similar to: Need help with Cisco 7960

Hi all, I am trying to isntall the v1.6 version of Asterisk on my Intrepid system, but I get an error after I have typed make: [CC] manager.c -> manager.o manager.c: In function ?action_getvar?: manager.c:1732: error: ?SENTINEL? undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears

Hi all, Sorry if this is a little off topic, but I just want to know a thing here. What system is used for sending out the caller's number in the US? Here in Sweden we use DTMF to send the number out. I just need to know what is used in the US since I don't think I will be able to use an American caller ID device over here. Many thanks for any info, Christian

Hi, My messages to the list don't get through. This must be the tenth message i am trying to send! Please ignore this test message.

Hi, Sorry if off topic, but is anyone here on this list using it? I am currently searching for a good router for my home network wich supports SIP. Many thanks!

Mine goes to chan_bluetooth. Somewhat of a pain getting it going but I am totally floored with how cool it is! Thanks, Steve Totaro

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On the Python Tutor mailing list Kent Johnson uses a script to find the top posters for the year. If this or something like it has been posted, sorry for the noise; 2008 ==== Steve Totaro 796 Tzafrir Cohen 749 Tilghman Lesher 496 Alex Balashov 354 Olivier 334 Philipp Kempgen 251 Gordon Henderson 242 Atis Lezdins 239 Jay R. Ashworth 230 Doug Lytle 207

Working on Rails Engine. I want to make the controllers customizable whereever the Rails Engine is used. Therefore, I was trying to use `extend ActiveSupport::Concern` on the Engine controller class and include it in MyRailsApp. https://gist.github.com/4185823 # code in my rails engine moduel MyEngine class SomeController extend ActiveSupport::Concern def engine_some_method

I tried using the iLBC codec, and whlie I like it, I ran into a strange issue. I did a few searches on Google and haven't found anyone with the same issue as this. Anyhow, I was using a Plantronics analog headset and box plugged into a Digium TDM card, dialed out over my VoIP provider's IAX channel to the PSTN. I was in a conference call which is running on an Avaya PBX (which

I have a client that want to try the softphone with USB handsets route to see if hardphones will even be needed. I always push for hardphones (Polycom) so I am not sure about softphones or USB handsets. This is going to be for a 300+ seat call center onsite and many offsite, I plan on using OpenVPN for the offsite machines. Any advice on softphones, handsets, or practical experience with

I've just moved into 3.3v PCI servers and found that my clone X100P cards were lying about the 3.3v supported notch. Can I use a Wildcard TDM400P without any modules as a timer for MeetMe in a 64 bit 3.3v server? Will I still need to plug the hard disk power cable into it? Is there a better cheaper 3.3v MeetMe timer? (Boss doesn't trust the kernel timer.) -HJC

Hello, We are creating a web-based conferencing application using Asterisk as the voice conferencing server. This as an open source project. We are trying to determine if there is interest of the community and perhaps work together to improve the application. Using the web application, you can upload your powerpoint presentation, manage the participants in the conference thru the web interface

I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? Regards AK

>Hi Jerry > >As others have eluded to, the 'PTT' feature can mean different things to different >people depending on their background.> > >Is it fair to say that you're looking for a one-touch button which initiates a call to >the other end and causes the other end to automatically answer in speakerphone >mode? >If that would foot the bill then have a

I've been noticing that my group of Polycom IP 501 phones seems to randomly reset themselves nearly every night (I guess it usually happens at night, since I've never seen it happen while I've been at work during the day).. When I say "reset", I mean, the hands free volume and ring volume are set to the default and the call logs (received calls, missed calls, placed calls)

I'd like to get some feedback from users of USB headsets as to what they like/dislike about the unit they own (manufacturer/model number). I'm looking to buy some. Is there already a thread somewhere or a review (I tried to find one with no luck), discussing this topic? Any advantages/disadvantages to using a USB headset over a regular headset slaved off the internal sound card?

Anyone using Asterisk in a Call Center environment? And more importantly is anyone supporting home based remote call center agents with an Asterisk backend? My experience with Asterisk is limited, however I have set it up and installed it previously and had it working for home usage and for simply playing around. My background however is with Cisco CallManager, Cisco IPCCX for call centers as

What headsets have people found work well with the Cisco 7940 and 7960 phones? To date, I have tried a couple of the headsets within the Plantronics H series (H41-N), and noticed that the volume of my speaking is lower over the headset than on the regular handset. I am currently looking for headsets that are known to work well. I do know that Cisco lists the H-91 and H-101 as certified to

Hi all, First off, apologies if this list is not the correct place for this message. Although we have used Xapian with our email services for a long time (over 10 years) we haven't had the opportunity to contribute or participate in the Xapian community until now. We might therefore not be entirely familiar with acceptable protocol here, but we thought others might be interested in a new

This coming out of WWDC. Pretty cool to see this kind of thing happen. -- J "Brien" | HybridIndie Productions | http://hybridindie.com | hybridindie@gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://wrath.rubyonrails.org/pipermail/rails/attachments/20060807/380ae29c/attachment.html

I have Asterisk running on my LAN with softphone clients (SJPhone) and Cisco 7940/60s, all using SIP. I also have a few remote sites connecting to my Asterisk server. I am getting an echo back of my voice when talking with one particular site. The caller does not hear an echo on their end. All calls on the LAN or to other sites do not produce an echo. When the caller places his SJPhone on