similar to: Linksys SPA-942 Phones

Anyone use the LDAP feature yet on the polycom phones? If so how well does it work? How are you using it in your environment? http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/applicati ons/corporate_directory_access.html Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 roy at

Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy

Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy at manistee.org -------------- next part

Just wondering if this is possible: Make a call from a registered sip extension (Doesn't matter if it's internal or external) during the call press a key sequence let say *90 to start recording call. When the call ends the recording automagically goes to their voicemail. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee,

I've been reading up on followme app for asterisk 1.4 and I have it working but I was wondering if the following was possible: Based on followme.conf present the caller with the option to locate the person: Call comes in (external or internal) and rings extension with followme configured. Before the followme app is initiated the caller is prompted to locate the person (by pressing 1 which

Hello List, I'm looking at the page command. I was wondering if there was a way to set a wild card to dial all registered sip devices. For example page all 1XXX extensions. Thanks in advance Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy at manistee.org -------------- next

Is there a way to tell asterisk to beep every few seconds rather than play MOH. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy at manistee.org -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to understand something that just doesn't seem to compute. How can companies like Cisco justify selling their hard phones for as much as they do? I know there is a matter of recouping R&D costs but when you look at the iPhone with all its amazing features for less than $500.00 it just doesn't make sense. Am I the only one that thinks this? Roy Anciso Director of

I'm not sure if anyone has done this before or not but, I was able get the Cisco IP Communicator soft phone to work with Asterisk using SIP. Thought I would share my experiences. The key is on the installation. To have the software use the SIP protocol type the following command: "msiexec /i CiscoIPCommunicatorSetup.msi /qb SIP=1". After installation configuration is just like

Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy at manistee.org -------------- next part -------------- An HTML

Hello List, For those of you using Cisco phones, did you have to purchase a 'SIP license' for each phone? Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy at manistee.org -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List, We purchased a TE120P card from Digium and it works great. The only problem is that we are still experiencing echo on some calls. I've tried various echo cancellers (right now we are using OSLEC) and still no luck. My question has anyone gone from the TE120P to a Sangoma A101D-X Single Port T1/E1/J1 w/ echo cancellation? Have you noticed a difference? Also I called

Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with

For those of you running Cisco phones, did you start out with a Cisco CallManager and move to Asterisk? And if you did switch do you find that you or your users are missing features they once had? How have you handle the issue? Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy at

Hello List, Not sure if this will be helpful but I made changes to the original Cisco directory.php.txt script and applied them for use on the Polycom phones. This will create an extension directory and alphabetize it based on the sip registrations you have setup in sip.conf. Note that this only seeds the phones and does not synchronize them. Anyway thought it might save people some time. To

Hi people, I'm having problems of connection with a Linksys SPA IP PHONE 942 when I use the WAN port, most of the time when I try to connect to the network or restart the IP Phone I can't get internet connectivity . I tried using both static IP and DHCP, but the problem is the same. Some have had similar problem with this brand of IP Phone? Thanks for the help and attention. Hugs

phones or extensions in the system. On 3/12/08, Anciso, Roy <roy at manistee.org> wrote: > Is there a limit on how many phones you can use? I couldn't find > anything on the website about this. > > > > ________________________________ > > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua

Victor Hoodicoff wrote: > > > I think your impressions of Aastra are outdated. Install the latest > firmware, download the latest documentation and test and THEN give an > opinion! Did you miss the part when I wrote I have Asstras sitting on my desk collecting dust. I program on average about 5 per month, deal with about 40+ per day. They're as impressive as that Hyundai in

Hi I have a customer who is using Linksys 942 phones. When they try to transfer a call the Asterisk CLI reports that both legs of the call must exist on the server. The call they are trying to transfer then drops. Does anyone know why this is and how to fix it? TIA Regards Jon ___________________________________________________________ Want ideas for reducing your carbon

Hello I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921 I'd like to have some user feedback about how those phones perform, and whether their LCD screen displays both the caller ID name and number (The GrandStream BT-100 only displays numbers, which isn't very helpful). Thank you.