similar to: transfer call

Hi.. Iam new with asterisk PBX, and i have read about asterisk video call.: my question: 1. Is imposible to establish system video call (from Phone with GPRS/3G enabled to Computer Running Softphone like X-Lite) over Asterisk Gateway.. 2. If posible what requirement (Hardware and Software on my Asterisk,PC or My Phone) Thanks Joko Pitoyo -------------- next part -------------- An HTML

Hi, I had asterisk installed, ISDN-adapter, some x-lite software-phones and I can call betweens the softphone- and 'normal' phones during the ISDN-card. 2 questions now 1) Is it posible to create policies, so that some SIP-users can dial ALL numbers, and some SIP-users not are allowed to dial eg. 900xxx-numbers, 30xxxxxx (mobilphones), 40xxxx(long distance) and if possible on time

Dear all, I have tried to run an asterisk instance together with XLite on a single machine (a PowerBook). The intent is to take advantage of IAX connections to easily cross NATs while traveling. While the IAX setup proved 'easy', just having to fiddle a little with working configs at both sides, I did not succeed so far in getting XLite to connect to the local Asterisk server, AND be

Hello; I'm now using mainly R for windows, mainly because I'm writing a tcl/Tk interface for some people, and I've got two questions. I'm an absolute beginner with tctk or tcktk use under the R GUI. 1) Is it posible to create a shorcut that launchs the R GUI and automatically reads the "source code" of the tcl/tk script to also launch the tcltk interface? 2) Is the

Hi, I am can make local extension to and from SIP X-Lite softphone, but I can't dial out using X-Lite but local analog works just fine. Here are my conf files any idea? Thanks, David my sip.conf [general] bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) allow=all [3000]

I have the very same situation in one of my networks. To solve this you can dial out from the softphone and to move call to the phone you can simply transfer call to the same user (just if you were transferring call to yourself and the other device will ring. While, as you notice, you cannot dial a device, you can surely call your user to tranfer from a device to another. Please note that call

In some configuration experiments, I have one IAX softphone connecting over the internet to my Asterisk system. (I should note that, while there IS a NAT firewall between Asterisk and the public Internet, I have port forwarding set up on that firewall to send all IAX traffic to the Asterisk machine.) The IAX softphone connects to either an analog phone on an FXS port on the Asterisk server,

Hi all, Im having problems using a fax machine conected trough a grandstream 286 sip ATA, it must be able to send and recive fax from pstn, but fax always ends with communication errors 252/244/232 and others. Im using alaw/ulaw codes on pass trough mode, also have tried asterisk faxdetection, nvfaxdetect, disable echo cancellation by hand always with same results. Grandstream ATA is using

Hi all, i''ve coded htb-gen, a GPL htb frontend and much more... htb-gen is meant to be an easy, scalable, yet powerfull, bandwidth management tool. You can set up/down portions of bandwith for each host or network, that goes trough your router/firewall. Prioritary traffic(web, mail, gaming, ftp, voip, streaming) is preferred over Junk traffic(kazaa, emule, etc). Also dynamic bandwith

After searching around, I've been unable to to find any relevant info on this. Perhaps the group can help? I am seeing something strange with a new Sipura SPA-3000 (and I've noticed this also with an IAX softphone): When I dial 777, this dialplan (in extensions.conf) is run: exten => 777,1,Dial(Zap/1/2345678) exten => 777,n,Hangup The number is answered by the called

Hi. I have a beginner conceptual question about Asterisk: Let's suppose that there are 4 softphones registered in my Asterisk and all of them are currently online. In addiction , there is no call. Suddenly, one of these softphones sends a SIP message to the Asterisk. In this case the dialplan will execute the instruction ' exten => 2005,1,Dial(SIP/2000&SIP/2001&SIP/2002,

I have to install a VPN between three places (A, B, C), the A is a Windows 2000 will be the VPN Server and the others are Windows98 and are the clients. B and C doesnt need to comunicate.I have the following questions and i would appreciate some help: 1) I only have to install tinc in the Server, or i have to install it in the clients too? 2) I read in the documentation that tinc is compatible

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, Currently i'm running asterisk 1.4.2 on fedora core 6 x64, using sip i configured x-lite on client n its runs well. But the instant message won't work. it says a notice " the person tou are sending messages to is using an earlier version of x-lite, which does not support rich text and emoticons. you may have to send some of your

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), I have several extensions that can register 2 separate devices (chan_sip) ( FreePBX calls this Devices & Users mode : Users are extension/internal number, devices are the 'SIP Accounts' for the internal 'endpoints' ) (this

Greetings, I have problem using Dom0 as NFS(v4)-server and DomU as client. When ever the client actually tries to copy data to nfs-server the virtual network (bridge) between the two hosts completely freezes and stops working. After this of course the client hangs waiting nfs-server to answer and server continues to try to contact nfslock daemon on the client. Dom0 is still accessible from

Hello, any one advise a good, strong and free softphone that can work with SIP or/and IAX lines and supports attended transfer ? Thanks for help. Mit freundlichen Gr??en / best regards Andr? Herrlich IT-Operator / Developer ____________________________ LetMeRepair LMR Service and Consulting GmbH Fichtestr. 1A 02625 Bautzen Tel.: + 49 - (0)3591 - 2722 - 1451 Fax: + 49 - (0)3591 - 2722 -

Hello, Can you explain how to test blind transfer in asterisk. Here is my test case that hasn't succeeded: I have configured blindxfer => # in features.conf. I have called an iax user from my iax softphone. The called party responds to the call, and tries to transfer the call by clicking the # key followed by the number of another iax extension where I want to transfer the call to.

I can call/receive fine from a zap fxs connected phone to my voip provider through asterisk. I can call from a sip softphone extension to the zap extension just fine. However using a sip phone extension connected to asterisk and calling out through the voip line there is no sound. I was able to resolve the problem temporarily by making the asterisk server the DMZ and entirely dropping the

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the