similar to: play message to callee before connecttoincomingcall

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

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,

sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. the behaviour is just like MoH, but the problem is, that the caller has to hear a soundfile from the

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

Hi, I'm testing a Polycom IP600. With firmware version 1.1 the phone reboots at any time. With firmware version 1.2, the first reboot was an endless reboot. Then I moved the phone to another lan port, then it worked fine. Then I installed again in the initial lan port and the phone works well. However after some time of inactivity (1 hour?), the IP600 stops to send and receive calls.

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:

Hi... I just had a real quick and simple question... I have a asterisk implementation setup w/ real time off of a mySQL database for SIP peers and queues, voicemail, agents etc... I after the upgrade to asterisk 1.4.3 there are some new configuration features i would like to use. I was wondering if i could just add to the database table a column for the new config option? if this will work or

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000

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:

Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using

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

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

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

I've got the call cuing all setup and working, but im trying to get the Callswaiting,you are caller #, etc, and its not working. I have the following inthere as stated: queue-youarenext = "queue-youarenext" ("You are now first in line.") queue-thereare = "queue-thereare" ("There are") queue-callswaiting = "queue-callswaiting" ("calls

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

I think I solved the errors I was getting with my patch, sort of anyway. Brief over view: Tell all the callers their position in the queue. When they move, tell them their new position. I was receiving Thread xxx already blocked by xxx. I found that if I only tell caller 4 and above (Which becomes caller 3) that their position changed, I do not receive the errors.

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:

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:

I'm trying to play a wav-file on a channel. This is what I see in the asterisk debug console AGI Rx << STREAM FILE "test.wav" "12345" [Aug 20 16:10:19] WARNING[25219]: file.c:602 ast_openstream_full: File test.wav does not exist in any format So it doesn't find the file, or it's in a wrong format? I can listen to it with windows media player... it's a

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 -