similar to: problem with outgoingcallsUnabletocreatechannelof type 'ZAP' (cause 34 -Circuit/channelcongestion)

Nik, This definitely helps! Please check your dial command. You've got "Dial(Zap/0/mynumber)" and I think you might possibly want it to be something like this: Dial(Zap/1/mynumber) or Dial(Zap/g0/mynumber) I don't recall there being a zap channel zero, but it is common to have a group zero. I would recommend trying Zap channel 1 - Dial(Zap/1/mynumber) - before trying the

Nik, Looks like you're making some progress. When I first started using A@H I had trouble getting the outbound dialing to work. I wasn't sure where to start, so what I did was skip the macros in the dial plan. I wanted to play around with exactly what digits the telco wanted to see. So I put a specific extension in my [default] context like this: exten =>

I think rapply() was changed to act like lapply() in this respect. In R-3.1.3 we got rapply(list(quote(1+myNumber)), evalq, envir=list2env(list(myNumber=17))) #[1] 18 rapply(list(quote(1+myNumber)), eval, envir=list2env(list(myNumber=17))) #Error in (function (expr, envir = parent.frame(), enclos = if (is.list(envir) || : object 'myNumber' not found lapply(list(quote(1+myNumber)),

Another aspect of the change is (using TERR's RinR package): > options(REvaluators=list(makeREvaluator("R-3.1.3"), makeREvaluator("R-3.2.0"))) > RCompare(rapply(list(quote(function(x)x),list(quote(pi),quote(7-4))), function(arg)typeof(arg))) R version 3.1.3 (2015-03-09) R version 3.2.0 (2015-04-16) [1,] [1]

Bill, Is your conclusion to just update the code and enforce using the most recent version of R? Dayne On Wed, Jul 15, 2015 at 4:44 PM, Dayne Filer <dayne.filer at gmail.com> wrote: > David, > > If you are referring to the solution that would be: > > rapply(list(test), eval, envir = fenv) > > I thought I explained in the question that the above code does not work.

On Jul 15, 2015, at 12:51 PM, William Dunlap wrote: > I think rapply() was changed to act like lapply() in this respect. > When I looked at the source of the difference, it was that typeof() returned 'language' in 3.2.1, while it returned 'list' in the earlier version of R. The first check in rapply's code in both version was: if (typeof(object) != "list")

Thanks a lot for all the suggestions! Unfortunately, it still gives problems. Most common error message is "ast_realaudio_callback Failed to write frame" after "paying the beep". Then it says "User disconnected". Also, it doesn't react to any extension entered and doesn't do any forwarding (as it should in "exten =>

David, If you are referring to the solution that would be: rapply(list(test), eval, envir = fenv) I thought I explained in the question that the above code does not work. It does not throw an error, but the behavior is no different (at least in the output or result). Using the above code still results in the x object not being stored in fenv on 3.1.2. Dayne On Wed, Jul 15, 2015 at 4:40 PM,

Hi everybody: I configured my Asterisk to register to my VoIP provider, and I can make outgoing calls, but I can't receive any calls with it. I used Ethereal to sniff the activity of it, and I found something that might be causing the problem: When my provider's gateway does the "Request: INVITE mynumber@my-voip-provider.tld ..." my Asterisk asks for "Status: 407 Proxy

I'm having trouble connecting a asterisk server to a SIP Express router. Inbound calls to my asterisk server works just fine, but when i try to make outbound calls I get the following error message: Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to WWW-authenticate on INVITE to '"username" <sip:username@mysipprovider>;tag=as5399a078' I'm

Hello, I'm trying to figure out how to get Asterisk to dial an extension when a call comes from the outside and contains the extension already. (Somebody wants to call a user of Asterisk with extension "111" from the outside) For example: I've hooked Asterisk to sipgate.de and received a landline phone number (say 0781205237). Now if you dial 0781205237 and and an extension

Hi helpers, I've been programming in R for a few months now but I still have doubts about my code - I would like it to be completely S4-compatible. The current code works fine but is probably 'unclean'. I read the interesting article in the last R News and it helped me understand the difference on the whole between S3 and S4 classes, but I need a practical example. Could anyone

Nik, I'm not sure that "NOP" is correct, but I'm in the states so I'll to defer to someone who knows E1/PRI. When I run zttool I have "OK" under the alarms. Is there a way you can call the telco and confirm the settings? Make sure that framing, coding and D channels are set up on their end the same way you're set up. As for asterisk, here's what I get

Okay, I've read as much as I can, and I think i've followed instructions, but I'm still having problems with * and firefly... I can get outgoing to other freshtel working, but not incoming (I get the "not available" voicemail), or outgoing to landline. I'm using the debian asterisk package (0.9.1-RC1-4) My iax.conf has in general (under my FWD register, which

Hello, Anyone knows how can I do this in a cleaner way? mynumber = 1001 as.numeric(unlist(strsplit(as.character(mynumber),""))) [1] 1 0 0 1 Thanks in advance, Gustavo

Hello, after I solved my problem with the fax processing after receiving, I got another problem while sending a fax: the header is not set properly. I use a PHP_Script to upload a PDF file and to generate a call file. A bash script is looking for existent call files in the web directory and moves them the asterisk's outgoing directory. Ok, my call file looks like this: ====== $cf_commands

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found

Hi Guys, I have already tried this one on the developers list. I have not been successful getting much back there and I have notified them that I will post this on the users list instead. Hopefully somebody have tried something similar and can help out. I am developing AGI scripts on Asterisk and have run into some very strange behaviour and I think this is a bug, but I am not completely sure.

Hi, 1) is there a function like "zap destroy channel" to destroy sip channels? Zap/10-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/-081aee08 (pstn-out s 7 ) Ring Dial Zap/g1/0123456789|50|g Zap/8-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/-081aee08 (pstn-out s