similar to: Polycom 320 Issue

Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying

Hi all. I just installed bunch of IP450's and everything went well and my customer is happy.... except that they are unable to transfer calls to other extenstions. They can dial them directly just fine. However, when the user is in a call and presses the transfer soft key, they get dial tone, and start typing the extension, say 1008. But by the time they get 100 typed in, the phone tries

hi there I'm setting up asterisk@home and I'm using Polycom IP 500 phones. When I call a number that has a digital receptionist (i.e. "dial 1 or such and such, dial 2 for this and that...") the Polycom doesn't seem to send the extra digits. When I try it with X-Lite things appear to work fine, so I think the problem is with the Polycom configuration. Here's some

I recently had a bear of a time getting a Polycom Soundpoint 500IP up and registered.. Now that its registered I ran into a problem w/ the dialplan. Needing to dial x101 I'd dial 10 - then get a fast buzy.. Also making a local call - dialing 95551212- would give me a fast busy after the 7th digit - so 9555121.. Same w/ LD calls... This dialplan really got me down as I didn't find

I'm stumped on this one and any help would be greatly appreciated. I'm just trying to get my Polycom 601 to have multiple extensions on it. For example, on line 1 I want extension 21, on line 2 I want extension 22, and on line 3 I want extension 23. Ideally I'd actually have each extension appear on 2 lines and therefore filling up all 6. I should be able to do that with the

Calling all Polycom gurus: I am using Polycom IP601 phones with Asterisk 1.4.21.2 In all Polycom phones, I set the following in sip.cfg. <dialplan dialplan.impossibleMatchHandling="2"> </dialplan> (I leave the digitmap unchanged because I thought setting impossibleMatchHandling will ignore the bitmap) ...so that I could dial any number by entering a variable-size

I'm using Polycom Soundpoint phones and I want to use some extensions beginning with # for features setup. I'm getting the fast busy "can't match it" signal. I want to match #50 for call forwarding, for instance, and #505551212 to set the call forwarding number and turn it on. I have tftp set up and sip.cfg contains the following: <dialplan

I was hoping that someone may be able to shed some light on some issues I'm having on trying to get an Asterisk test server up and running. At the moment I have the basics, two Polycom hard phones (301 & 601 with expansion unit (which oddly will not power up)) that can call each other, log into voicemail (one touch) and have custom directories & buddy lists. Unfortunately some of the

I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650. I attempted to simply reuse the existing config files for the old phone on the new phone, but the new phone would lock up on the 4th digit when attempted to dial out certain numbers. So, I downloaded the newest firmware and config templates from Polycom, and attempted to migrate the settings. Seems I'm missing something from

I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above,

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

Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channel 50 DDI service. Zap show channels show and ztcfg -vv looks ok and the zttool show

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

All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch,

Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct.

I hope this doesn't already exist, but I couldn't find anything to help. I am installing a brand new Asterisk server, and want to use the Polycom IP331 phones. Does anyone have any steps on how to configure these? I have softphones working just fine, but for some reason I can't find a clear step by step on provisioning the Polycoms. Any help is greatly appreciated! Mark J.

All, I have a Polycom 650 phone, when turned on displays "Checking application". Can any give me some information as to what is wrong? I have copied the CFG files from a 601 phone to work with this 650. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070429/daadda2a/attachment.htm

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL: