similar to: Problem with ringing (or absence of...) with Polycom forwarding

Hi, This is a bit off-topic, but still related to telephony. Is there a barebones TAPI driver that exists that would allow me to call up a command line with, as parameter, the number to dial. For exemple, Outlook integrates with TAPI, so that TAPI driver would allow me to call my own app with the phone number as argument. ex when clicking on 555-555-5555: the TAPI driver would call

Hi, I'm trying to make an apparently simple thing work, but I don't see how it is possible with Asterisk. This is my extensions.conf: exten => 1234,1,Dial(SIP/123456/555-555-5555|20|D(7777)) ;After call connects, send DTMF 7777 exten => 1234,2,VoiceMail(1234@context); What I obviously want is that if nobody answer the call, go to voicemail. Basic stuff. Problem is Asterisk

I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension

Interesting. Are you using the included cbend.php script to terminate conferences? I occasionally get questions about using WMM with Confbridge, and to date I have not had an answer . If you can provide details, even vague ones, about how you did it, I can update the WMM package. Dan -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at

Here is what I had to change on the phone1.cfg file: I had this value in my 1.6.7 file, put in there following suggestions from the Wiki (http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501) : reg.1.server.1.expires="30" Now, this worked flawlessly with 1.6.7. But with 2.x, this seizes up the phone with a huge CPU load (approaching 100% at times) and makes it

I have an odd issue, where a polycom 601 stops ringing, or more properly, maybe, stops *being* rung, when a call comes in. Other phones/extensions, continue to work fine, they being run at the same time. My dial plan works fine (?) seems it will ring properly, right after a reboot. It works fine for outgoing calls at all times. Hints? joe a.

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 have polycom phones setup in an office connected to a cloud asterisk server. The polycom phones can call out just fine - audio just fine. However a call coming into the cloud asterisk answers fine - get the autoattendant, enter the extension and the polycom does not ring. The CLI shows that the correct SIP extension is being Dialed (SIP/524) Looks like I'm getting a 401 permission

Ok, so I emailed the list earlier about my polycom phone not ringing when anyone called in. Well, polycom support said that is impossible that this could happen because of a change in a configuration file. However the new phone arrived today (a refurb.) and it also would not ring. So I obviously got rather frustrated and blasted away all of my configuration files from the FTP server. I then copied

Hello list: I have a very odd problem. Seemingly randomly, my Polycom IP600 phones will ring without a call being placed to it. That is to say, a random phone will ring. Nothing shows up under Caller ID. Even the buttons that light up to show an incoming call do not light up. If you pick up the handset, you can hear the phone ring through the speaker. Hanging up the phone makes it stop

Hi there, I''m working with some legacy data where customer phone numbers are stored with hyphens between the area code, exchange, and number (e.g. 555-555-5555). Is this the best way to store a phone number? Perhaps not, but it''s the way they were being stored, so I have to work with this format. Right, so when I save a record the log tells me acts_as_ferret indexed the

Hey, all. I've got an office set up with Asterisk, and forwarding's got a bit of a glitch: When they forward, they listen for the remote phone to ring, then hang up. If the remote phone doesn't connect, it goes to the original phone's VM. Is this Polycom's "fault," or Asterisk's? I've been reading up on blind/supervised forwards, and, honestly, have

Hi, I've had a functioning Asterisk system (1.2.18), which I haven't reconfigured in any way, that is just now refusing to forward calls. I only have Polycom phones. When I use the phone's forward feature (forwarding the phone with extension 204 to extension 206, which used to work as recently as yesterday) I get this in the console: "called sipreg-12344". No ringing,

Hi, I ma using Asterisk 1.2.18 & FreePBX 2.2.1. I have assigned every users in office with Polycom with 2 extensions as below 555 8555 I have configured Follow-me to ring when the users doesn't picks the phone on line 1(555) after 10 seconds & then ring the line 2(8555). But this is not a standard telephony which I have been advised to change like below. If someone calls

Hi, all. When I get calls on my SoundPoints, I only see the number -- is there a way to get the alpha portion of the CID, as well? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi, Where should I go to get the Polycom`s latest official (non-beta) version? I am registered on the Polycom customer website but that doesn't seem accessible. Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061102/284b67a7/attachment.htm

Hi, I know that if you have Polycom phones properly configured, you can use "sip notify polycom-check-cfg SIP_REGISTRATION_ID" to have the phones download the new configuration from the provisioning server and reboot. Is there anyway to send the same command to all peers (let's say I had 50 polycom phones that I wanted to reboot)? Thanks, Mike -------------- next part

Hello, I''m having a heck of a time getting Single Table Inheritance to work in my app. I have the following models and have scaffolded out Person and Contact. person.rb class Person < ActiveRecord::Base has_many :contacts end client.rb class Client < Person end contact.rb class Contact < ActiveRecord::Base belongs_to :person end phone.rb class Phone < Contact end In