similar to: Polycom Digitmap

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 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

Hi BK - >> The blind transfer does not work. >> >> The way we try to blind transfer a call: >> 1. answer the call >> 2. press transfer >> 3. press blind softkey -> the display shows "Blind transfer to:" and >> cursor is in the second line >> 4. enter the number -> when we enter the second digit of the number >> 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

<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

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

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

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

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

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

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to

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

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

Just installed Asterisk 1.2 on a brand new clean machine running RedHat 9.0. I have a TDM400 card inside. When I boot, the card seems dead. When I do: modprobe wctdm modprobe Zaptel the lights come on and all seems fine, until I reboot that is... After a reboot I have to repeat the modprobe. I shouldn't have to do a modprobe every re-boot should I? How do you get the drivers to load

> >> 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

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

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,

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

So, it seems David beat me to it, and the assert has already been removed, but the failures are still inconsistent. A9-check-all, compiled with GCC: Tests XPASS: LLVM :: ExecutionEngine__MCJIT__test-common-symbols-remote.ll LLVM :: ExecutionEngine__MCJIT__test-global-init-nonzero-remote.ll LLVM :: ExecutionEngine__MCJIT__test-ptr-reloc-remote.ll Unit-tests pass. A9-self-host, compiled with

?Hola! torisa.c (from zaptel 0.50) says: #define MASTERVAL 0x41 /* Enable E1 master clock on Rev. B board */ Does that mean that there is a torisa revB? If so, are the schematics downloadable from somewhere? (www.zapatatelephony.org only has Rev. A ones) Does the Rev. A support E1? (just changing the DS2151 for DS2153, the DS crystal and closing the E1 jumpers, or something else is