similar to: double dtmf in incoming SIP call?

We're still having problems with DTMF detection on our X100P cards. Incoming callers that hold down the "1" button for too long are being connected to extension 11. One would think fat fingers were uncommon, but it happens to alot of people. I suspected this was related to our having to increase the txgain, but I tried turning it down with no effect. I also tried disabling the

hi, i have a pronlem connecting call from pstn with the following confuguration, please advice extensions.conf [from-trunk] include => from-pstn [from-pstn] include => from-pstn-custom include => ext-did include => from-pstn-timecheck exten => fax,1,Goto(ext-fax,in_fax,1) extensions_custom.conf [from-pstn-custom] exten => s,1,Answer exten =>

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

Hi there; I didn't see any "G" option in the example above, and the usage for the option parameters is entirely undocumented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial The G options are as below G - If the call is answered, transfer the calling party to the specified priority and the called party to the specified priority plus one. context exten

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)

Hello listers, I have a couple of 1.4.37 through 1.4.42 boxes running at different sites. These systems run a fairly simple IVR that uses waitexten and Read to get credentials and plow on through a set of contexts. I am experiencing two problems in my setup: 1. In environments where DAHDI is the trunk of choice, this snippet drops digits, especially if the user

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

Hello, I have a problem of DTMF duplication. I receive call from my provider with SIP protocol. These calls pass through an interactive voice menu, using the application Waitexten to enter a client code. The menu works fine, but sometimes I have DTMF duplication that prevent proper code entry. All DTMF come twice. my sip.conf ----------- [general] context=default allowguest=no

Hello list, I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom or Asterisk that makes the trouble. I'm playing a prompt, then make a choice for "2" : [Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] -- <SIP/test1-00000701> Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (language 'nl') [Jan

I am revising our DialPlan strategy for our Asterisk system (1.4.2) and looking for some info on 'best practices' for this. Here's what I'm trying to do: I have an ACD menu that gives the caller the options as follows: - Press 1 for sales - Press 2 for support - Press 3 for customer service - Press 8 for a 'Dial by Name' list or enter the extension number at anytime

-------------------------------------------------------------------------------------------- Originally posted at http://forums.digium.com/viewtopic.php?t=18045 -------------------------------------------------------------------------------------------- Hi! I'm trying to configure a DISA setup (Asterisk 1.4.11). Only, executing DISA seems to prevent any DTMF detection capability when using

Hi, I have a Digium TE410p T1 card and I've noticed that under asterisk 1.4.17/18 I have problems detecting DTMF in IVRs. I think I've narrowed the problem down to some sort of interference between the greeting that is playing and the DTMF tones. DTMF detection seems to work very reliably when I am in Read() or WaitExten(), but is absolutely unusable while in Background(). I hope someone

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

Still grasping at straws trying to solve DTMF detection issues with one of my asterisk servers. This particular server is now running Asterisk 1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console only) with 2 X100P cards. I have tried adjusting channel gains, turning call progress and relaxdtmf on and off, switching echo cancelers, just about everything that Google turns up and I can't

I'm in the process of setting up a 'phone tree', and are running into some problems. My goal is for users to dial a phone number, the asterisk system picks it up, plays the greeting, and users can type whatever they want into the system. What actually happens is users dial the phone number, asterisk picks up and additionally goes off-hook on another line, plays the greeting and

Every so often, and it seems that it happens only when a call is in progress, all 24 Zap channels get reset. All channels are opened and then timeout. This causes the in-progress calls to terminate. There are no corresponding Red/Yellow alarms on wither the PBX or Asterisk although we do receive a fair amount of Blue Alarms. The Asterisk server is connected to a legacy PBX through a Digium

Dear all, Can someone help me about DTMF in ISDNPRI. I'm seting up E1 PRI found problem 90% Hangup when put digit. my E1 card work well with other telco. for this telco it's first time to use (telco in Thailand but i don;t know about equipment they use) I try to debug and got message below. Please let's me know if you can help Regards, Dome C.

Hi, I try to set up planet VIP-050 with asterisk. Everything works fine instead of the call transfer. When I press # console says something like this: >Oct 5 11:11:20 DEBUG[25104]: chan_sip.c:2222 sip_rtp_read: Oooh, format changed >to 1024 >Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc >frame that isn't a multipleof 50 bytes long from RTP

Good Morning, I've recently gotten Asterisk installed and configured our IVR using FreePBX. Things seem to be going well except a few of our inbound callers are ending up in the wrong place when trying to connect to a specific extension. The example I had this morning was someone trying to call extension 212 and getting connected to the Sales queue which is option 2 on the IVR. I looked in