search for: handphones

Hi All, I have one asterisks server and 3 client (i'm using voip sip client for my handphone). I've configured sip.conf and extension.conf with 3 user different. And nothing wrong with them, i could make them to make a call too. what i want to ask is, i was try to use 1 user (ex:1001) in 2 different client. example: client 1 (1001) make a call to client 2 (1002) --> ok then in client

...phone, the Motorola CN620. Don't know whether that one ever made it to the market or not, but I read a review on c|net about another upcoming model, the A910. The A910 is Linux-based, and offers WiFi on top of GSM, GPRS and Bluetooth. You can see the picture at http://asia.cnet.com/reviews/handphones/0,39001713,39094707p,00.htm? Looks like it also has color screen, but not much other specs listed yet. Flynn

Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of my FXS zap extension created. dialparties.agi: Starting New

Hello, Have a look at the plm package https://cran.r-project.org/web/packages/plm/index.html It has a convenient way to structure your data into panel according to some id. Best regards, Jeremie On Wed, Jan 10, 2018 at 5:41 PM, deva d <devazresearch at gmail.com> wrote: > dear all, > > i need some help in structuring my data file for a hierarchical time series > analysis.

Howdy, I'm building a webapp to allow my techs to do minor dialplan edits and trigger a reload on my PBX's running 1.8 I have no problem triggering a 'reload pbx_config.so' via manager, The problem is how can I see the results of my reload? For example a missing close parenthesis which would show in /var/log/asterisk/messages [Apr 10 13:46:16] WARNING[23911] pbx_config.c: No

hi all, i recently subscribe for an isdn and terminate it on a 3825 router. i used it as a sip trunk for my asterisk. i'm a newbie when it comes to ISDN. and i've been experiencing some issues: 1. Call Hangup: When hangup is initiated from the outside the extension (softphone/ip phone) does not hangup, is this normal? shouldn't asterisk hangup the extension as well when it

dear all, i need some help in structuring my data file for a hierarchical time series analysis. can someone help please ? i have a 600 row database in the nature of a panel data, with 3 time series values of interest. the data also has 4 classificatory variables comprising a code for each entity in the panel, a value for time (year), and classification of type of entity and a further sub-group

Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_ W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewIte m Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid =27 But looking for real world feedback. Cheers,