with a polycom IP500 phone. I like the idea of having each line button on the phone as a separate sip device. If I understand it right, each phone could have three extensions (one for each line.) This would be great since I could then use the dialplan to forward calls to the desired extension. I envision something like this: Extenson 101 - Company-A Extension 102 - Company B

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

Hello, I have configured my polycom ip600 and ip500. The phone works well. But the clock is wrong and flashes the whole time. Drives me nuts! I have set the time offset on the DHCP / boot server. 36000 (I'm in Australia!) It doesn't seem to help... Mike

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

Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? Also, the post about these phones at voip-info.org mentions some

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

So far no one I have talked to has either had this issue or does not know a answer. I currently run asterisk 1.0.9. I have two issues to deal with. 1. The caller waiting caller ID does not show on the uniden hand held that is hooked to a sipura spa 1000 or my Polycom ip500. 2. When a caller is calling in and I hear the caller waiting beep on the line when talking with somone, is there no way