similar to: Waiting for dial tone in Dial cmd

Hi, After some testing I've found out that my client's hardware recognizes DTMF only if digits are sent 50ms apart with 50ms of tone duration. This was tested using a test device which generates DTMF. Now asterisk doesn't do it by default because digits going out from Asterisk are not being recognized. Using command sendDTMF, I can control inter-digit duration, and using

<DIV><FONT face=Arial size=2>hello All,</FONT></DIV> <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV> <DIV><FONT face=Arial size=2>our Asterisk pbx is sitting behind a normal analog hardware pbx, we have to dial 9</FONT></DIV> <DIV><FONT face=Arial size=2>to take an outside call through the hardware pbx, our

I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank

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:

Hello, everyone. I'm having a strange problem with my asterisk. After dialing and before the other side picks up the phone I should hear the tones (I'm not sure what are they called: piiiiiiiiiiii---piiiiiiiiiiiiii....) and in almost all cases that is true. However there is a range of numbers where I'm having this problem. There is no tones, just silence, until someone picks up the

Hello, When dialling a SIP phone which is already in a call the caller hears a "regular" ringing tone and does not know that the called party is engaged in another call. Is there a supported way inside SIP to tell the calling party to play a stuttered ringing tone? Thanks! __Yehavi:

I am wondering if it is possible for an extension that is served by a zaptel device to revert to dial tone once a call disconnects. For instance, if I make a call to another extension, talk with them, and THEY hang up, can I then be presented with a new dial tone rather than a congestion tone? Further, can an extension be set up so that, once the call goes back to dial tone, if the user does NOT

Oh geez nobody respond, I guess I have to be more descriptive. I'm install Asterisk and FXO FXS Lite dev kit from Digium everything has installed and I am getting an error message that my sound card is not enabled. I'm using a Dell Optix (whatever). Everything seems to be in order. I connect an analogue telephone to the FXO port? and try to dial. Dial tones work however not dial tone.

Greetings to the list: this is my problen when I make a call from my asterisk towards a nortel PBX , the call is made but in my telephone sip I do not listen the dial tone or the busy tone but the call it is completed normally. sip-phone-g729-------------asterisk--------h323-g729--------------nortel-pbx thi is may configuration: RedHat 8 2.4.18-14 Asterisk 1.0.7 The NuFone

Hi, Just noticed Asterisk is not playing 'ring' tone as defined in indications.conf when Dial command is used with 'r' option. For example: [test] exten => 123,1,PlayTones(ring) exten => 123,n,Wait(5) exten => 123,n,Playback(demo-congrats) exten => 123,n,Hangup() exten => 321,1,Dail(LOCAL/123 at test/n,60,r) When I now dial with a SIP phone - 123 I can hear nice

New system install. At what point, in bootup, should I be able to get a dial tone on the phone ports on a tdm400p? There are two fxo and two fxs ports. I know which to plug into <g>. At boot up, as soon as wctdm is loaded, all the ports "go green, yet I do not get a dial tone on the phone ports. I thought as long as zapata.conf is correct, the board should be

Dears, There is a PSTN line shared between 2 asterisk servers, (openvox 4FXO lines) The outgoing call of the one server may be conflict with the established call of the other one, is any way to force the Asterisk or Dahdi to dial after hearing the Dial tone ? -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL:

When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I have tried a few different things to get asterisk to place the call in an answered state and send the DTMF 1 with the Dial macro. I found Malcom Davenports wiki page regarding Google calling which has been very helpful in troubleshooting the issue.

Hey; What happens is that a script logs into the AMI and originates a call. When the call is answered, it jumps to a context in the dial plan. Thanks Much; John V. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Victor Villarreal Sent: Monday, May 15, 2017 12:08 PM To: Asterisk Users Mailing List - Non-Commercial

The way I have my dialplan configured, an internal extension is routed to a different context (with Goto()) on pretty much the first button press. 2 -> internal extensions 0 -> operator 5 -> VM 9 -> outside line etc. So a "201" will go to the internal extensions context, s,1, do some setup and then match on "01". The thing is that when the 9 is entered, I

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, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how i could do that - so that it looks really natural? It would be silly to create long-long-long

Hi, I'm looking fore a way to play a dial tone before our IVR platform answered the phone line. I want to use for the following reason: When a caller calls our Voice Platform, the call will direct dial out to a number. I want to dial out before the inbound call is answered. But now the inbound call here's nothing. When the outdial call is picked the inbound call will here

Hello, I have an Adit 600 (3 FXS cards) hooked up to a digium T1 card in my asterisk box. I 'connected' the slots to the a:1 T1 interfaces via the command line. The slots (3 fxs) are configured with 'ls' signaling. I configured the T1 card with the same line settings as the T1 interfaces on the adit and I get green lights on both the T1 card and the T1 interface on the adit (so

I've been searching the archives for the proper "Wait for Dial tone" command in the "Dial(Zap/g1/18005551212)" dial sting. Does anyone have an example of it's use? Arick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040906/6f8b21a6/attachment.htm