similar to: : PSTN calls

I'm moving asterisk to a laptop, so can't use the dahdi board. Is there any supported USB dahdi device ? I see the Sangoma USBfxo device, but the dahdi driver no longer supports it. Anything else ? sean

Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : alice at pbx.com should be able to call bob at

What is the easiest way to connect my "black phone" to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? Thanks -- Diogo Saad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090526/77fcb91b/attachment.htm

Hi all, I am running centos 5.1 and I wish to change the domain name and dnsdomainname of my PC. currently the settings are-- $ hostname sipx.com $ hostname --fqdn sipx.com $ domainname (none) $ dnsdomainname com I have searched in the net for tips but everywhere only the hostname change is provided. I need to change/set the domain name and the dnsdomain name on my pc to sipx.com and this

I'm looking for a reliable ATA FXO/FXS adapter. Linksys 3102 - a lot of echo problem + two of them died within a year (not reliable) Sangoma USBFXO - problem installing drive in Gentoo. I've tried two Chines units: AG-188N and YGW30B none are of them have real FXO port that will register with Asterisk. Any other recommendations; (I don't like internal cards). -- Joseph

Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. The project manager instead choosed sipX and said it scales well for large user base. I had an Asterisk running

Hi, I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me what is the good ratio for incoming and outgoing analogue/ digital PSTN lines. Regards Smir

Hello community, I have successfully set up asterisk free PBX server and I am also able to connect to it by softphone. Now as next step I want to extend this to PSTN , My Required scenario: I need a number which will connect outside PSTN world to my PBX and by applying extension particular softphone or connected normal phone should get connected. Which hardware I need for it. Also please

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

Hello I'm contemplating building an Asterisk voice server out of the compact Asus EeeBox: http://www.asus.com/products.aspx?l1=24&l2=165 But they're so compact, they don't have a PCI slot to handle an analog phone line. I'd like to minimize footpring and cables: Besides analog/SIP boxes like Linksys (extra cables + transformer), does someone know of a USB adapter that is

How do you configure your Polycom phones? Is it enough to configure one line appearance? Or is there a way to configure a roll over? Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts

I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse

Hi all, I want to deploy a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router <http://www.voip-info.org/wiki/view/SIP+Express+Router>: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org <http://www.voip-info.org/wiki/view/Vovida.org> * sipX

Hope some can help me. I have a PSTN coming into TDM400 into Asterisk. We also have direct telephones connected to the PSTN bypassing the Asterisk. When a call comes in on the PSTN the direct connected phones ring first and if no one picks up , Asterisk picks and get routed to internal sip phones. I am not able to find what I should tune to make the calls always go through asterisk without the

hi, I got a digit networks x100p card and instaled asterisk. everything went fine and upon calling the phone asterisk issues a notification. Now i plan to turn it into an ivr and modified extensions.conf to first record some messages , problem is 1-)I am not able to understand how extensions refer to in my case ( a single analog phone line plugged into line jack and a phone into the phone

Hi Vishal, Ruijie, Thanks for your interest in the project. To get started, the first task would be to merge the 5 patches on top of trunk llvm. The list of patches are listed in the project description: http://llvm.org/OpenProjects.html#llvm_mergesim Please create an account in llvm phabricator (reviews.llvm.org) if you haven't already, and put your patches there. Let me know if you have

FYI, The LLVM Performance Workshop will be held at CGO 2017. The workshop is co-located with CC, HPCA, and PPoPP. If you are interested in attending the workshop, please register at the CGO website: http://cgo.org/cgo2017/workshops.html When: Saturday February 4th, 2017 Where: Austin, Texas, USA ---- Hi, CGO workshop and tutorial organizers, This is a friendly reminder that early

Hi Stefanos, Thanks for your comments. I added both as reviewer. > One question though. Are you sure that this: > This helps with LICM when instructions inside a conditional is loop invariant  > is not achieved with the current LoopRotate pass? Because AFAIK, it does. Basically it inserts > a guard (that branches to the preheader) and then passes like LICM hoist invariant

Hi I followed the steps on http://llvm.org/docs/GoldPlugin.html#lto-how-to-build to build the gold plugin on FreeBSD but ! Heres a link to the screenshot of the error: http://postimg.org/image/anlpuufbl/ This is the error message that it shows and so I am also unable to get ld-new. I checked and no CFLAGS were set in etc/make.conf. How to proceed with the installation? I am using the deault

Hi, I have observers for a group of radio buttons, an observe_field for each radio button (I think its not possible to have one observe_field for a group of radio buttons). Now, the problem is that when I click on one radio button, all the observers get trigerred, so the method which the observe_field calls gets the wrong result. I just want that method to get the radio button that is