similar to: Tone detection while Dialing

Hi everybody, I don't know how to do this, I redirect a call and then dial someone, but I dont want the ringing tone to be listened while the dialing part is waiting for the dialed part to pick up the phone. exten => 555,2,Dial(${STRING4},30) I have tried when the option 'm' , but I don?t want the default music on hold to be listened neither. I want nothing (silence) to be heard

Is the a way to turn the ring tone OFF during dialing? When I'm in a macro mode I have to listen to ring the tone for 20sec before macro finish and I get connected. -- Joseph

Hi, I use VoIPTalk as my provider and unsure of a minor issue. When people call me they get a US ring tone instead of UK. Is this a Asterisk configuration issue or one for VoIPTalk ? I am running 1.6.2.0 and IAX2 trunks. Thanks - Phil

Hi, I'm facing the following request : "When someone is starting an assisted transfer using Asterisk's features codes, he will ear a prompt saying "Transfer" and then a dialing inviting him to dial the number he tries to reach. This tone volume is qualified as a bit too load." Is it possible to change that and have a more delicate volume ? A quick look inside

Hi all Can iptables have log and deny rule together? if no. how can I make a deny rule and log rule and the log rule can limit the log entry eg: 200 if yes, how can I make it I am using freebsd ipfw. eg: ipfw add 22 deny log all from any to x.x.x.x thank you ____________________________________________________________________________________ Take the Internet to Go: Yahoo!Go puts the

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

Hello ! I 'm using a TE405P with a HW echocanceller module attached on it. dahdi version is dahdi-linux-complete-2.2.0.2+2.2.0. As far as I know, the fax tone detection is done on the FW board. How can I verify that the echo canceller has been turned off ? When I do a cat /proc/dahdi/1 for span 1 I see still the VPM450 entry near the channel although this was a fax call with CED tone.

Hi List; What is the command and where I can write it to block specific code from calls (then no one will be able to place call for any number start by that code)? --------------- Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: + (965) 9849460 Yahoo ID: bilmar_gh at yahoo.com MSN ID: bghayad at hotmail.com

Hi all, I have a Diva Server V-BRI-2 card, which support, as written in reference guide: Extended tone processing (human talker detection, generation and detection of country-specific tones) I would like to detect human speech and fax tone with asterisk. I think that the diva card transmit a DTMF code when detecting voice, but chan_capi doesn't receive this DTMF code. I verbose it

Hello everybody, I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip installation. I actually have a problem when a PSTN user calls and hangs up. The disconnect tone is not detected by the SPA, the the call continues and, for example, leaves an empty message on the voicemail before hanging up (because * hangs up). How could resolve this problem?. I set, Detect

This is a resend - customer complains that some phone systems they call don't respond to key tones, please advise. Thanks Rob. ----- Original Message ----- From: Softprofit Solutions To: asterisk-users@lists.digium.com Sent: Monday, November 24, 2003 7:34 PM Subject: tone detection problem Some commercial systems don't hear the tones dialed from an analog handset, same

Morning, we have gateways with FXO port registered as SIP endpoint in Asterisk. To be able to use this port, the gateway ask for prefix -lets say 9- then send dial tone and here the user enter the calling number. We want to cancel this step for the users so they can enter the entire number and Asterisk will deal with the gateway. Does Asterisk have a possibility to manage this? We tried with

Hi, I have a problem with TDM2400P card. The card is detected ok, I can make a call but only with pulse dialing (not tone dialing) without hear sounds from the headset. When I receive a call, I can to establish a communication, but without hear sounds from the headset. When I dial any phone key, I can hear dtmf tone. I'm using Elastix 1.5.2. These are my configuration files:

Is it possible to detect SIT tones on an outbound call? Specifically this is for an outbound call generated via a .call file over a PRI. I get answer supervision when the SIT tone starts and Asterisk believes this is a successful call. I'm using Asterisk 1.4.6. Thank you! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I have the X100P card which as to sockets (LINE - for fxo line ) and PHONE (to connect a n analog line) This card is setup as fxs_ks I was getting dial tone but suddenly no Dial Tone....Help appreciated.... When I try to route the Call using - Dial Zap/1) to this FXO Line I get this error: ------------------------------------- linux*CLI> -- Executing

Hi , Thanks danny nicholas. Finally we get the things done with following. If i specify busypatten=500,500 then asterisk does not recognize hang up signal. After removing it only all are working fine. I choosed 2nd option as per your suggestions. working chan-dahdi.conf: ==================== signalling = fxs_ks busycount = 3 busydetect = yes callprogress = yes progzone=in usecallerid=yes

Though asterisk support AMD which is based on silence detection but I did not found support of tone / beep detection in asterisk to record a voice message for answering machines after detecting tone Will appreciate any help in this regard Best Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT Unified Communication Telemarketing

Hi, I'm trying to dial out on a vonage line connected to a zap channel using stuff like: exten => 200,1,Dial(Zap/2/${EXTEN}) but it doesn't work - when i dial in the extension, i can see on a phone connected to the same line that it's gone active - but there's no dialtone. also tried adding a wait period before accessing the line and exten =>

Dear all, I connected a normal phone line to the FXO port but the call is not being processed. The following is the output to asterisk console when I dial 9150 "9 is the prefix I configured and 150 is a local service in to know the current time" *CLI> -- Executing Dial("SIP/4444-d365", "Zap/1/150") in new stack -- Called 1/150 -- Zap/1-1 answered

Hi, I also have the same problem, from memory is this not progress indicator=8 that deals with the dialing indicator ? Anyway I am also stuck with not having any dial indication, has anybody got an idea. MO <mailto:asterisk-users@lists.digium.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: