similar to: Connecting Asterisk to GSM

Hi all, I am interested in interfacing a GSM modem to *. I've seen a few comments about doing this, but I'm not clear whether people have actually made it work. I've used GSM modems for various data jobs, mostly high volume SMS (no, not nasty marketing stuff - high volume solicited SMS :-) ) . These only have analogue ports for voice. Does anyone know of units with digital voice

28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the

Hi All Does anyone know if one could use bluetooth on a cell phone with *? Would be nice to have your cell as an office phone combo. I heard that there is a bluetooth module for *? If so this should be possible? Thanks Doug --- NOTICE - This message contains privileged and confidential information intended only for the use of the addressee named above. If you are not the intended recipient

Editing subject line to reflect current status. On 1/26/06, Nilesh Londhe <lvnilesh@gmail.com> wrote: > Since T616 is not answering (and incoming calls are going to Cingular > voicemail after 30 sec,) I suspect the problem focus area is... > > > -- Executing Answer("BLT/T616", "") in new stack > > Is

The short of it: I am unable to compile chan_bluetooth on Asterisk 1.2.1 on CentOS 4.2. I installed using the Asterisk@Home 2.2 iso. Server is a plain Celeron 2.93GHz box. Asterisk source is in /usr/src/asterisk, newest chan_bluetooth source is in /usr/src/asterisk-test/bluetooth/chan_bluetooth (I have two older versions in other directories). Steps taken: Followed the instructions here to a

Hello, I?m trying to use my phone with asterisk to get GSM connectivity but I can?t compile the code. I got the asterisk, chan_bluetooth, zaptel and libpri sources, the last two ones compiled perfectly. I have added this to the /usr/src/asterisk/channels/Makefile: include /usr/src/chan_bluetooth/Makefile and I?ve also added chan_bluetooth.so to the CHANNEL_LIBS var. When I do "make

Just to close the thread. The problem was that I was using an old version of the code. If anyone has the same problem, you can download the code from here: http://www.thetechguide.com/howto/asterisk/bluetoothfiles.tar.gz Good luck, Danko ----- Original Message ----- From: "Danko Miocevic" <danko@santirso.com.ar> To: "Asterisk

Hi All, I have a telephony project for which I need to build a prototype to demo for management. The prototype must work on a GSM phone network. In the demo system, a call from GSM phone comes into the demo box. The demo box runs CallWeaver. Callweaver picks up the GSM call, answers it and plays a sould file, then dials out to a second GSM phone somewhere and connects them so they talk. My

asterisk-users-request@lists.digium.com is believed to have said: >Bluetooth enabled phones to talk to Bluetooth headsets; I guess there's a >protocol for the phone to talk to any Bluetooth headset, no matter who made >it. This protocol would have to include something to allow voice to pass >from the phone to the headset and vice versa. It might also include >something for

Because the VoiceBlue is only 4 channels and I am supporting 100 cell users, I needed a way to overflow calls to the PRI of all 4 channels are full. Unfortunately, there seems to be no built-in mechanism to determine if the gateway is full, so this script parses the output of "asterisk -rx sip show channels XXX.XXX.XXX.XXX" to determine the number of channels currently in use. Hope this

Hi, i was reading about connecting a cellular phone over chan_bluetooth. I was wondering, if one is able then to make/receive concurrent calls or if you can make just one at time? Peter

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI> Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas?

I'm looking at version.h installed by Asterisk 1.2.3/4 - and the default value is 000000. I thought the value should be 010200. I know many people have problems compiling chan_bluetooth because of this inconsistency. Anyone has the last word on this?

Hi List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! Also it seems that VoIPJet takes forever to return 'circuit busy' while NuFone does it instantly. At any rate, is there like a reliable third VoIP provider I can use for fallback when the two others are busy? Cheers, Jean-Michel.

Hello, I use asterisk to to IAX2 trunking between London POP & Reunion Island pop. I would like to know if it's possible to do a kind of call-limit (i.e. restrict to XX) channels but on a per dialcode and / or destination basis. For example: [trunk] ; reunion proper, i want to send no more than 24 channels exten => _0262XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN}) ; reunion mobile, i want

Hi List, I was wondering if anybody had tried running Asterisk inside virtualization software such as Xen. Are there known problems doing it? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

I am running Asterisk 1.0.9 with chan_bluetooth (kernel 2.6.11) I have managed to get the module loaded and it connects to my phone and dials (Nokia 6310i) okay, but once the call connects I hear no sound on either end. It is not who I am calling since it works with any other bluetooth headset or without one. It isn't the SIP phone or codecs because they work with any other number. I think

Has anyone had experience with Theo's chan_bluetooth with asterisk-1.2.0 on FC3. When I try to make a call to my home through the channel my phone shows calling but my home phone never rings. I get this on screen: == Spawn extension (outgoing, 3234600, 2) exited non-zero on 'SIP/2299-9131' -- Executing Dial("SIP/2299-7743", "BLT/John/3541107") in new stack

I have been looking at chan_bluetooth, so far being unable to make it compile with Asterisk SVN trunk. I was wondering about the different ways it can be used. What I have read so far implies two possibilities: 1. Asterisk pretends to be a handsfree unit, and can use the cell phone for placing calls over the mobile network, or answer inbound calls from the mobile network. 2. Asterisk