similar to: threewaycalling

Based on the following diff in CVS: -" 't' -- allow the called user transfer the calling user\n" -" 'T' -- to allow the calling user to transfer the call.\n" +" 't' -- allow the called user transfer the calling user by hitting #.\n" +" 'T' -- allow the calling user to transfer the call by hitting #.\n" It

Hi, We're evaluating asterisk, somebody has measured maximum asterisk load (simultaneus calls, calls per seconds...)? Are there any stimation? Thx. Best regards. .G

Hi folks, Is it possible (asterisk 1.6) to trigger the playback of an audio file in the middle of a call using the Manager Interface? I'm looking for something like AMI PlayDTMF command but for audio files. Thanks a lot, G. -------------- next part -------------- An HTML attachment was scrubbed... URL:

My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it works in version 1.0.5 (was working with 1.0.3). I'm using SPA-3000 and dtmfmode=inband -- #Joseph

Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller

Hello everyone. What I'm doing: I've made a replacement for app_queue that uses MeetMe to connect the calling party with the agents. When the call comes in it gets put into a MeetMe room with a nice AGI_BACKGROUND so the calling party can listen to music and announcements until an agent becomes available. So far everything works fine. Now I want to give the calling party an

There is experimental echo cancellation code. Look in speex_echo.h and mdf.c. Unfortunately there's no sample code, and I don't think anyone's had much luck getting the echo cancellation to work at all. Would anyone like to prove me wrong? I'd love to see this work! I know it's really hard but just tonight the person I was talking to asked me, "Is there ever

Hi... I have the next problem.. I have a FXO card with i can make calls but i cant recive calls. At the consol, i get the next error: -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/2-1 WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook in strange state 6 on channel

We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed

Hi! I'd like to ask whether someone did test Opus in real-world VOIP (SIP). Did someone e.g. some characterization about sending faxes or DTFM through Opus? Does it work and if yes for which bitrates? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20130518/907c5cbf/attachment.htm

[Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid extension '11', but no rule 'i' in context ...[snip]... When testing IVR and pressing "1" from my Grandstream SIP-phone, the above message is printed on the Asterisk CLI. How come Asterisk receives my "1" as "11" ?? Settings in my SIP-phone are : Send DTFM : via RTP(rfc2833) &

Hi Gang Not a specific Asterisk Question. But I wonder, if the called party replies with 183 + SDP indicating support for telephony-event. Should the caller be able to send DTFM Tones? Swiss Railways uses an IVR that kicks in before the call is answered. So far I have found no SIP Phone which would allow sending RFC4733 during the early audio phase (so I cannot test if Asterisk would forward

I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The

Hey all, posted this the other day, but re-read it & realized I didn't give enough info to be useful.. so I thought I'd try again.. I'm using AAH 2.0 (* 1.2.0) and am unable to transfer a call when I initiate the outgoing call. In AMPs general settings, I've tried changing the Dial command using tT but transfers are only available when I'm the recipient of the call, not

Hi all ( dev & user list ), I'm starting to implement the missing features in Meetme application : 's' -- send user to admin/user menu if '*' is received Line 438 -------- app_meetme.c ----------------------------------------------------------------------------- else if ((f->frametype == AST_FRAME_DTMF) && (f->subclass == '*') &&

Hi! I'm trying to set up an asterisk based PBX with a TDM400P +2 FXS +2 FXO modules in UAE/Dubai for home switching / voicemailing. I am using the card Asterisk/Zaptel 1.4.0. I want to include a special route when a certain caller calls into via PSTN. The problem is that I cannot detect the Caller ID. I tryed various setting (cidsignalling, cidstart) in my zapata.conf, here is the last

Me again, another problem. As I said before, I have 2 lines going into "incoming" context. When client calls, I press Flash, client hears music on hold (only on voip line as said in previous post), when I get back and press Flash again to get back to my client I cannon hear him, but he hears me without problems. I have just tested in on the LAN, same situations, happens everytime.

Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller

Hi, When i include a ',' in the application data, for example: exten => 1,1,Plaback(echo,file); [My file name is "echo,file"] the Playback application receives echo|file as data parameter. How can avoid asterisk change the ',' character? (The change code is in asterisk/pbx/pbx_config.c Ln 1150). I'm developping an application for asterisk and i need to pass

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the