similar to: My new Polycom 450's can't xfer to 4-digit extension

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'm not sure if this is related directly to asterisk or not but on my Polycom 320 when I try to dial a number smaller than 4 digits I get an error on the phone saying "Enter more digits". The dial plan section is listed below. <dialplan dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1"> <digitmap

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

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 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

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,

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

Sirs, According to RFC 2705 (MGCP), these are the parameters that are used in the transactions: ReturnCode, Connection-parameters <-- DeleteConnection(CallId, EndpointId, ConnectionId, [Encapsulated NotificationRequest,] [Encapsulated

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

I'm working on replacing a SoundPoint 600 with a 650. I need to merge these two sets of digitmaps in the polycom sip.cfg file, because the 650 locks up when I try to use the digitmap from the 600. I've included the default digitmap from a 3.1.3 RevB polycom release. I'd like to merge these two digitmaps, but I don't want to reintroduce the lockup issue I was having with the

> >> Jason Brown wrote: > >> | Anyone have experiece with polycom phones? > >> | > >> | I am experiencing a really weird problem. In an office > where I have > >> | the following extensions: > >> | On the Polycom phones, when I want to dial from extension > >> 100 to any > >> | extension 120 or above, or dial out, it

Hello, I have a mixture of Polycom SP IP 500 and 300 phones. I have been reading through the administration manual to try and solve this problem, but I do not seem to be able to find the answers to my question. I figured I would ask here and see if anyone has some suggestions. The problem is kind of annoying. After dialing 4 digits, the phone seems to pause and miss the 5th digit, often

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723

I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer => # This works for all my

<digitmap =20 dialplan.digitmap=3D"[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxx= x x|[2-9]xxxT" dialplan.digitmap.timeOut=3D"3|3|3|3|3|3"/> Don't think it's been modified from the original supplied. ...brig -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf

via google, I found the reference regarding digit maps for the Polycom phones: http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html But I don't see any instruction for prepending digits to the number dialed. Does anyone know how to prepend a digit to the number dialed (from the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura. i.e. Say I want to