similar to: Strange DTMF issue with callback

The call quality is really pretty good. I think better than Vonage over an FXO bridge. If you are looking for a home provider with direct SIP support and local phone numbers this is a good choice. If anyone has questions or comments about my configuration please pass them along. I have noticed that if you don't put fromuser=phone# then the extension caller id passes through. Also the

We have PSTN lines connected to FXO lines of a TDM400B. I just got a complaint that overseas callers who are using cellphones sometimes find that DTMF digits aren't working - they press digits and the menu goes on as if they hadn't pressed anything. Since it sometimes works, and other IVRs work over the same cellphones, it's not that the cellphone isn't sending the digits.

Hi, We are running 1.4.22 and have been experiencing problems with certain IVRs and DTMF Tone duration. We would like to be able to increase DTMF Tone duration by 50 to 100ms over what the user is pressing on his phone. We have a PRI test circuit and an analyer in between to measure tone duration. We have tried setting chan_dahdi.conf parameter 'toneduration', but that does not do

Hi All, I'm at the end of my tether here and would really appreciate some help. I'm trying to implement DTMF based pause/resume of call recording. I'm using Asterisk 1.4.22.1. Here's the scenario: The caller (SIP or ISDN, doesn't matter) dials into the asterisk which executes the following code: exten => _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb)

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost

Hi all, I have a problem with DTMF and SIP. When I dial an extension with 2 of the same digits in a row (ie '445', two 4s in a row), i have to wait at least one second between the digits. If I don't wait and just continue pressing digits only the first digit gets detected and the second (and third if there is any) is skipped. Everything works fine when there are none of the same

I've this setup : CiscoAta186 -> asterisk with oh323 chan -> gsmgateway dtmf doesn't work, tryed inband, with g711a and g729 codecs CiscoAta186 -> gsmgateway works, even with g729, so it seems the problem is in * oh323.conf has inBandDTMF=yes, what else may I need to tweak ?

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

I have a Motorola Razr successfully connected to asterisk using a bluetooth dongle and chan_bluetooth. Here's some issues I've run across: - You have to ignore the instructions in bluetooth.conf, saying to run "sdptool search --bdaddr xx:xx:xx:xx:xx:xx 0x111F" to determine the correct channel to use for your phone. My phone reported Channel 7, but will not work with anything

Hello everyone, I have problem with DTMF recognition when calling from PSTN, my Asterisk box won't read DTMF tone at all. I've tried use cellphone, normal telephone and voip lines, nothing worked. softphone to softphone within extensions are ok. I'm a newbie at this, can anyone point me out where to look? I'd really appreciated. Thanks a lot Nate -------------- next part

I thought I sent this earlier this week, but I didn't see it. If I missed it, I apologize for the resend. We are running asterisk 1.2.2 with a TDM04B connected to PSTN lines. On incoming calls from cellphones located overseas, DTMF is not recognized - we have many single-digit choices in our menu so the problem isn't that some digits aren't working, it's not listening at

I have built latest Asterisk w/ OpenH323 channel driver. We have a SIP softphone registered to the Asterisk. We can place outbound calls from the SIP phone to the PSTN via OpenH323 connection to our gatekeeper. Everything works okay - DTMF and Audio... But in the reverse - if we call from a cellphone or landline the PSTN number we can get the SIP phone to ring - we answer and can hear the

Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different mobile network providers. Nothing. I played with the rx/txgain values from hearing nothing to too loud... I have no more ideas. Marc -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Steve Underwood Gesendet: Donnerstag, 4.

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

Hi again, I am trying to get my DTMF to use RFC 2833 rather then inband, so that I can utilize lower bandwidth codecs through my FXO. After much tinkering I was able to get my gateway (wellgate 3701A) configured to a point where I have some success, but no real joy. I have configured the RTP Payload type (or RFC2833 Payload type) to 101. I don't have a clue what this means, but I took

Hi all! I would like to solve some problems: I have a sip provider that lets me make pstn calls after listening some stuff and entering a pin number: 1) How can I make Asterisk enter the pin number? Then wait 1 second and enter the phone number? I have in extensions.conf: exten => 6*,1,Dial,SIP/2002@myprovider,60,tr I have tried with w (like with ZAP channels) but it does not work, nor

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

Thanks. I'm able to getting the asterisk calling back to my cellphone. But when I get to the authentication I get this message when I start to dial in my password: NOTICE[5178]: rtp.c:509 ast_rtp_read: Unknown RTP codec 96 received Is this a DTMF failure of some sort? Thanks again. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com

Hello All, I don't understand where is the problem... I have Callback setup and it works fine when tested within US. Works fine meaning the DTMF tones are passed when prompted to enter the phone number. But when I test with some international countries, callback works but DTMF tones are not passed... Is it: - a) Asterisk problem? b) Callback problem? c) VoIP provider problem?