similar to: Two DTMF tones on keypress with Handsfree cell

Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is

While most cell phones are fine, some cell phones don't seem to produce DTMF digits that Asterisk/Zapata will detect. One of our salespeople has an AT&T model that never gets any digits through. Is there a known solution? I understand, of course, that with cell phones the DTMF tones are actually created at the base, and that pressing the key longer usually does not result in longer

Hey People, Sorry if this has been covered before but for the life of me I couldn't find the answer. So if someone could just point me in the right direction that would be greatly appreciated. Is it possible to set up in the dial plan to assign keypad numbers a sound file on a bridge call. I know that you can do it if the user just rings into the dial plan but was wondering if anyone

When my Asterisk box is called from my company's standard type of phone, it does not detect normal button presses on the keypad. You can use the phone's 'touch tones' feature to send data to the box, but that is a bit cumbersome. My Norstar voicemail system (that is being replaced) does not have this limitation. Is there anything I can do on the Asterisk side? Thanks, Bob

Since upgrading to 7.2, I've noticed a random problem where I dial a number and hear all the correct tones in the handset, but the display won't show all the numbers I dialed. So you sit there waiting for the dialplan to kick the call off (b/c you heard the proper amount of tones played and think it's all good) but the phone is just sitting there b/c it somehow "missed"

I encountered something strange. A local business has an ACD that, when I call it using a Polycom 550 connected through an Asterisk system, will respond to button presses only if they are short. Calling this business with our old (non-Asterisk) phone system or with my cell phone works because they only generates short tones when pressing the keypad during a call. The Polycom, however, generates

Just to add.... P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] Sending :160 bytes 2 MISDN P[ 0] misdn_jb_fill: written:160 | Buffer status:256 p:861fee0 P[ 0] misdn_jb_empty: read:128 | Buffer status:128 p:861fee0 P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] PH_CONTROL: channel:1 oad2:07nnnnnnnnn dad0:820055 P[ 1] --> DTMF

If you're an asterisk 1.6 user, and use the 'Directory application', have you noticed that the first keypress is always missed if you press it during the part of the announement where alison says "using you touch tone keypad" If this includes you, have a look at mantis bug https://issues.asterisk.org/view.php?id=16409 If you are keen, please apply the patch and report

I have been looking forward for months to get chan_mobile working. I am limited to using prepackaged Asterisk code, mostly Trixbox. I have recently heard that chan_mobile is considered 'beta' and there is no effort to move it into the main code of Asterisk. Not even for Asterisk 1.6. So what is the future for chan_mobile? It is easy to make a very strong case for a cellular trunk.

Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and

Hello, Are there any gateways which allow me to hook a cellphone to Asterisk and use that line for routing my calls? Basically, I'm looking to play around a bit and if I can get to connect a cellphone with Asterisk then that would be great. Thanks, Hitesh PS: I have tried to search on the web, but didn't find any pointers on how to do so. -------------- next part -------------- An HTML

Hi all, I managed to get most functions of Asterisk operational with Messenger. Accessing menu's have I done with DTMF generated tones by Windows Media Player, because this seems to be the only program without sound-card conflicts with Messenger. The tones are mixed with microphone-input. Does any of you know a nice dialer with a graphical keypad that does not grab the sound-card exclusively.

Dear friends, As I commented some while ago in the list, occasionally when DTMF Tones are sent, they appear in RTP Payload and in Events too, producing duplicate tones being recognized. This behavior happens in Asterisk as well as in Gateways such as Cisco, for which we had the opportunity to observe the error and extensively debug it. We ended up recognizing good digits by adjusting audio gain

Everyone, I have a problem with the numeric keypad while using a Centos 7.5 new installation with kvm. When I login to the host machine using gnome3 or classic gnome the numeric keypad does not function. If I activate another 7.5 guest machine with gnome3 the numeric keypad on the guest does not work either. If I login on the host machine using kde as the desktop the keypad works fine and

Hi, My system is like this currently: ATA-186 <-> *1 <-> IAX2 to Europe <-> *2 <-> i4l <-> voicemail at cell provider When I dial up to my voicemail at my European cell phone provider I can't press '#' to get into their menu. It seems like it just ignores any DTMF tones or doesn't get them. When I call a human on the other side of the i4l they

Just setting up Asterisk. I'd like to be able to dial out through VOIP providers and have customers type in a code in response to a prompt. So far, I've been able to set things up to make the call and play the prompt. However, my problem now is the DTMF tones; they don't register when I call get_data. When I make a call in person (from the DIAX softphone, through a VOIP provider,

Greetings, I am PRIs connected to a Cisco 36xx gateway, which in turn connects to Asterisk via SIP. The problem I am having is that DTMF tones originated on the PSTN side are not heard on the SIP device. On the other hand, tones originating on the PSTN side are received by Asterisk when talking to voicemail or an autoattendant. >From the Cisco debug, I can see the Cisco sending NTE (RFC2833)

I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The

Hi, What is out there in terms of SIP enabled handsfree speakerphones? Looking for something that works well and also fits a low budget. I am used to using a Cisco 7940. It is a great phone but a bit expensive. Thought about the Polycom SoundPoint 300 until I realized that it does not include speakerphone functionality. Thanks, Adi

Guys. I have a te100p with unicall and an E1 and Im having problem with DTMF tones but the weird thing is, I only have problems sending the tones to certain phone numbers, anybody seen this behavior? Asterisk shows on the console the dtmf tone been pressed but seems the other side is not getting them, and this just happens with certain phone numbers, not all.. Any clues(tips?