similar to: accepting a call, macros, and key presses.

Hello, What i am trying to do is ..... Dail a number and ask if you wana talk to XXX press 1 and if you dont wana talk press any other key. For this purpose i am using this link<http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial> . *I am using this option :- * *M(**x**)*: Executes the macro (x) upon connect of the call (i.e. when the called party answers). IMPORTANT - The CDR

I have a caller screen queue setup. Basically a caller calls in, goes through a IVR, and before that caller is put into the queue, they get a sub ran on them first asking for them to say there name. That gets saved and they are entered into the queue using Queue(mainqueue,,,,300). In the queues.conf i have a list of members these are local/extension at external-default, there are two

Hello all, Could I create a script to delete the first messages on my voice mail? In this script should I update any "messages index file" or there isn't any file to index them? Could you share any script to do that? Thanks in advance. VoipCrazy. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Yo, Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate call divert-feature. This one validates if the extension a call-forward is to be set to is actually valid for the current context and additionally saves this context into the DB and always uses it to originate the divert from, as you can't expect the

Hi, Samba team! I am trying to install samba4 on FreeBSD 9.2 as a domain DC to join an existing samba4 domain controller on FreeBSD 9.2. I followed the instruction of: Samba4/HOWTO/Join a domain as a DC Everything is OK until I run the following command: root at mtm:/var/named/etc/namedb # samba-tool drs showrepl Default-First-Site-Name\MTM DSA Options: 0x00000001 DSA object GUID:

Hi Stefan, Thank you very much for pointing me to the wordspace package. It does the job a bit faster than my C code but is 100 times more convenient. By the way, since the tcrossprod function in the Matrix package is so fast, the Euclidean distance can be computed very fast: euc_dist <- function(m) {mtm <- Matrix::tcrossprod(m); sq <- rowSums(m*m);? sqrt(outer(sq,sq,"+") -

See comments below. Henry Devito wrote: > Here's the problem. When I call 555 to login, it asks for the agent ID > which I enter as 501, it asks for the password which I enter as 1234, > then it asks for the extension I dial 501 It then says that extension is > not valid. What am I missing? Of course 501 is valid I can make and > take calls from it now. > > >

Hello, I've seen this already asked and answered but it is still a no go for me. I'm trying to do some preprocessing in the middle of a call, before bridging. I've seen two choices: M() and G() parameters of the Dial() command. G() was discarded because I don't know if it is possible to bridge channels after processing. With M() I've done something like that: macro

Hi all, In sip.conf: [general] regcontext = autoreg [devabc] regexten = 543 creates "exten=> 543,1,Noop(devabc)" in context autoreg when devabc registers. But I can't use "exten=> _5XX,2,Dial(SIP/${EXTEN})" in the dialplan, because there's no device SIP/543. Now I know I can add a line like "exten=> 543,2,Dial(SIP/devabc)" for each and

Hello, I have a problem with asterisk when dial out to a SIP provider. Asterisk send a INVITE with no credentials, the provider reply with a 401 Unauthorized. However, Asterisk DOES NOT resend the invite again with credentials. But it hangs there (maybe waiting for a ok) It is this a bug in asterisk or the provider is supposed to send something else rather than a 401 as answer for a INVITE ?

I have verified this on two machines, but it would be helpful if others out there can reproduce it too. Also, I do not know if it is Xorg or the FreeBSD keyboard drivers, since I see no way to reproduce on the console (i.e. turn off repeat). In an xterm, type: "xset r off". Then try some multiple-key combinations (i.e. keep holding first key(s) when you type the next one): po (o does

The subject says it all. I'm betting there's a way to do it, but so far I haven't found the dialplan runestone via web searching. Thanks. b.

If '#' is pressed during a call the CDR that is written at the end of the call contains '#' in the dst / destination field rather than the number that was originally called. How do I avoid losing that original number so that I can use the CDR for billing? I've tried not having a '#' target in extensions.conf and I've tried calling ResetCDR(w) in the

In our IVR we have a user enter a 6 digit number and information if returned. Our problem is that no all of the digits that the user presses are being recevied correctly. Its not as if the first digits are being cut off or the last, its just some digits aren't coming through. Our setup is that we have an IAX channel coming in from VoicePulse going to our Asterisk box via the T1 coming in to

Hi We have a few conference numbers and all use MeetMe using the D option. We have noticed sometimes that the server is picking up more key presses than were actually done, i.e. the user presses 1234 for the pin and in the logs we see something like Created MeetMe conference 1022 for conference '12234' or Created MeetMe conference 1022 for conference '112334' Has anyone else

Hi, Does this report ring a bell here? I didn't check, but the image should carry an isohybridized 4.02 version of isolinux. The working (lenny) version is 3.71. Thanks, Feri. -------------------- Start of forwarded message -------------------- Subject: Bug#604245: Syslinux fails Date: Sun, 21 Nov 2010 14:13:41 +0100 From: Gyorgy Jeney <nog.lkml at gmail.com> To: submit at

Hi people! I need a help. I connect with Asterisk Manager and execute an Originate Action that asks for Asterisk to call for a number 09194**** and to transfer to peer 101. This command enters in my dial plan extension that I made exclusively for tests. The problem is: If I do the call through ORIGINATE Action, the Monitor() just makes the record file, but this is empty, if I do

As Jim has indicated, memory usage problems can require very specific diagnostics and code changes, so generic help is tough to give. However, in most cases I have found the dplyr package to be more memory efficient than plyr, so you could consider that. Also, you can be explicit about only saving the minimum results you want to keep rather than making a list of complete results and extracting

Hi We've been running asterisk 1.4.17 (deb package) in a production environment for some while now and are finally taken the plunge to update to 1.4.38 (Ubuntu servers). All of this is using the RealTime Architecture I have upgraded the asterisk version in one of our test environments and blind transferring seems to have suddenly stopped working. It was working fine under 1.4.17 So, call

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>