search for: externsive

Is anybody here familiar with the meaning of INVAL packets for IAX2? Every few days I get a dropped outgoing call in the middle of the conversation (the outgoing call has been connected for few minutes) when an incoming call comes in. The log reads the following when this happens: [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Immediately destroying 2963, having received INVAL [Nov 17 15:25:04]

Pretty good question Gabor. I can execute R once it is installed (if someone with rights installs it before) but not the installer. I can download the installer (with some pain). I know that some installers are actually compressed files in disguise, but I think this is not the case with R, right? I will study the exact nature of the restriction, and get back to you. Nevertheless, having a

Can you clarify what the nature of the security restriction is? If you can't run the R installer then how it is that you could run R? That would still involve running an external exe even if it came in a zip file. Could it be that the restriction is not on running exe files but on downloading them? If that is it then there are obvious workarounds (rename it not to have an exe externsion or

Hello, i have a simple but annoying problem. I have the following entry in /etc/asterisk/externsions.conf file: ---<Cut Here>--- exten => 10100,1,Wait(4) exten => 10100,2,Playback(transfer,noanswer) exten => 10100,3,Dial(${PHONE30},30,t) exten => 10100,4,Background(extension) exten => 10100,5,Background(is-curntly-unavail) exten => 10100,6,Voicemail(9999) exten =>

Dear all, I am trying to use the clipboard when writing a table. Typing: write.table(object, file="clipboard") leads to the message: Fehler in file(file, ifelse(append, "a", "w")) : 'mode' f?r die Zwischenablage muss unter Unix 'r' sein My interpretation is that I am not allowed to write into the clipboard from a program called from R. Is there a

Hi, I'm trying to get h323 communication working between asterisk (0.7.1) and Innovaphone Gatekeeper + innovaphone phones. chan_323 installed OK with currently recommended pwlib_1.5.2 and openh323_1.12.2. Registration asterisk with the gatekeeper works OK, externsion for my test(sip) phone gets registered with gatekeeper. when establishing a call between a h323 phone and asterisk I run into

We have a running Asterisk on a small server (RH 9.0) und were able to make calls via SIP. Since the quality was rather poor (one of us has only a 150/75 kbps DSL connection, iLBC did not work(?)) we tried to setup clients (FireFly) and Asterisk for connections via IAX. In some point seems to be a mistake. Asterisk says: -- Executing Dial("IAX2[83022777@83022777]/2",

.. and from a newbie no less :-) I have configured my BT101, and hooked it up to my * box. All is well. I have entered the following in externsions.conf, and this bit works: exten => 613,1,Answer exten => 613,2,Playback(demo-echotest) exten => 613,3,Echo exten => 613,4,Hangup If I pick up the BT101, and dial 613, sure enough I get the echo test.. All good. I have a TDM400 Card

Hi I have been lurking for a while, but now have a small problem or 3. 1) I have my inbound line via sip from VOIPUSER.ORG and have a simple extension selection menu on my * box. Internally the DTMF tones are present, (for xlite and * on same LAN), however calling in via the sip line from a pstn doesn't register any tones in asterisk. I have tried all the different DTMFMODE settings in the

I would simply like to dial an extension and get an individual Live MP3 stream but am unsure of how to do this. I'd like it to be different from my music on hold (not the same source) This trick works for music on hold: in musiconhold.conf ;default => mp3:/var/lib/asterisk/live,http://sourceofstream.com:8001/ I still wish to use local files for music on hold but want to dial an

I am not sure of the best way to do this, so I thought I would query the list. I have about 100 internal extensions ranging from 2000 - 2100. Each internal extension has a external DID number. For example: 2001 = 5552871620. As you can see the internal externsion and DID don't match in any way. What would be the best way to set the DID for when a extension dials out on the PRI? In

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

What is wrong with you guys? I asked for a zip, like R Studio has for example. Totally clear. I cant execute exes. But I can unzip files. Thanks Gabor, I had that in mind, but can't execute the exe due to security restrictions. Geez, really, treating people who ask questions this way just makes you don't want to ask a single one. On Thu, Jan 25, 2018, 11:19 Gabor Grothendieck

Hi, On Sun Jun 12 09:11:13 CDT 2005, Rich Adamson wrote: > > > exten => s,1,Wait(1) > > exten => s,2,GoTo(s,1) > > > > If I'm on the console when a call comes in, it loops through this bit of > > code a bunch of times. I'm guessing I could lengthen the "Wait(1)" time, > > but is there any other way to do this? > > Sure there is,