similar to: Polycom Communicator Drivers on Wine?

Hi, I know the answer is probably negative, but there's no harm in asking right? I have a Polycom C100 communicator and from what I can find about it, the real work done to make the audio as good as it is, is done in the software they install on Windows XP. It is a USB connected desktop handsfree mic/speaker device and it does work on Linux O.K. but I get some problems with echo when

Hi all, I'm seeing a lot of these messages: [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '2e4a02807750b7024d5ff09c668fa0f7 at 10.0.0.2'. Giving up. [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '0755ad8f40b9d09d491b635e70bb8905 at

Hi all, This isn't meant to be spam I thought some of you might find it interesting. Packt Publishing approached me a few weeks ago and asked if I would like to review a book or two for them on my blog. The first one they sent me is called Asterisk Gateway Interface Programming and has only just been released. It was written by Nir Simionovich. You can read my review here:

I lurk and comment a little on here and have been playing with * for a short while. I am interested in hearing about the pros and cons for using a database backend to Asterisk. My current setup is simple, out of the box with config files in /etc/asterisk and logs etc going into /var. I notice a great many of the contributors here seem to use a db backend (is this also called Real Time

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 Guys This has been discussed a little in the list before so my apologies for sendig it again but I have done what others have done in the list but to no avail. I have configured Asterisk to send the callerID of extension phones as "firstname lastname" and that seems to work well and extensions show calls originating on other extensions in this format. I set the following in

Hi, I have three Snom M3s at the moment but getting pretty fed up with the issues :( I am UK based and would be interested to hear of other peoples recommendations. Key features :- * VM Notification * Good Range * G729 codec support * Common/Private Address Books per Handset(s) TIA, Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be

Hi, I have questions about the Polycom 601 and side card.... 1) In the side card the lights all time off... But all functions it's ok. I need help with extension module of polycom... All works fine... But lights not work.... So... I don't know when any person or extension is busy... Any ideas? , Olger On 3/27/06 11:34 PM, "asterisk-users-request@lists.digium.com"

Hi, You need to have completely replaced the Microsoft driver, because it doesn't support the headset or ctp Bluetooth profiles. This gave me fits! I followed the instructions at http://www.windowsdevcenter.com/pub/a/windows/2005/07/05/bluetooth.html and it works with both a Plantronics and a Motorola Headset, and I can answer calls with idefisk, eyebeam, x-lite, and kapanga. If you end

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No

Hi, I know this is a little OT but there are many Asterisk users of the excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is probably newsworthy for them. One of the biggest bug bears has been no mute function on the handset. When I woke up this morning, the handset told me there was a firmware update. I updated and then visited the web site to find out what had been

Hi Alexandru, > The compilation error is : `error: ‘LD100’ was not declared in this scope.` > > On Mon, Jan 28, 2013 at 11:31 AM, Alexandru Ionut Diaconescu < > alexandruionutdiaconescu at gmail.com> wrote: > >> Hello everyone, >> >> Can you please tell me if it is possible in LLVM to cast a `Value*` to an >> `Instruction*/LoadInst*` if for example

Hello all, I''m mucking around trying to get my desktop computer (Ubuntu 11.10, BlueZ 4.96, ASUS USB-BT211 USB Bluetooth adapter, no microphone hooked up at the moment) to act as a a hands-free device for my phone (Nexus S). It was relatively simple getting it set up for A2DP, and I was able to play music that is on my phone through my desktop''s speakers (using PulseAudio

I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible

Hi! im trying tu run "microsoft office communicator 2005" and i cant resolve this: fixme:ntdll:NtConnectPort (0x1434f8,L"\\RPC Control\\epmapper",0x33ecd0,(nil),(nil),(nil),0x33ecf8,0x33ece0),stub! i google it all nigh long and i just cant find the way!!!. I need to connect to LCS 2005 because my company switch from Jabber to LCS. I tried pidgin and miranda-im+sip but didnt

The compilation error is : `error: ‘LD100’ was not declared in this scope.` On Mon, Jan 28, 2013 at 11:31 AM, Alexandru Ionut Diaconescu < alexandruionutdiaconescu at gmail.com> wrote: > Hello everyone, > > Can you please tell me if it is possible in LLVM to cast a `Value*` to an > `Instruction*/LoadInst*` if for example `isa<LoadInst>(MyValue)` is true? > In my

> Hi > > I am trying to setup icast2 and SAM2 Broadcaster > > I have looked at the icast2 config and some of it was a little confusing - > basically all I have changed from the default was the passwords - all IP and > ports remain as the default. > > I have setup SAM2 Broadcaster for MPEG3pro encoder and pointed it and SAM2 > to 192.168.0.3:8000 and mount as /C100.

I am able to pickup, hangup, and flash, using the buttons on the phone with all of the soft clients and both of the headsets I mentioned below. I don't believe I had to change anything on the client side, just had to get the ctp (cordless telephone profile) working in the bluetooth stack, which was a pita. I struggled with the same issue...I could use the headset, but that's not very handy

Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip communicator to pingtel or Voip phone. but now i'm can't make calls between 2 sip communicator.. it mean i can able to make a call and receive.. but

Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip communicator to pingtel or Voip phone. but now i'm can't make calls between 2 sip communicator.. it mean i can able to make a call and receive.. but