similar to: Channels Idle Status Ring // cdr entries

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")

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,

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

Hi Folks, using http://wiki.samba.org/index.php/Windows7 and trying to join I get this Error Message: "More data available" with no log entries on the smb side. It does not seem that the Windows machine talks to samba. Mapping shares work well insteed. Can anybody help? Thanks and best regards Ralf

Le 17/01/2020 à 11:54, Administrator a écrit : > > Le 15/01/2020 à 19:24, Administrator a écrit : >> Hi all, >> >> we face a strange behavior while connecting an Asterisk16 instance >> with PJSIP to 2 providers: we receive error 401 Unauthorized, both of >> them having Kamailio as front-end. With other providers -we don't >> know if they run

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

In 3.1.2 eval does not store the result of the bquote-generated call in the given environment. Interestingly, in 3.2.1 eval does store the result of the bquote-generated call in the given environment. In other words if I run the given example with eval rather than evalq, on 3.1.2 "x" is never stored in "fenv," but it is when I run the same code on 3.2.1. However, the given

Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context

I am trying to enable dial-by-email by using LDAPget to query an Active Directory server. I've got it retrieving the phone number fine. Unforunately, the numbers stored in active directory are either in the format: (xxx) xxx-xxxx or xxx-xxx-xxxx. Is there any way to parse characters out of the dialed phone number so that I only end up with digits (remove spaces, parenthesis and dashes)?

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

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",

Hi, ? I am trying to store queues.conf to a MySQL database using dynamic realtime. I have a working ODBC connection and the queueing system already works but I want to store the queues.conf file to a database. I am following the guide from Asterisk the definitive guide, the ebook can be found at: http://ofps.oreilly.com/titles/9781449332426/asterisk-DB.html ? I have a database called asterisk

Since Thursday evening my asterisk box has been failing to register with broadvoice. I haven't changed any of my config files in the last week. Can anyone suggest anything? Asterisk is reporting: *CLI> Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout: Registration for '703XXXXXXX@147.135.8.129' timed out, trying again -- Got SIP response 404 "Not found"

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

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 =>

Checkout http://www.voip-info.org/wiki-Asterisk+variables I believe that should have the answer for you. furthermore assuming that your number is always going to be 12 digits. exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number. Hope this helps. Umar On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote: > Hi, > > this

Hi, I have a case where SIP channels will not be destroyed, resulting in further calls to ChanIsAvail() to fail. The process (I believe) to replicate this is the following: - Make a call to another SIP phone that is an "intercom" call (Auto-Answer) - For whatever reason, the phone happens to go UNREACHABLE during this call - Phone comes back REACHABLE, but channel still exists in

Hi! Im looking for a very basic example for the following simple problem. I've been searching voip-info.org and looked in the ORA book without a clue. I have a SIP account at sip.provider.com and my own asterisk server. What I want is the following: I. Register my phone to my asterisk server, not directly to provider.com II. My asterisk server should ring my phone when somebody calls me

I've searched WIKI and Archives but nothing. Im getting: -- Called username:password@iaxtel.com/1800somenumber@iaxtel Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call rejected by 69.73.19.178: Unable to negotiate codec -- Hungup 'IAX2[Iaxtel]/8' == No one is available to answer at this time -- Executing Hangup("SIP/104-b8eb", "")

Dear list, today I installed a new asterisk machine, bound to replace my current pbx. I am using a Fritz ISDN card; on the old machine with the drivers coming together with the super-old rpm asterisk installation of SUSE 9.2. The new machine is finally on asterisk 1.0.9, with chan_capi 0.5.4; now I am doing a nightly test. Apparently I can receive calls, but I can't dial out. I seem to