similar to: Dial D option not working?

I've got a new setup (different building) where Asterisk is sitting between the PBX and phone company on a E&M T1 line. Mitel PBX <-> Asterisk <-> Phone company Inbound works. Asterisk gets the in-band digits from the phone company and hands the call off to the Mitel just fine. Outbound is weird. Asterisk seems to expect that the mitel will send routing information

When using the dial command and the D option to send DTMF digits when the channel is answered, is there a way to allow for some dead air, and then send more DTMF digits? I would like to automate a call, and it requires entry of a few short dtmf digits all a couple seconds apart from each other. Thanks!

So I reached the point where my PRI is accepting incoming calls, but I cannot dialout. I must be doing something stupid, but I can't figure it out. The Asterisk box is sitting between the Mitel and the phone company, and has PRI lines to each. Asterisk was built from CVS r1-0 Log for a call from mitel heading outbound: ------------------------- -- Accepting call from '' to

Dear All, The documentation of the Dial Command, says the following about Option D: D([called][:calling]) - Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. However, in my experience, the timing the call get bridged is not consistance, sometimes even before sending the DTMF strings. Anyone share this experience? How to make sure

List, Having a little trouble with the following. Let me prefix by saying I have blind transfers working from the following setup. Inbound call [from-zap] (SIP/sv0071iv) answers. Zaptel -> Asterisk -> SIP extension SIP extension then blind transfers [from-sip] --- SIP extension -> Asterisk -> Zaptel During this whole process, the original channel off the trunk (lineside T1) is

Anyone have a suggestion on where I can get a decent new MB with 5v capable PCI slots. It seems like every decent server MB on the market has 3.3V slots only. Is diving into the junkbin my only choice if I can't afford to replace the 5v quad-T1 wildcard? Thanks Mark Farver

I'm hoping people can suggest some ideas for debugging a problem that I'm having with DTMF. Unlike most of the DTMF problems reported here, it has nothing to do with Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones on outbound calls on a PRI connected to a TE412P card. I'm running * 1.4.10.1 with Zaptel 1.4.4. It is important to note that these problems

Greetings. I am attempting to configure a system based on Asterisk 1.2.3 to be used as a backup should our aging voice mail/auto attendant system fail, which seems increasingly likely given its advanced years. The first part of this task is getting the auto attendant feature to work correctly, which I would have figured to be relatively easy. I have successfully built a menu structure, but cannot

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

>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

for the particular configuration of software/hardware that connects to my asterisk pstn gateway I need to do something like the following : [...] exten => _X,3,Dial(CAPI/02xxx.b${EXTEN},60,M(senddtmf)) [...] [macro-senddtmf] exten => s,1,SendDTMF(*) but the DTMF must be sended to the caller channel, and not the called : SIP -> * -> ISDN SIP calls some ISDN number, when ISDN picks

Greetings All! The Asterisk Call Manager works great. But I have one question for anyone who has used it. I cannot get the system to send some DTMF tones down the channel once the call has been made. Below is the script I am using to make the call, and start recording the channel. I am starting to make a system the will use asterisk to become an automatic random quality monitoring system

I have been trying to figure a way to SendDTMF into a MeetMe room using the Manager API. I can't redirect everyone into another context and then bring them back because that would mess up my logic. I am trying to use local channels and the originate Action to accomplish this. Exten: 3441115 Priority: 1 ActionID: actid-00000001 Context: senddtmftones Action: Originate Channel:

Hello, I did this post a long time ago but never solved the problem, so i'm trying again after something like 10 months, hopefully i'll find someone that found a solution ;-) When i call an external number that sends audio before call has been answered (like some PBX of public offices do here in italy), strange things happen: I'm using chan_capi, with Early B3 active, i can listen

I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have

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

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

I have an X100P connected to an extension of a Panasonic PBX. When a call from the PSTN comes in, it is routed directly to the extension where the x100p is . I want * to answer the call, play a message and then transfer the call to another extension via the Zap channel where the call was received (I need to flash the zap channel) . If this extension doesn't answer I want then to dial an IAX

I am trying to send a DTMF digit automatically every 15 seconds to keep a call connected to an alarm panel. I tried using the dial command L and recording a dtmf tone for the beep, but obviously that didn't work. Does anyone have a suggestion for merging the L option and the sendDTMF or the D option? Any other suggestions would be appreciated! Thanks! Paul Gentilini

Problem: Flash on X100P does not flash. Phone line has Call Transfer. With this line plugged into a regular phone, it can receive a phone call. Then, depress the hook momentarily, release. Dialtone is now available. Dial a different number. Call is answered. Hook Flash again, now in a three way call. Hang up. The other two parties are still in communication. Now, plug same line into the X100P.