similar to: Re: Send DTMF tone Like 'C' on connected call

Dear to all someone know how is possible to have a DTMF tone like "C" AKA Alpha Tone (connect tone) to the caller? Thanks in advance Dimitri

Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context

I am trying to run the make command to install Asterisk, but I get the following error: make ... ... checking for tgetent in -ltermcap ... no checking for tgetent in -ltinfo ... no checking for tgetent in -lcurses ... no checking for tgetent in -lncurses ... no configure: error: termcap support not found I am running Mandrake 9.1 on a Pentium II 200MHz. Could this be a hardware issue? I

Just wondering if there's any significance to the jacks on the TDM cards. They appear to be RJ-45 instead of RJ-11, and I wasn't quite sure if that's something that makes a difference from the user perspective. . . Thx. B.

Im trying to inter-connect my current PBX system and Asterisk. Asterisk has some users from different networks (internet).. I used cisco router using 4 fxs to pbx and SIP to asterisk. Is there any way i can allow the ip address of cisco to connect to my asterisk using SIP? IP Address of cisco is 192.168.0.254 here's a part of my sip.conf [general] port = 5060 bindaddr = 0.0.0.0

Fry's Electronics has a new Linksys 2 line ATA box for sale for $59.99 retail. They have a version with a router for $89.99. We picked the non-router version up and it seems to be a rebadged Sipura SPA-2000. The box has a Vonage service package inside as well, but it does work with other services. The box also has a "User Guide" meant for end-users that is very well written [no

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

OK here is a possible tricky one. I have a rocom door entry system which connects to an FXS port on my TDM400P card. When the door button is pressed it initiates an 's' extension which dials a number of SIP extensions. When a SIP phone is picked up the user can speak to the person at the door and press the 7 digit which sends at DTMF tone to the rocom unit opening the door. All this

Hi, in asterisk 1.6.2.16 I get spurious DTMF recognition over SIP from an Audiocodes. I think the DTMF recognition is the Audiocdes' fault, the Audiocodes log seems to say so as well, but I want to make sure, and fixing the Audiocodes is not an option in this particular case - don't ask. Can someone explain to me what the following means *exactly* [Feb 10 21:15:40] DTMF[2538] channel.c:

Hello Does someones know a good low-level way to debug the DTMF that is arriving (or not arriving) at Asterisk? We've already used the DTMF logger level. The picture is the following: We are developing an application that calls to a customer to get authorization information through DTMF. We've used at development environment SIP/IAX channels and everything was great and the work was

Nick Bachmann <nabachmann@yahoo.com> wrote: > I am trying to run YDKJv3 on Wine. I have successfully installed it, > and I am able to run it with Wine HQ and Codeweavers Wine. The problems > with both of these is that I can't use they keyboard! Pressing keys has > no effect. > > I have searched wine lists and bugs, and it looks like I'm not the only > one

Antonio> Agreed? Is Mailmate an option for you? Never heard of it, will have to look into it. I'd be willing to change the iPhone mail reader for sure, the Apple one kinda sucks overall, especially for sorting and such. IMHO. >> On Aug 10, 2020, at 7:51 PM, John Stoffel <john at stoffel.org> wrote: >> >> Antonio> Thanks a million for your insight. I

Agreed? Is Mailmate an option for you? > On Aug 10, 2020, at 7:51 PM, John Stoffel <john at stoffel.org> wrote: > > > Antonio> Thanks a million for your insight. I believe this is the thread to which you refer: > > Antonio> https://dovecot.org/pipermail/dovecot/2019-October/117219.html > > Yes, that's the thread. It looks like it's an Apple

Antonio> Thanks a million for your insight. I believe this is the thread to which you refer: Antonio> https://dovecot.org/pipermail/dovecot/2019-October/117219.html Yes, that's the thread. It looks like it's an Apple Mail client bug/feature/screwup. I hate it, since it messes up my main text based mail reader doing IMAP. Maybe I need to move to Mutt finally. Ugh...

Hi, I am attempting to create a intercom buzzer system using asterisk as a back end. Most is figured out except the actual action of buzzing the door. I need to detect whether a DTMF key was pressed by the the called party (the resident). Is this possible to do using just a dialplan? I can't see any options on the Dial command that would lead to this, am I looking in the wrong place? I looked

Please point me in the right direction as to where to increase the length of time I am sending DTMF tones on my T100P. I can't remember where/how I set it last time, and I need to do a fresh install. I'd sure appreciate it :) Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977

Is there a way to get my X100P card to dial "slower" on the line? Mine seems to dial the digits too short/fast for the switch to catch all of the digits and roughly 25% of outbound calls fail to complete. I've monitored on the line with a test set and the audio sounds clean, and I hear the X100P send out digits really fast, but sometimes the switch seems to time out and issue

Hello all, I am using voicepulse DID's to receive calls via IAX to and asterisk IVR dial plan I have put together. The problem is after 3-5mins the system cant pickup the DTMF tones I am sending... I have tried different telephones... It just repeats menu options over and over.... I have to call back and then it works again for another few mins... Any ideas... iax.conf? issue? Thanks, J.C.

Good day, I am hoping that someone can assist me with a work around. I have an IVR system that I am attempting to connect asterisk to, however the IVR was written Some time ago, and requires tones to be approximately 2 seconds in length. I am using SIP Phones, Is there any way to adjust the Length of the Tones that asterisk sends out the audio path? Are there any phones that would allow

Every once in a while calls get interrupted with a brief silence followed by a DTMF tone on our T1 lines. We are using a TE411P card. It does not happen on SIP connections, just the T1. Has anyone experienced this or know a possible cause? Thanks, -Bill