similar to: Syncing phone numbers DB with cellphone?

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.

Hello, I'm trying to fill CID data into the astdb using AsteriskWin32's asterisk.exe, to no avail: The batch file stops after the first line, and just waits: ---------------------------------------- rem c:\cygroot\mystuff>import.bat rem rem c:\cygroot\mystuff>C:\cygroot\bin\asterisk.exe -rx 'database put cidname 123 "My cellphone"' rem rem Asterisk module

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 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

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

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 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

Hi, I've been trying to dial out to a cellphone, but all my calls get redirected to 066 (the emergency number at my city, like 911) does anyone know how to fix this, any ideas,? does anyone from mexico has done this? Any comment will be highly appreciated, Regards, Claudio -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

Hi there, I have a cablemodem, ARRIS brand, model tm502G. It has two FXS ports. I was wondering if anyone has details about the correct signalling of these FXS ports when connected to original X100p. Tests: fxsks on the zapata.conf and zaptel.conf files. From my cellphone I call the ARRIS, it starts ringing but the zap channel sees no call coming in. fxsls on the zapata.conf and zaptel.conf

Hi, I've connected a Nokia 6310i Cellphone via the USB Port to an ESXi 4.1 Host and added it to a CentOS 5.6 64Bit server. The problem is, that I wont get an suitable tty device (/dev/ttyACM0) like I get when I use Debian.. here is the output of the log files: dmesg: usbcore: registered new driver rndis_host usbcore: registered new driver cdc_acm drivers/usb/class/cdc-acm.c: v0.25:USB

-----Original Message----- From: asterisk-users-bounces at lists.digium.com on behalf of Steve Edwards Sent: Fri 7/9/2010 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] False answer() being sent by cellphone providers On Fri, 9 Jul 2010, Mike Ely wrote: > (off list) Continuing to veer off-topic... > Yes indeed we do. The telcos

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

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

Hello list, I'm using the following macro when calling an external callphone/GSM number : [macro-press1] exten => s,1,NoOp() exten => s,n,Playback(/var/lib/asterisk/sounds/prompts/press1) exten => s,n,Read(INPUT,,1,1,1) exten => s,n,NoOp(input : ${INPUT}) exten => s,n,GoToIf($["${INPUT}"=="1"]?exit:hangup) exten => s,n(exit),NoOp(call accepted) exten

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

Friends: The icecast server is very robust, but has some goofiness with regard to use with cell-phone players. Has anyone else encountered these issues? When I use a Blackberry to access the icecast stream, I get orphan and duplicate client threads. Background: > testing with Blackberry Curve 9330 > Access icecast stream by clicking on the link on the website:

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

Hello A user seems unable to type DTMF in our Asterisk IVR menu. Can this be due to their phone or PBX that disables DTMFs when a user is off-hook? Thank you.