similar to: Unable to create channel of type 'Zap' (cause 0)

I know this is not the most appropriated list to this, but I will try: Does anyone know what is the criteria to the generation of the transaction identifiers in MGCP? I mean, are they generated by a randomic method? I'm using Asterisk and MGCP eyeP Phone and observed that the RSIP and NTFY (methods created by the gateway) use high values in the transaction identifiers, while the RQNT, AUEP,

I spoke the other day about my preliminary tests with office-wide paging with Cisco phones using the new SIP 6.1 image which supports auto-answer. I've got a small and crude recipe for those of you who want to experiment and hopefully create some better and more complete examples than the one I've thrown together below. Create a new line on each of the Cisco phones, and put the

Hi, I have the following situation: I've installed Asterisk at Machine 1 (M1 - IP: 192.168.1.145) and X-Lite (X_lite-Xten-Win32-1103m.exe from www.xten.com) at Machine 2 (M2 - IP: 192.168.1.100) and Machine 3 (M3 - IP: 192.168.1.200). I need to catch the SIP and MGCP messages that will appear when M2 calls to M3 and vice versa. The SIP messages are working (I don't have problems with the

Hi, I've just got a TE110P card and installed at Asterisk. I configured zapata.conf, according to www.digium.com/index.php?menu=configuration, but the following error is happening: ... ... ... [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver)

Sirs, According to RFC 2705 (MGCP), these are the parameters that are used in the transactions: ReturnCode, Connection-parameters <-- DeleteConnection(CallId, EndpointId, ConnectionId, [Encapsulated NotificationRequest,] [Encapsulated

Hello list, I'm trying to use zaphfc, the module loads ok, and it identifies the hfc boards in the machine. The problem is: whenever i try to ztcfg -vv I get the following: 8x------------------- Zaptel Configuration ====================== SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual

All the aforementioned techniques need change everytime on the dialplan. I need the office secretary to edit a file (call file) and place it in a particular folder in their windows PCs. this folder is the outgoing folder of LINUX shared through samba in LAN. i need to make it as easy as possible, please. On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist at linuxista.com> wrote:

hi all i m using remote_function , & i need to update two divs at same time (so no :success/:failure).... is there any posiible soln for this thanks rohit --------------------------------- Yahoo! Photos Ring in the New Year with Photo Calendars. Add photos, events, holidays, whatever. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, have anybody a hint /starting point how i can enlarge my code that's posiible for me make any calculaton's with variables in the 64 subset's. Example: I want calculate difference of two variables inside every of the 64 subset data.frames and getting this value in tList! Many thanks, Christian tasign <- paste("tList[[n]] <- try(dtree[dtree$class02 ==

I tried to change the default computers container with the redircmp.exe but it does not work. With this tool from microsoft it should be posiible to redirect computer accounts to a different ou. Does this tool work with samba4 or is there an other why to do this?

I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in extensions.conf changed recently? exten => 5555,1,NoOp(${CALLERID}) hestia*CLI> -- Executing NoOp("SIP/2944093-d24d", ""Cletus the Slaw Jawed Yokel" <2944093>") in new stack == Auto fallthrough, channel 'SIP/2944093-d24d' status is 'UNKNOWN' This

Hello, I searched a while to find out, if there is an sftp-server implementation which provides an option similar to the defaultroot of proftpd. A typical use would be: DefaultRoot = ~ The option does the follwing: Once the use logs in, it determines the home directory of the user .ie /home/u1234 and takes this as the users root. The user cannot escape that root (he can not look at /tmp

Hi All, I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, "enter new password" ok, "re-enter new password" ok, "password has been changed" There are no entries in the mysql.log setting the

Dear R-experts! I have y(x) data, dim(y) == dim(x) == c(2000, 2) I'd like to fit them with nls: fit.result <- nls ( y ~ f(x, p1, p2, p3), start = list(p1 = ... , p2 = .. , p3 = ..) ) Actually I want to fit y[,1] ~ x[,1] and y[,2] ~ x[,2] *simulaneously*, with the same parameters set {p1, p2, p3}. I tried to feed R tha above formula, R errors with:

I've been using * at home for a while now and I'm quite happy with how it works. Having voicemail emailed to me and notify my cell phone via SMS is a great way to impress my friends. :-) The inbound context for my X101P looks something like this: exten => s,1,Dial(SIP/analog1&SIP/analog2,20) exten => s,2,Answer exten => s,3,Voicemail(u1234) exten => s,4,Hangup The

The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox

Dear friends, please excuse me if my question will be trivial. I've installed and started Asterisk (stable 1.0.7, but with CVS HEAD I experienced just the same problem), and changed a bit sip.conf: [general] ; ... dtmfmode = inband disallow = all allow = ulaw allow = alaw allow = gsm run kphone, and call the 1235 extension. According to sample extensions.conf, Asterisk would

> Actually.. I seem to have jumped to improper conclusion.. One thing you will find abour spandsp is that some fax machines will just plain have a problem sending to spandsp, period. Mr. Underwood has localized it to cetain HP fax machines and I can confirm that HP all-in-one SOHO machines exhibit this problem, but I have also noted it with Canons and Panasonics. In my context, I have several

I'm running SVM from e1071 package on a data with ~150 columns (variables) and 50000 lines of data (it takes a bit of time) for radial kernel for different gamma and cost values. I get a very large models with at least 30000 vectors and the prediction I get is not the best one. What does it mean and what could I do to ameliorate my model ? Jerzy Orlowski

Hi All, I'm having trouble setting up a queue: I'm using AgentCallBackLogin to login in the queue, but: 1 - When an agent answer the call and another call arrive his phone rings again. 2 - When no there are no one answer the queue the system goes to voicemail of agent 1234 I'm using asterisk-1.2.0-beta1. My configuration is below, Any ideas? Many thanks, Joao Antunes