similar to: DTMF not working on overseas cellphone calls

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

Has anyone had any experience with applying Speex on cellphone audio? If so, I have a few questions: * How has Speex performed with the audio, e.g., amount of loss/intelligibility? * Since cellphones use the typical POTS sampling of 8Kbps, would it help to use a multiple (16,32Kbps) or fractional (12,18,24Kbps) resampling? * At what sampling rate do you think a resampling of 8Kbps would result in

Hello I don't have a landine and use a VOSP to provide access to the telephone network. In case a call comes in and I'm not home, I'd like Asterisk to log the call, and then send an SIP message to my VOSP so the call is forwarded to my cellphone and is thus charged to the caller, without Asterisk having to dial out to my cellphone through my VOSP at my expense and bridge the two

Hello Even the cheapest GSM gateway I found for a simple SOHO server costs over $300. If possible, I'd rather have a local solution than use a remote VoIP provider to have Asterisk make outgoing calls to the GSM network. So... I was wondering if there are some entry-level cellphones that can somehow be hooked up to a PC running Asterisk and used as an el-cheapo GSM gateway? Thank you.

Hi, We ran into a little problem recently with our phone provider (Telus Canada): we are unable to dial numbers outside North America. This is what happens: the phone number 011... is sent out over our T1, Telus sees the correct number on their switch. However the switch thinks it's a North American phone number (and thus has to have 10 digits) and rejects the overseas number as

Hello I've never used Asterisk for a three-person call, and would like to check that MeetMe is the way to do this. The ADSL modem provided by my ISP offers free calls to landlines/cellphones when using a handset connected to an RJ11 port on the modem. A three-person call can be set up by using the standard PBX sequence: 1. Using the handset, call party #1 2. Hit "R" key on

Ladies and gentlemen, good morning! With my first post to the list, may I begin by saying how impressed I have been by Dovecot's performance on our Debian server. We've configured it to run with Exim and Fetchmail, using Dovecot to manage professional and domestic emails for a small home office, serving a number of Windows and Linux clients both on the internal network and, when

Its not quad band and in my opinion doesn't perform well enough to be used for anything but basic email and phone calls. This phone, even on the newest version of firmware (Sprint) hangs when syncing with exchange to the point where you miss calls even though you tried to answer them. If you turn on wifi or Bluetooth, it simply compounds the problems. It will also require (literally) a

hi everybody, I'm currently planning a small-sized web-applicaiton allowing users to call-in via phone. the phonecalls should be recorded and processed further by some custom scripts - sounds like asterisk is a perfect match for this app. however, during prototyping I have no ISDN-connection whatsoever available, so I was asking myself if it's possible to connect a cellphone via

Hi List, I am a newbie, just came to know about asterisk a few days back while installing suse 9.2. I have a question for which I am sorry to say,but I havntread through all the archives, but AFAIK i didnt get the answer in the archives. My question is like this. I am in Nebraska, US. I have broadband cable connection at my home. And I have friend and family in other country. Using asterisk

Celliax development and download site: www.celliax.org Celliax is a channel driver for the Asterisk Free PBX that manages GSM and CDMA cellular phones through an adapter, composed by a datacable (for commands) and an audiocable (for the voice) interfacing the computer soundcard. chan_celliax is also capable of making and receiving Skype calls through the cellphone, and has an app like

Hello, Avaya has a nice feature that allows you to a) ring both a cellphone and a desktop phone at the same time b) transfer calls (and access other PBX features) from the cellphone that recieved the call, as long as the call is bridged through the PBX c) while talking on the cellphone, pick up the handset on your desktop phone and the call is automatically moved there, hanging up the cellphone

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

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know

Hello When a new callers calls in, I'd like to save his number into a database, and send this to the user's cellphone, effectively syncing the Asterisk call database with a cellphone Contacts database. Is there a way to do this that will work with any brand/model? Thank you.

Hi I have the following setup to make outgoing calls: X-Lite (build 34025) at home behind NAT -> Internet -> Asterisk at work behind NAT -> Internet -> VoIP provider -> GSM gateway -> cellphone. I just tried calling my own cellphone, but there is no sound either way. Here's what I did on the X-Lite at home in the Topology section: IP address : Discover global address

I will be out of the office starting 12/09/2005 and will not return until 16/09/2005. Dear Sir / Mdm, I'm currently on course and are not in office. During this period of time, I have minimal access to internet and email cccess. As such, I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. In the meantime, for any technical assitance, please

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Hi, 1. Do you have any experience with receiving incoming SMS on an analog or ISDN landline ? How can then you differentiate an SMS call from a voice call ?