similar to: Execute Macro AFTER connecting to a channel

this is the point, and the strange thing: DTMF is set to rfc2833, but is working both on incoming and outgoing calls, it is not working only on calls generated with the Originate AMI command, or with the queue member that point to Local dialplan, as you suggested 2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston at gmail.com>: > Looking at your logs it looks like you may need to

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

I'm running Asterisk 1.8 from Debian. I have some analog phones connected via a TDM400P. I'm testing them with these simple extensions: exten => 600,1,Answer() same => n,Festival(This is an echo test) same => n,Festival(Hang up or press pound when you are done) same => n,Echo() same => n,Festival(Good-bye) same => n,Hangup() exten

thanks John yeah, your approach is much siple, i've tried it but i'm not able do detect DTMF tones. it seems that on calls that i receive DTMF tones are handled correctly, but on calls generated from Asterisk to the world when the called side sends some DTMF digits they are not detected: -- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in new

Dear all i have a TE205P connected to an Asterisk 1.2.18. Yes i know, the version is old but since now the system was stable and i don't have the necessity of an upgrade. The system provide an IVR service that: 1) receive the call 2) verify the queue length 3) hangup if queue length is > 1 4) put the call in the queue othervise Then, there is an AGI php script that 1) verify the queue

Is there a way to dial DTMF after bridging the call. The current option D() in Dial will dial DTMF before the call is bridged and this doesn't do the job. I need to dial DTMF after the call is bridged and the message is played with "Background" -- #Joseph

Hello, I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1 FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5 dialplan. I have connected an analog phone to TDM FXS port, but when I pickup the phone to make a call, Asterisk "hangs up" the call. Let me explain: In another system, when I pickup the phone, Asterisk give me tone to dial: >---

Hello, I just setup the Asterisk to integrate with Panasonic legacy PBX. Config as followings, PSTN <-- PanasonicPBX--TDM400P(FXO)--AsteriskPC --> Internet * is for AA / Voicemail and VOIP in/out Currently the AA / Voicemail function for incoming PSTN calls are working well. My problem is for the incoming VOIP call. It can ring my internal extensions and talk without problem. But

Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

>From WIKI: D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. (You can also use 'w' to produce .5 second pauses.) When I use the D option to send a call to my paging system and pick a zone, the Tone is too early. I have tried the 'w' option, but it does not appear to work. No matter how many 'w's

When I use DISA I get congestion when I try to reach 1-800-number: Here is the context: [disa] exten => 087,1,Answer exten => 087,2,DigitTimeout,8 exten => 087,3,ResponseTimeout,20 exten => 087,4,Authenticate(985) exten => 087,5,DISA(951|disa-access) [disa-access] include => tollfree include => outgoing-voipjet [tollfree] ; ; terminate toll-free no.'s via fwdnet ; US

Good Day All, I am experiencing some weirdness using the E&M channel and hope you can offer a little assistance with the problem I am having. 1) call comes into channel 25 (Second Span first channel of a Digium Quad PRI from SBC-PRI) 2) Call is sent to channel 1 (First Span first channel on the Digium Quad PRI connecting an ADTRAN via E&M Feature Group D) 3) Between rings one and two

I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual Band Analoog FXO) working with Asterisk. I had a working FXO configuration to a analog port of a small home 1/4 ISDN pbx. I used this same configuration to connect a GSM Gateway that is supposed to be connected to the external(FXO) analog port of a pbx. I can get my configuration to dial the mobile number via the gateway, but

Hi, I am about to pull my hair out after trying to get our PRI up and working. We are switching from a Cisco gateway to an Asterisk box which provides the 23 phone lines for our office. So, because the Cisco gateway is working I can assume I have all the settings right (b8zs, esf, dms100, etc) and the PRI is live (because we are switching over). When dialing from PSTN, I get busy signal. When

I receive several calls from this scamer: Senior SafeAlert It is an automated call and they keep rotating their caller ID so it is harder to block them. Does asterisk have a "fax" sound tone? If I block their number and play "fax" tone/sound maybe they will remove me from their calling list. I've tried to call them but nothing helps. Any better ideas? They keep calling

I want Asterisk to do the following: - call a voicemail system by dialing a number and playing a DTMF tone - record what is said by the called party and save the recording to a file - end the recording when a particular phrase is said by the called party - put that recording into an Asterisk voicemail box and notify the user I've made a start below (on the easy bit). Any further pointers on

Upgrading an ancient customer installation... was running 1.4.23.1 (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running fine for 5+ years. Customer getting anxious about hardware failure, so we built a new box and installed 1.8.24.0, Dahdi 2.7.0.1, and a new Sangoma A104D. The single active span is an RBS T1 B8ZS/ESF/E&M Wink. I tried to move one span over one

YUP, this is the way that asterisk works. It is going to quelch all DTMF that goes out via a SIP gateway via asterisk. I spent a long time working this through and it has to do with the way that asterisk deals with DTMF and the DSP.c module that sits inband to the RTP/audio stream. There is a flag called DSP_DIGITMODE_NOQUELCH that is broken that might allow the inband DTMF after answer to work

I've got a piece of CPE equipment that has an FXS port that I have tied to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a standard telephone connected to it, I get a dialtone. If I dial a digit, and send a hookflash, the device will provide a dialtone back for the next available channel on the device. I'm trying to recreate this same behavior with Asterisk,