similar to: Pass variable to context (NOT macro)

Hi, We have a Data/Voice service supplied through an integrated T1. Does anyone know if Digium T1 card will support the splitting of the Voice and Data? Regards. __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

What is the best solution? I dont want to have modify firewall's at all or do port fowarding. Ideally I would like a solution that with either a softphone or wireless hardphone one could connect via friends, family, or hotspots without reconfiguring their devices. What are people using? STUN? SER? Thanks in advance! -blake -------------- next part -------------- An HTML attachment was

Hi, Is there anyway to eliminate the color coding (for example [1;36;40m) to be stored in asterisk log file? Regards. __________________________________ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/

Hi, I'm planning to get my Asterisk box out of the LAN, get rid of my router and make the box acts as a Router, Firewall, DHCP Server (with Shorewall). I'll do that to be able to use some SIP clients remotely. Does anyone doing the same with the Asterisk box, is it a good idea, is there any other solution for the SIP emote Clients. Regards. __________________________________

Hi, I noticed that the sound directory is missing from asterisk-1.4.0-beta4.tar.gz. This directory (7 M) witch existed in asterisk-1.4.0-beta3.tar.gz has GSM Core Sounds and some MOH. Does anyone know why it has been removed from the latest beta? Regards. ____________________________________________________________________________________ Sponsored Link Mortgage rates near historic

Hi, I was wondering if it is possible to connect a skype phone adapter, for example: http://zonetusa.com/DispProduct.asp?ProductID=191 http://www.actiontec.com/products/communications/ipw_usb/index.php http://www.eradian.com/ERadianUS/staticpages/SkytoneRST301Details.htm http://www.dlink.com/products/?pid=466 http://www.usr.com/products/voip/voip-product.asp?sku=USR9620 to a TDM400P, so that

Hi, I'm looking for a tutorial or installation guide for SER to be used with asterisk to solve the remote SIP agent problem. All the documents available are for large scale installation. Any help is highly appreciated. Regards. __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

I am trying to make and install a version of llvm that I check out of cvs last night. When I configured LLVM, I set the prefix to a directory inside my home directory. However, when I did a make install I got the following errors: llvm[3]: Installing Debug Bytecode Archive /lib/libc.a /usr/bin/install: cannot create regular file `/lib/libc.a': Permission denied make[3]: ***

I checked llvm-gcc 4.0 out from svn yesterday and am compiling it on 3 different machines. I was able to compile it on 2 of the machines, but the compilation failed on the third machine with the errors below. The machine that the compilation failed on is running Fedora Core 4. The processor is a AMD Athlon(tm) 64 Processor 3700+. The version of gcc I have on the machine is 4.0.2. I tried

I apologize for my poor English..... So, i don't really understand 'how to' realize thus.... When you use the cmd Dial and want to get $ from caller channel to callee (or callee channel from caller), which way is the right way ? Sorry, i've take a look to the wiki and asterisk code and is'nt limpid (for me) use Macro in the dial cmd.... hum it's on the caller side and i

On Wed, 22 Aug 2007, Ryan M. Lefever wrote: > I checked llvm-gcc 4.0 out from svn yesterday and am compiling it on 3 > different machines. I was able to compile it on 2 of the machines, but > the compilation failed on the third machine with the errors below. The > machine that the compilation failed on is running Fedora Core 4. The > processor is a AMD Athlon(tm) 64 Processor

Chris, I'm a little confused. I am experiencing a crash when compiling the llvm-gcc frontend. According to the bugpoint documentation, bugpoint is used to debug "optimizer crashes, miscompilations by optimizers, or bad native code generation," which seems like it implies that the frontend compiles. Also, the http://llvm.org/docs/HowToSubmitABug.html documentation seems to

Hello I just coding a AGI script for billing. - For external calls, I pass the call directly on a trunk. I do : Dial(trunk1/extension) -> OK ! - For internal calls (shortcode, others users ...) I am Dial(Local/extension at context/n) The problem is that through chan_local.so, I sound as it cut! Example if I call the voicemail ... "You have No messa ..." or "You have

When a caller in the queue is connected to an agent, the call is placed to the extension and context specified using Agentcallbacklogin. This allows for me to add extra things to the diaplan *before* calling the agent. Now, I want to be able to use a device, rather than agents. So I can use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device

hi all - i'm just wondering what sort of code people write to essentially performa an aggregate call, but with different functions being applied to the various columns. for example, if i have a data frame x and would like to marginalize by a factor f for the rows, but apply mean() to col1 and median() to col2. if i wanted to apply mean() to both columns, i would call: aggregate(x, list(f),

I am migrating my app to Asterisk12 and pjsip, but I cannot find chan_local, what happened?

Hi all, Is it just me and not reading the docs right, or has anybody else had problems with the AbsoluteTimeout application and the 'T' extension when used inside a macro? [macro-attended] ; ARG1 is the device to dial out on, SIP or Zap, or whatever ; ARG2 is the extension to dial using 'attended' dialing exten => s,1,AbsoluteTimeout(30) exten =>

Happened twice. There might also be a race condition and some bad pointers in chan_local.locals_show. First the segfault. CLI> show locals <unowned> -- 6001@default Segmentation fault (core dumped) [root@mars asterisk]# ll -tr total 22260 [...] Loaded symbols for /usr/lib/asterisk/modules/chan_local.so #0 __pthread_mutex_lock (mutex=0x5d8) at mutex.c:99 99 mutex.c: No such file

I am using Asterisk 1.4.4 on a CentOS 5 machine for a small call center with 6 agents. I am using realtime for queues and sip and I am also trying to use realtime static to load agents.conf. The only problem I am having is that no agents are loaded when I start Asterisk. I have to manually do a "module reload chan_agent.so" so the agents get loaded from the database. Obviously

Hi. When we don't have too many IP addresses available and we want to run for example www server inside a jail, but use the same IP address as the main system, we need to actually use an internal IP address and forward http port with firewall from external IP to jail's IP. In that way we know that if somebody breaks into out jail, he cannot run sshd server (we have keys, I know) or any