search for: sofphone

Hi, we are trying to setup a prototype Asterisk machine for our call center (15-20 users). We are encountering some difficulties in finding a 'good' softphone (SIP/IAX). Suggestion/experience? Is there some product available for Windows with modifiable code? Is there some freelance developer potentially interested in creating custom version for us? Thanks in advance -- Mimmus

Hi everybody, This is the console output of the asterisk server. debian-te410*CLI> sip set debug peer 2002 SIP Debugging Enabled for IP: 172.26.48.113:5061 I have a sofphone with user 2002 registered on the server on the ip 113. I am trying to place a call to the sofphone on this ip. I have written a simple php script which utilises the exec_dial function inbuilt in phpagi.php file. I have tried diff ways but can't seem to get it work. Can please some one sugge...

Hello to all I need send a data to sofphones screen when I use a Dial () . Thanks a lot Regards Andres Gomez -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070430/d26033b2/attachment.htm

...ces but it dind worked because the lack on the sound and the microphone do not work on the thin clients using terminal services, we tried to install Linux in the Thin Clients but they are to small to to a have a decent OS inside, im waiting a demo version of Citrix in order to see if we can get the sofphones work i dont know what else to tink because in top of tha we nee to get a dialer tha supports this enviroment. PLEASE ANY HELP WILL BE MORE THAN WELCOME.................................................................. Ignacio. -------------- next part -------------- An HTML attachment was scrub...

Hi, i'm using an elesign voip gateway esc1700 to call to one iax sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when I make the call using the esc1700 the communication is dropped, this is the log portion: Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Call accepted by 200.93.220.21 (format ulaw) Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Form...

...some changes in extensions.conf to test the incoming calls so that these are derived to a SIP extension, I found something that draws attention to me: if I test calling to my PSTN line from a mobile phone, when take the call from the SIP extension (softphone), if the mobile phone releases the call, sofphone do it too without problems, but if I release the call from sofphone, from the mobile phone I see that the call chronometer continues advancing as if the mobile phone not yet releases the call. Which can be the problem? Seeing in logs of the CLI, I observed the following thing: - -----------------...

Hello all, I'm a newbee and i'd like to test some analog modems i have before purchasing any new hardware. I'm using: - Pentium 500 MHz - Several 56K and 33.6K analog modems (internal and external). - ISDN BRI 1 port card. - A LAN i've successfully tested some SIP sofphones. - An ordinary telephon line (and an ordinary phone :) ). I only modified extensions.conf in order to test two softphones but i have no idea of configuring Arterisk to test a modem (special drivers, signaling configuration, complex dialplan configuration). Can anybody provide me a simple s...

...No 2-) Which is the best codec to use with different softphones a-) SJphone : b-) Lipz4 : c-) Gnophone : 3-) Hardphones need to be plugged into the DSL router ( basically need an internet conn ) ... and need no digium cards. a-) for recieving calls : Yes/No b-) for making calls : Yes/no 4-) Sofphones run on machines different from the one running asterisk and can connect to asterisk , which routes the different phones according to contexts. Yes / no 5-) Softphone can run on the same machine as asterisk and register the same way as if they were ona different machine : yes /no 6-) Softphon...

...172.26.48.113 D N 5062 Unmonitored 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 1 offline] 113 is my ip. This may be the reason that when i do 'sip show registry' no value is displayed even though i get message of successful registration on my sofphone. debian-te410*CLI> sip show registry Host Username Refresh State Reg.Time Please help, what may be the problem here, should the status be different? I want to make a call from server to the 2001 user through a php file, how can I do so?? Thanks in advance Nikh...

I do part time consulting work. I need to setup an asterisk system to allow me to record both inbound and outbound calls to clients. I have one client that is just a PITA. The client has changed their mind three times so far and we are at step one. I have a spare slackware box and a seperate phone line for the consulting work. I have MCI Neighorhood as my carrier. What I need to know is: 1.

Hello everyone. I recently boght a Nokia BH900 headset and USB bluetooth dongle and I'd like to use them to make calls from a sofphone. I managed to this with boxe XTen-Lite and the Zoiper - but they both see the device as a simple sound card through the BlueSoleil drivers. While this is allmost usable, the headphone seems to be kept in "transmission" mode all the time and I get a constant "hissss" in the heads...

...tion till now is to restart my softphones.... I must say that sometimes it works fine and i do not encounter this problem. But it happens very regularly, too often i would say. I would like to know if some of you noticed that problem ? And if there is a solution of course... I tried with different sofphones : iaxlite, or other i do not remember the name... I also have the sources of iaxComm (an open source softphone from iaxclient project), do you think it could be a softphone problem, and if i can do something to change the code.... To illustrate my problem, here is the full output of a failed call...

Anyone know?a?good?IAX2?softphone?for?Windows that has?g729?and?it is?free? att Eduardo -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110418/024f95c7/attachment.htm>

Hello, I am just asking this because I am note sure if the problem is on my side or not, I saw some comments on SIP realtime today so I was wondering, has anybody has SIP realtime working with a softfone ? If yes, please confirm, that would give me a light. My previous message to the list is below. Thanks. Frederic ----- Original Message ----- From: Frederic Jean To: