similar to: DTMF heard at end of AGI Record File

Hi there I have two different asterisk systems (both 1.4) whose dtmf tones are not being picked up by a particular conference system users are dialling into. I can call myself with the phones and hear the tones, but I am guessing perhaps they are too short or somehow different. I have looked and looked but can't nail down the reason. I don't believe this is a general issue, rather some

Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a couple of Linksys PAP2T-NA & Grandstream HT-502 extensions that are receiving random DTMF tones on their side, but that are not heard by the outside party. I have been using Asterisk 1.6.6 through 1.6.10 and have always had this issue. I am only using SIP on the Asterisk server and all extensions and trunks are set to rfc2833; outside of this issue DTMF operation works fine.

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over SIP trunk from which calls get routed to third server (C) (1.6.0.9) via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP

I just loaded from CVS this afternoon and in the debug output I see... DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: m on Zap/16-1 DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: u on Zap/16-1 I knew about DTMF 0-9, A-D, *, and #, but I didn't know about m and u :-).

Semi-offtopic, Anyone know of voip providers who can provide tollfree number service? E.g. route 800-xxx numbers to our * ? Even better if they are familiar with * or can speak IAX ... -Dan

Hi. I've spent the past few hours trying to get SSL working right in Dovecot 1.0.5 and now I must turn to you for help. I purchased an SSL certificate from Go Daddy. I pointed ssl_cert_file to the .crt file and ssl_key_file to the .key file, but the client (Mail.app) complains: Mail was unable to verify the identity of this server, which has a certificate issued to

It would be prudent the test for success and continue rather than failure and drop. For example: exten => s,5,GotoIf($["${SYSTEMSTATUS}" != "SUCCESS"]?105:6) That way only the result that you know is good, Will continue a call.. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On

I'm having a weird problem with IAX2 in today's CVS HEAD. I have two boxes with T100P cards connected via IAX2. Calls between them work fine, but when I press a key at one end, it comes out the other end as a click, with no tone. I've tested the DTMF on the T1 using SendDTMF with an outgoing call, and it sounds fine in that case; it seems to be only IAX2 that has the problem.

Hi all, Tiny, but very important question for me: what it can be when standard agi-test.agi script, like: print STDERR "Testing 'waitdtmf'..."; print "WAIT FOR DIGIT 10000\n"; my $result = <STDIN>; &checkresult($result); on the call from KPhone application does NOT return any DTMF code back (I use dtmfmode=inband in sip.conf): Testing

Does anyone how you might detect a period of x milliseconds of silence using AGI ? Rgds, Stuart

Hi all. I've been trying to build some new systems, and no matter what I do, if I load the zaptel and tor2 drivers, the system panics within an hour, even with no traffic. These systems are using dual Athlon MP 2800 chips with one, two, or three T400P boards and 2 GB of system memory. I'm currently using Fedora Core 1, but I also went back to our old reliable Red Hat 7.3 and the systems

Does * support ISDN Primary Rate Interface (PRI) - 2B Transfer Capability for T-1/PRI? In other words the ability to take a call and join it to another call and then drop off letting the CO-switch take over. -Kevin Kevin Fjelsted, President AltiCom CTI, Inc. Track Me Down! One number Access, Press 11# during the voice mail message greeting to have me F-O-U-N-D! Phone: 612.259.0722 Fax:

Good day All. Is there a way to pass DTMF signals to AGI script during conversation ? Actually here what I want to make: Users are usually dial using dialplan and when someone press *4 (during conversation) I want to have agi script to deal with that, but those users should keep talking and even didn't notice that one of them press something. Is there a way to do that or it's complete

Hello, I am working on a project where I need an AGI script to read DTMF digits and send MF digits in the same instance. I am attempting to do this with a standard zaptel interface. I probably missed something, but I've been looking through the source and can't find anything specifying how to do this. Any help would be greatly appreciated, Mike

I'm using get_data in an AGI script and am having a problem when, after a long time in my IVR, when I ask for a 10-digit phone number, the first few tries are always invalid -- the number it reads back is very strange, almost like the DTMF tones from other answers were being cached and then dumped on the call to get_data. Anyone ever experienced this before? I have to do some major

HI, I am trying to write an AGI that will authenticate users befoer the call is allowed to proceed. Befor you ask, i have tried using the authenticate() function but it does not work for me as this function messes up call accounting (authenticat() awayas awnseres the channel, thus causes CDR to bill for 'ring' time). the AGI will (1) playback a voice prompt over an unawnsered channel (2)

Hello, I am searching for a DTMF issue on my setup ( 2 years and counting ), and I am wondering why rtp.c has code to mute DTMF ( the rtp->dtmfmute variable ), but this same mechanism does not exist in dahdi. I am sending a DTMF over SIP w/ RTP & RFC2833 to the asterisk box with the dahdi card. The dahdi card sends it out on the PRI line. Trouble is, the DTMF is echoed back and the

I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for '192.168.200.99' - Username/auth name

I am having trouble with a DTMF-based application on Asterisk 1.0.3. Specifically, when two IAX2-based devices are talking, when they send DTMF to eachother, the other side only hears clicks, and maybe a millisecond of DTMF tone, but not any real duration. Furthermore, when one IAXy device calls the Echo test program, we can hear our echo, but when we punch DTMF in, we get the same effect