search for: konference

Hi List, I am trying to get DTMF into conference room. for conference I am using Konference module. Konference don't have an option of DTMF gets. Is there any way by which I can get DTMF within conference room? ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists....

hi,list i installed App_Konference in my Asterisk 1.6.2.11. and i write in dialplan like this: exten => 95040,n,konference(1234,RVxTH) it works fine. but I want to record the conference, if use MeetMe , i can use 'r' option to do this. but there is no 'r' option in konference , Could you tell me how to do this?...

HI, I have installed asterisk 1.6.2.18 with konference 1.7, All things are working fine but when we start taking DTMF then key 3 not get my asterisk. When we use landline number(dedicated number) than all DTMF is capture and asterisk work fine. In case of mobile only key 3 don't work. Strange when I use my touch screen number then most of the DTMF...

...my mobile, I am getting DTMF into my asterisk server. For getting DTMF I have use one opensourse application which gets events from asterisk server and store into database. And after that I made my own script to gets these DTMF keys and play some IVR into the conference. For conference I am useing Konference module. I have check sip.conf and set *relaxdtmf=no* I am using Dahdi and set chan_dahhi.conf as below [channels] resetinterval=never context=incoming usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallf...

Hi list, I have write down my code on which chanspy not working when I make a group with name of spy. Please help me where is the issue on that. a) caller will call this number to join konference and spy group exten => 43681111,1,Answer() exten => 43681111,n,NoOp(****${CHANNEL}****) exten => 43681111,n,Set(GROUP(${CHANNEL})=spy) exten => 43681111,n,Set(a=${GROUP_LIST(spy)}) exten => 43681111,n,Set(b=${GROUP_LIST()}) exten => 43681111,n,Konference(VADSTR) b) spy will dial...

Hi, I have a requirement where the DTMF entered by a member in konference is passed on to the other members. But the DTMF is not being recognized, when checked the events from manager API, I do see DTMF event being passed, but some how it is not passed to other members. This tells me - may be it is not an asterisk issue, but more a konference application issue. Is thi...

Hello I've never used Asterisk for a three-person call, and would like to check that MeetMe is the way to do this. The ADSL modem provided by my ISP offers free calls to landlines/cellphones when using a handset connected to an RJ11 port on the modem. A three-person call can be set up by using the standard PBX sequence: 1. Using the handset, call party #1 2. Hit "R" key on

Hello I'd like to know what my options are to bridge two channels after calling each through Dial(). I know about MeetMe, Conference, and Konference. Are there other options available just to bridge two calls? www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe www.voip-info.org/wiki/view/Asterisk+cmd+Conference www.voip-info.org/wiki/view/Asterisk+cmd+Konference I'd like the simplest possible solution since Asterisk is running on a non-x86 a...

...e: Unable to write to alert pipe on Local/conference at veco-044d;1 (qlen = 1): Broken pipe! -- Executing [conference at veco:2] NoOp("Local/conference at veco-044d;2", "Trying to start conference ConferenceA_test") in new stack -- Executing [conference at veco:3] Konference("Local/conference at veco-044d;2", "ConferenceA_test") in new stack [Feb 10 14:14:38] WARNING[15571]: channel.c:1065 __ast_queue_frame: Unable to write to alert pipe on Local/conference at veco-044d;1 (qlen = 2): Broken pipe! [Feb 10 14:14:38] WARNING[15571]: channel.c:1065 __...

There is one more thing to try: http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html I would appreciate if anyone can comment on the feasibility of playing an audio file to the caller and callee using ControlPlayBack and appkonference. Much of the reviews indicate that appkonference is an over-kill for an audio as its main functionality is with video. Going pa...

Hi, I have released AppKonference 2.5 today. This release fixes a bug that can cause audio problems when conference frame caching is enabled. It also fixes the spy feature so that more than one spyer can spy on a channel at the same time. If more than one spyer is unmuted, their audio is mixed and whispered to the spyee. -- Paul...

Hello, I tried the migration from samba 3 to a sernet-samba-ad-4.1.9-8. My samba 3 has tdbsam backend, and "Full Name" (pdbedit -Lv) contains umlauts. If I run /usr/bin/samba-tool domain classicupgrade, I will see error: "Conversion error: Incomplete multibyte sequence" I had in my old smb.conf: display charset = iso8859-2 dos charset = cp852 unix charset = iso8859-2 In

Are there any adapters that would allow me to connect asterisk to wifi or we are not there yet? I have Digium adapter S101i that was discontinued but similar device that would connect to wifi network and a cell phone would be handy. -- Joseph

...ek wrote: > it can be done with sccp/skinny protocol with callmanager, it's > possible to make ad-hoc multiparty conference with audio mixing in > callmanager (not in phone) > but probably, it's not possible with sip :-( > even with sip phones, this is not true multiparty konference, but > only three-way calling :-( > imho, currently only way to make true multiparty conference with > sip/asterisk is meetme, but it's not very usable, because calls > must be initiated from each user... > .. so that makes me doubtful. So! The questions are : 1. Is t...

...Flash(), which is apparently the equivalent of hitting the R key on European handsets 4. Asterisk calls the number, and once the remote party has answered, bridges the two channels Ideally, I'd like to do this entirely through extensions.conf, and avoid callling an AGI script or having to add Konference: This is an appliance, so RAM isn't plentiful, it runs uClinux instead of run-of-the-mill Linux, and I would like to avoid having to patch Asterisk. I've seen articles about Call files. Is this the easiest way to solve this problem? www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+...

Hi all, I'm using speex through my own python wrapper based on ctypes. In my application I'm forced to process relatively large chunks of audio data (250ms). I was able to supply sufficiently large buffer to the SpeexBits structure and then encode using (code snippet) _speex.speex_bits_reset(byref(self.bits)) for i in range(0, len - self.frameSize, self.frameSize):

Hello, I need migrate Samba PDC to Windows Server. Is there somebody, who had managed to do this? Does exist some manual how to do it? Thank you, Best regards, Jan

Hello, I have Dovecot configured for multiple domains (usernames are user at domain.tld) and I wanted to enable shared mailboxes. But there is some problem - if I share a folder, other user can't see it. In the log I foud: > Apr 22 19:21:02 veverka dovecot: imap(user at veverka.tld): Error: Couldn't create namespace 'shared.' for user petr: userdb didn't return a home