similar to: analog channels calling out not detect DTMF

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

I would like to hear from anyone good or bad as what their experience has been in recent times with STABILITY of current builds of Asterisk and drivers for TDM400P. The sort of configuration is: 6 incoming POTS lines. ie. 2 TDM400P cards. I am not concerned with: price points, or the advantages or disadvantages of using POTS vs ISDN technology, but simply RELIABILITY & stability of the

I have a 6 - 7912's. I have a TDM2402E echo cancel card. Asterisk 1.2.12.1 Some times extension to extension the audio is only heard one way. Some times extension to TDM2402 calls audio is only heard one way. I have turned off the echo suppression on the 7912's config. Any idea why it would be doing this? Something to try? THanks, Jerry

Hi I have a problem with dtmf recognition an analog lines connected to Sangoma A200. The digits (in most cases the first one) are doubled and so my IVR is useless. I tried to adjust the rxgain, toneduration and relxing the dtmf but nothing worked. I also noticed one thing it only happens during the background application: exten => s,1,Background(soundfile) exten => 111,1,Dial(SIP/111)

Hi all. I have a DTMF detection problem by my new analog card (ATCOM 2 FXO port). When i`m playing a voice with 'GET DATA' AGI command, sometimes asterisk do not receive DTMF from caller while the voice is playing. But if user waits to the end of playing voice, there is no problem. I`m using Asterisk 10.3.1, dahdi-2.6.1 on CentOS.6.4. Could you please help me? Here is my configs:

I've tried this before, with no luck. I've got to try again this evening, and I'm looking for some help. Here's my configuration -- pretty simple, really. Asterisk box -> T100P -> TA750(20FXS/4FXO) -> phones and outgoing lines I have an analog modem (Ok, it's a TIVO) that I need to be able to dial out. Right now, I have the modem connected directly to an outgoing

Is there any way to fill in the DID information for analog Zap channels? I've got a TDM422P and since I know the phone numbers associated with each of the 2 FXO channels I'd like to set that so that future extensions contexts can use it and the caller-id info in the form mydidnum/callerid like I can with VOIP DIDs. I haven't found if there is a variable that can be set with this info.

Hi I am trying to use Asterisk as a backend to send and receive faxes over analog channels connected to a Siemens HiPath 3550 switch and a TDM400P card. Receiving faxes, including multipage ones, works really fine, we have no issues at all. But when it comes to send faxes using the app_txfax application, spandsp can't send multipage TIFF files for some reason, it stops sending in the

I wanted to get everyone's opinion on an issue I am having. I am currently using linksys PAP2NA ATA adapters to terminate analog calls from my auto dialer to the voip termination co. The problem I have is when I call the PSTN everything goes fine until the person being called hangs up the phone. Once they hang up on the PSTN side it takes almost 15-20 Seconds for the ata to see the

Hi, I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send "qualify" queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Regards --------------

I own a TDM2400 board, with three FXO modules and one FXS. I'am having trouble with analog sip phones, from two different equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), sometimes when I am calling someone, then I press flash, and then call someone else, both calls stay connected after I hang up. [Sep 29 07:18:06] VERBOSE[3218] logger.c: -- Called g2/16 [Sep 29

Hello, I've commited the new busydetect routine to CVS. You need to cvs update asterisk of course and then choose it in asterisk/Makefile and recompile asterisk. All you X100P users that had the problems with false hangups or the card not being able to detect the busy tone please check that. In the asterisk/Makefile you need to find a line BUSYDETECT = and uncomment what you want/ comment

Hi, I am using zap channels, and by using sendDTFM application, I can control the duration between two DTMF digits, but I can't find a way to control the duration of the digits themself. Did search on the Internet and found out that I can change it in the asterisk source files and recompile asterisk. Wiki also says that it can be controlled using toneduration option in zapata.conf, but it

Hi, I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium TE410P card. Calling into meeting rooms that have been configured with the p option works fine. From ZAP extensions the # key does not work to exit, however from SIP extensions the # key works fine. This makes me believe that somehow the DTMF doesn't get through the ZAP interface. After furter experimenting

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

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

I have done some searching and not sure this is even possible, but here it goes... **Scenario** Let's say you have an asterisk server that you use to connect to a SIP provider that you push your PSTN-bound calls to using g711 and out-of-band DTMF. The SIP phones in use are Cisco 7960's and are set to also use out-of-band DTMF. For the most part, everything works great. However, a few

Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk

Dear Mr,Ms; I am planing for a custom IVR, for example to act as a simple installer! I mean there is some choice via 0-9 and # as *Next* and * as *Back* button. is there any way for me to detect if the caller pressed # vs * on Dialplan ? -- Regards, Ali R. Taleghani 0936 322 4069 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thank you Peter for you reply, I realize this problem occur because I take the CVS head (maybe a bugs get introduce), because when I rebuild using the checkout of the latest stable version (cvs checkout -r v1-0), I don't have the problem of detection of DTMF over a direct ISDN pri link. Hopefully this problem with be fixed before the next release of asterisk. Sylvain. -----Original