similar to: Re: Polycom phones-buggy SIP firmware or am Imissingsomething in the XML configs?

> > 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 dials just fine. If > I want to > | dial from extension 100 to

Anyone have experiece with polycom phones? I am experiencing a really weird problem. In an office where I have the following extensions: 100 101 102 103 104 110 111 120 130 140 141 150 200 On the Polycom phones, when I want to dial from extension 100 to any extension 120 or above, or dial out, it dials just fine. If I want to dial from extension 100 to extension 101,or 102 or 103 or 104,

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

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'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'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

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

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 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,

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

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

Does anyone know if the 7912G (which the wiki says can do either sccp or sip) uses the 7940/60 sip firmware? I ask this because the only firmware I can seem to find on TAC for the 7912G is sccp, no sip...if it takes it's own firmware and doesn't use 7940/60 firmware, can someone point me to the right location for it? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone:

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

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 need to purchase approx. 10 phones for a small office implementation. Nothing fancy is required besides a full-duplex speakphone, in the sub-$200 range. I am currently looking at the Polycom Soundpoint 500, Uniden UIP-200 and SwissVoice IP-10. I have searched around and found various posts regarding each phone's ability to work with asterisk (SIP, I probably should have mentioned), but

Hi, Last night I could dial *8 and pickup a call that was ringing to another phone. This morning, I searched on the Web for a solution to mixed up digits when dialing on a Polycom Soundpoint 501. I found that if you go to the SIP page on the phone's >Web interface and change the "Digitmap Impossible Match" setting from "0" to "2" that fixes the mixed up/eaten

I have a number of Polycom phones to setup with my * server. For my initial few phones I hand wrote configs. Does anyone here who uses Polycom phones have some form of management utility for automating their setup? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

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