search for: phonelabs

...hone that supports the wireless profile *NOT* the headset profile I know jabra has the A210 http://www.jabra.com/JabraCMS/NA/EN/MainMenu/Products/Accessories/JabraA210/ JabraA210 but it only support the headset profile .. I am trying to shoe horn my current braindead cell into DocknTalk http://www.phonelabs.com/prd05.asp, and the BlueTooth interface requires a WirelessProile

...rt for more than one sim card is a plus. Since it's a > home/hobby use, I would prefer a low-cost solution. Any ideas (may be > off-list) are welcome). > The solution I use works very well if you need to be able to take the mobile phone away with you when you leave the house. I use a phonelabs.com dock'n'talk with a bluetooth module connected to a standard digium one-port FXO card (XP100). I can make and receive GSM calls via my mobile from asterisk, treating it as "just another channel". The phone automatically connects to the dock'n'talk when it comes into blu...

Evnin' (o; As chan_sccp is pretty much dead, doesn't compile on FBSD anyway and isn't supported on * 1.4.x I tried going with chan_skinny... The Cisco 7970 registers and is being acknowledged by * but that's it... I see no lines on the 7970 display configured and it is not reachable or it can't make any outboudn calls... The docs are pretty non-existent for skinny and the

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

...is a skype to asterisk gateway in software (probably won't happen because of intellectual property reasons) so I needed three FXO ports and one FXS port. Below is how I plan to connect all these. FXO #1 for Vonage ATA RT31P2 www.vonage.com FXO #2 for Cellphone connected via Doc-N-Talk from www.phonelabs.com FXO #3 for Skype to RJ11 adapter (www.echostore.com) and skype http://www.skype.com/products/skype/linux/ and skypemate for linux http://www.yealink.com/en/download/install-SkypeMate.zip and the only FXS for the existing RJ11 house wiring that has been safely disconnected from PSTN several yea...

Hello, In my quest to figure out the source of the random echo on our shiny new asterisk install, I have been using ztmonitor on the TDM400p channels for the good part of today. I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to them (last 2 channels are unused but configured in zaptel). Even when the lines are onhook, the Tx values settle down to 0 but the Rx values still

When I installed my first home-PBX three years ago, I was looking at "cellsockets" -- devices which will accept certain cellular phones and provide an RJ11 jack, generating the ring-voltage and recognizing DTMF, which in turn makes your cell-phone look like a CO line. Pretty cool stuff, in theory, but it just didn't seem to be worth the cost, especially since it locks you to a

Hellooo... Is it possible to fill a variable of PHP-based-AGI-script from dialed extension ? This is what I need to achieve: If someone dial an extension, say 777, I want the dialed extension (777) be filled into PHP variable. I need the dialed extension become a condition of PHP script. Help please... Thanks romsun _________________________________________________________ This mail sent

Evnin'... Anybody got an idea where those CLI messages come from? [Mar 24 20:30:05] WARNING[4518]: chan_sip.c:12296 handle_response: Remote host can't match request NOTIFY to call '0354c42214142e5d6cb8e05568c59837@10.0.2.2'. Giving up. Interestingly all are caused by local IP used by asterisk-1.4.1 cheers rick

Afternoon A little off-topic...but... Does any1 know why recorded call with "IAX2" in the filename are not displayed within ARI? LittleJohn's website isn't a helpful place for ARI (o; cheers rick

Evnin' Now I tracked my problem down why ARI won't display most of the recordings... It write a recording for examples as: 1175031785-SIP-0615000995-0872a000.wav But it writes to the field "uniqieid" into MySQL database as: 1175031779.16 WHen I overwrite the "uniqueid" field with the value from the recording file, the recording is playable within ARI:

Evnin' As I didn't find any answer I'll try to rephrase the problem (o; Any idea why the latest asterisk-addons-1.4 write wrong uniqueid into mysql database? Asterisk-1.4.2 creates call record files with the uniqueid prepended: 1175107269-SIP-999-0876c000.wav But into mysql database it writes an uniqueid of: 1175107260.88 but should be: 1175107269 Any idea why the