similar to: Asterisk in support enviroment, some CTI maybe & RT integration?

Hello people. I would like to know if someone knows about any applet to include in a web page to start calls. What I am looking for is something that doesn't allow users to change numbers, or any other option, so I can include it in my web page and force them to call to me and no one else. I have tried JIAXClient, but it allows people to call anywhere, and what I want is just a configurable

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Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josu?

Hi all, Is it possible and easy to make a CTI server with Asterisk? Florent,

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Has anyone found a softphone that supports pulling it's configuration from a central server via TFTP/FTP/HTTP, much like hard desk phones use? I'm looking for something for a call center that I can provision from a central location by generating config files. If the phone has "soft keys" (yes, I know they're all soft - but you know what I mean; programmable buttons whose

Hi, Someone who has had experience with shoretel VoIP systems, can you please give me a run down of how Asterisk is either better or worse? I am completely unfamiliar with Shoretel systems, but someone had suggested we look into them. I said, you bring your Shoretel features, and I'll show you 10 things Asterisk does or can do that Shoretel doesn't do. I still believe that's

Hi, I would like to set up a call center with free seating agents. However I would like to indicate the agent status somehow on the terminal, to tell the agent if she has been logged out due to non-answer. Does anyone has a good idea how this can be achived? best regards -- Alexander Topolanek http://www.topolanek.at

Greetings. I have a scenario I would like some advice on. I have a 100,000 square foot building that we will be moving some work crews into. It has offices on each end of the building and a fiber line between them. I currently have an asterisk 1.2 system in place and about 30 phones. My problem is they want a few fax machines out in the warehouse area where I currently have no wiring for

Hello, I sometime forget to set the "Repository" when uploading a patch on Phabricator, and that prevents from adding llvm-commits as a subscriber. [cid:image001.png at 01D61A45.E388B060] Would it make sense to set 'LLVM Github Monorepo' as a default? Or subscribe 'llvm-commits' automatically when creating a patch? Thanks! Alex. -------------- next part --------------

Hi there, I am integrating CTI functionality to my java application and using Asterisk as PBX. I need some advices as to whether I am in the right track. ? Asterisk server is configured and working fine. ? I have a generic sip hard phone ? I have X-lite soft phone installed. To provide for CTI functionality, I am utilizing the asterisk manager API, I designed a java application that can do the

Hello, Is it possible to use the asterisk manager interface to originate multiple channels? like Action: Originate Channel: SIP/101&SIP/102 So that both extensions 101 and 102 rings simultaneously. I am using asterisk manager interface over http. Thanks

I have a pc with the following characteristics: Pentium IV 2.4Ghz HyperThreading 512 MB PC3600 Dual DDR RAM Seagate 80GB SATA HDD 4-port ethernet 10/100 PCI Card Netgear MA-311 802.11b Wireless Card On this machine runs a VPN server, an Apache server and an Asterisk Does anyone know or have experience about the number of users that could be supported for VoIP at the same time?It is

Im having a trouble understanding the config files setup even with some documentation ive read such as the handbook, maybe im just illiterate. Anyway do you think some one can point me to some examples of real config files. Such as IAX, Extensions, and Sip. I just cant grasp the concept for some reason. If someone would like to help me out, maybe even explain one on one? Thanks a lot

I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggestions from those who know? Jason

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Dear Sirs, First of all I would like to thank you for support PCM product in your project. Kindly please be informed that POWERCOM launch new USB-controller with productID 0X04. Updated USB-controller compatible with previous version which supported by your usbhid-ups driver. There is added some command and changed idProduct. Vendor ID stay the same 0X0d9f without changes. Detailed information

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We have a site that is running Mongrel and is down. Below is the Mongrel log file information. The server is a Linux server with 2gb of RAM. EV1 (hosting site) says that since this is a Linux server that the RAM is cached and that there is plenty of free memory available. Can anyone make sense of this error and have some suggestion of options that I may take to correct this issue and get