similar to: play message to callee before connect toincomingcall

Displaying 20 results from an estimated 4000 matches similar to: "play message to callee before connect toincomingcall"

2005 Jul 02
0
play message to callee before connect toinco mingcall
You can send both paties to a meetme conference with Manager Redirect. Or if you are feeling more adventurous you could load the Manager Bridge patch that I posted to the bugtracker two months ago. It allows bridging of any two existing channels together through a manager action: http://bugs.digium.com/view.php?id=4297 MATT--- -----Original Message----- From: Roland Zagler
2005 Jul 02
1
play message to callee before connect toincoming call
Thank you, Robert! The announcementfile plays well, but at Dial-option "m" i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. Before connecting to SIP Phone 100 the caller should hear a
2005 Jul 03
2
play message to callee before connecttoincomingcall
yes, robert, but how do i "join" the two legs inside a call file or in the dialplan? i have experienced that call files can do a call to a channel and if this call is answered it can either be connected to an extension inside a context or call an application with parameters. roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com
2005 Jul 02
1
play message to callee before connect to incomingcall
try this one exten => 999,1,Answer() exten => 999,2,playback(~.mp3) exten => 999,3,dial (sip/100) exten => 999,4,playbackground(~.mp3) exten => 999,h,Hangup() not sure abt playbackground should be before the dial command or after ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To:
2005 Jul 03
2
play message to callee beforeconnecttoincomingcall
Thanks for the suggestion, C F, but the problem is there is a rather big database application behind with many users, so a static configuration is not suitable for my needs. i am working mostly with realtime and agi. regards, roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: Sunday, July 03,
2010 Apr 26
1
play a sound from the callee before putting it in connection.
Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/111111, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2004 Sep 03
2
Using AVM Fritz!PCI as zap interface
Hello! Is there a way to use AVM Fritz!PCI as a ZAP interface and have it configured for ZAP channels? Thanx in advance! Roland Zagler mailto:r.zagler@fog.at @fog smart partners
2004 Sep 05
4
Asterisk & sudo from httpd
Hello! I want to use "asterisk -rx "show version"" from a php script called in the browser using the local apache, which runs as user "apache". Asterisk is running as root. I added the following line to /etc/sudoers using visudo: apache ALL = NOPASSWD: /usr/sbin/asterisk When i am on the command line of my linux box it looks like this:
2004 Aug 18
1
Asterisk as SMS Service Center
Hello! Is it possible to run Asterisk as a SMS Service Center and does it work over a directly connected ISDN (CAPI) interface card? Did anyone already use Asterisk for that? Roland Zagler mailto:r.zagler@fog.at @fog smart partners
2004 Feb 09
3
Problem with 'ov_open'...
Hey, I've coded an OGG player for Win32 (it uses AL for playback so it's portable to Linux/Mac), but every time the program gets to the 'ov_open()' function, the app completely freezes, and I have to use the task-manager to kill it. I am supplying it with a valid file handle that was just opened (FILE*) and the vorbis file is also a pointer that is not in use (set to null). Any
2006 Jun 24
2
Playing sound before dialing
Hi, I have configured asterisk now with ENUM lookups which are working really perfect. Now I want to play a small soundfile before dial the number to inform the caller which protocl is used (SIP, IAX2 or ISDN). How can I do this? With Playback it doesn't seems to work: [iax2-sipport-out] ; with leading 3 using IAX-sipport exten => s,1,NoOp(Dialing ${DIALSTR} with iax2-sipport-out) exten
2004 Aug 22
1
Spandsp - opencall.org offline
Please can someone send me the .tar.gz file of spandsp, the site is offline and i didn't find it anywhere! Thanxxxx! Roland Zagler mailto:r.zagler@fog.at @fog smart partners
2006 Jun 22
1
PCI or MiniPCI Hardware DSP for G.729, G.723.1 and/or GSM
Hi to all, we are searching for a hardware based DSP solution for use with Asterisk based on PCI or MiniPCI to reduce main processor load and to use embedded boards with Digium E1/T1 cards like TE410P. does anyone know about any manufactorer of those cards or someone who is able to develop/build such cards? Specifications: PCI or MiniPCI up to 120 concurrent transcodings Codecs: G.729/G.729A or
2006 Oct 25
2
Choice of soundfile format
Hello What soundfile format, is the one that uses least transcoding during playback? As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct? Kind Regards Jon Leren Sch?pzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 -
2005 Jul 21
1
SOLVED: TE410P card in an HP-Compaq DL380 G4 server
Hi to all out there using HP DL380 G4 servers, i found a way to get the Digium TE410P with older firmware running on a HP-Compaq DL380 G4 Server! Here's the step-by-step description: 1. download the latest BIOS (in my case it was 4.04 from date: 06/02/2005) for the HP-Compaq DL380 G4 using the "Systems ROMPaq Firmware Upgrade Diskette for HP ProLiant DL480 G4 (P51) Servers" Link:
2004 Feb 03
1
Re: Asterisk-Users digest, Vol 1 #2711 - 15 msgs
you can do this with MeetMe, but you don't have to. you can also use Parking, which makes things a little simpler. in either case, the strategy is going to be something like: 1. Record the soundfile 2. Park the inbound caller 3. Use a .call file or the manager interface to initiate an outbound call to the mobile phone 4. play soundfile and prompt the mobile phone user to accept/reject the
2011 May 02
2
Retrieving/Streaming audio/video files from DB using over AGI
On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] <all.eforums at gmail.com> wrote: > Hello All, > > Probably a silly question, but we're wondering if people have had any > experience and have data to demonstrate if the performance of the Asterisk > system might suffer in terms of latency etc. if we're to have it retrieve > sound files from a database using odbc as
2005 Dec 28
3
voip-info: Asterisk record calls
On this page http://www.voip-info.org/wiki-Asterisk+record+calls there is "Example by Mojo". I have done everything he said and I have sox package installed. [root@pbx recordings]# sox -help sox: Version 12.17.7 ... When I open this web page http://10.0.0.26/recordings/index.php I get this: No Recordings Found And there are recordings in /var/spool/asterisk/monitor Do I have to do
2010 Sep 22
2
Unable to open vm-INBOXs
Hello list, it seems that a sound file is not present on my system, although I have made a standard install... [Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full: File vm-INBOXs does not exist in any format [Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable to open vm-INBOXs (format 0x8 (alaw)): No such file or directory I do not find this particular soundfile
2007 Apr 24
3
auto dial out multiple destinations
Hi, I am searching for the most effective solution for the following scenario: Our users can call into our IVR menu and dial a specific extension and immediately hang up. This event should simply trigger Asterisk to make multiple simultaneous calls through a group of zap channels (5-10 calls). When the called parties answer, Asterisk should simply play a message and hangup. So I was thinking