similar to: Call transfering external calls to external lines

Hey all, Wondering if this is possible.. Incoming call is answered through X100P, then an extension is dialed using the same X100P card. Basically I want to dial in, enter 9 + <phone#> and have it do a flash then have it dial *08 <the same phone number> + # on the same PSTN line to have it transfer my call to another phone number. I realize this isn't very safe, but I would

i'm new with asterisk. i currently have 1 fxo port. my phone line connected to the fxo is capable of 3 way calls using flash. i'm thinking of a callback and then 3 way call from asterisk. is this possible? 1. phone1 calls asterisk thru zap/fxo. asterisk gets callerid of phone1. 2. asterisk will callback phone1 using zap/fxo. 3. phone1 answers and is prompted for a number to call. 4.

I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF "#", but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __________________________________ Do you

i am using the sample config files and get a dial tone. i have also gotten it to play greetings etc, but i need the phone to ring so that i am not tieing up the one phone line, please help, i know this sounds insanely stupid but i cant get it to work.

Hi How to set a threeway conference in asterisk only for VOIP (I am using only SIP channel). Thanks Nikhil

Hello, I'm trying to set up the flowwing data structure in R: A data frame with 7,000 rows and 4 colums. The rownames have some special meaning (they are names of genes). The 1st column per row is itself a data frame, and columns 2 to 4 will keep numeric values. The data frame contained in the 1st column will have 54 rows (with special names) and 4 colums (1st col is a response, cols 2- 4

Hi, I have a TDM400P (31B) in a PIV 2.8, 512Mb ram, CentOS 4.3, zaptel 1.2.5 and Asterisk 1.2.7.1 and a couple of standard analog phones with a flash button. A hook flash works fine for setting up a 3way call. But pressing the flash button doesn't do anything. The zapata config is below. Anyone have an idea what I'm doing wrong? [channels] context=local usercallerid=yes hidecallerid=no

Hi all - sorry for the off-topic posting - just that the list of consultants on the samba site is horrendously out of date! I thought that this would be the best place for me to find suitable resources... firstly I would like to say thanks a million to all the samba team for your ongoing efforts and great product. I just wish I had more time to experiment with it! We have the startings of a good

Hi all, Some simple questions about codecs: What codec does the Zap channel use by default? Can this default be changed, and to what? (g729 too?) What codec does meetme use? (I think this is ulaw, but asking to be sure) Can you use another codec, or does everything have to be transcoded to ulaw? Finally ... if I have a 3way call going, between 1 g729 caller and two other callers, do I need one

When porting patches from dash git to klibc git, where dash has a different directory structure those switches are handy: Exported with format-patch on dash side and used am as import for klibc side. Signed-off-by: maximilian attems <max at stro.at> --- Documentation/git-am.txt | 5 ++++- git-am.sh | 4 +++- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git

Hi, I was woredering if you could help me to put into practice this solution. The idea: Create a IVR-Voicemail The scene: PSTN------/6------PBX--------/12--------- Internos | /4 ports | IVR-Voicemail The Operation: 1)Where a call enters from the PSTN, the PBX flashes and

hello, i am a bit of a statistical neophyte and currently trying to make some sense of confidence intervals for correlation coefficients. i am using the cor.test() function. the documentation is quite terse and i am having trouble tieing up the output from this function with stuff that i have read in the literature. so, for example, i make two sequences and calculate the correlation coefficient:

Hi, everybody: As part of a paging macro I'm using SendDTMF to send digits to the called party. The section looks like this: exten => s,1,Wait(0.5) exten => s,n,SendDTMF(9531290) exten => s,n,Wait(1.0) exten => s,n,Set(MACRO_RESULT=CONTINUE) To test I direct the call to a live extension just to hear what's happening -- what actually happens is that only the 9 is sent, and

I have been trying to figure a way to SendDTMF into a MeetMe room using the Manager API. I can't redirect everyone into another context and then bring them back because that would mess up my logic. I am trying to use local channels and the originate Action to accomplish this. Exten: 3441115 Priority: 1 ActionID: actid-00000001 Context: senddtmftones Action: Originate Channel:

On Fri, Dec 07, 2007 at 04:30:09PM -0800, Xen patchbot-unstable wrote: > tools: Rationalise library soname versions. > > * Arrange for the sonames of libxenstore, libxc, libfsimage and > libblktap to be set from a single place in Config.mk. Grumble... I don''t like this at all. You just bumped libfsimage for no reason. Can we please fix libfsimage back to the correct

Hi all! I would like to solve some problems: I have a sip provider that lets me make pstn calls after listening some stuff and entering a pin number: 1) How can I make Asterisk enter the pin number? Then wait 1 second and enter the phone number? I have in extensions.conf: exten => 6*,1,Dial,SIP/2002@myprovider,60,tr I have tried with w (like with ZAP channels) but it does not work, nor

Hello all, I am trying to use asterisk to call a local access gateway by dialing a fix number, after getting connected, the is a IVR prompt for pin number and finally the real destination number. I manage to use asterisk to dial to the gateway but have no idea how to send the pin number and destination number. This is due to asterisk only process the next ext only if dial app has terminated. My

Hi everyone: The following code dials our prefix, sends a beep, and sends a DTMF "c" tone, then dials the phone number. I need to send the DTMF only if the phone is answered. [voip] exten=>i,1,NoCDR() exten=>i,2,Hangup() exten=>s,1,Wait(2) exten=>s,2,Background(beep||) exten=>s,3,DigitTimeout(6) exten=>s,4,ResponseTimeout(10) exten=>s,5,SendDTMF(c)

Hi! I am trying to setup a simple queue in Asterisk and I'm having a small problem. Our callers come in through a Bosch PBX and are immediately transferred to an Asterisk menu/IVR. If they select the option to call a SIP phone directly (eg. entering the operator's SIP extension) then the callee/operator can transfer the call to a phone within the Bosch system. What Asterisk does is

I'm not sure why but my call is being ended when I SendDTMF(*). I'm using agi to originate a call and set the context,extension,priority to test,1,1 respectively. I've got the following in my extensions.conf: [test] exten => 1,1,Answer(); same =>n,Wait(5); same =>n,Verbose(1, Sending *); same =>n,SendDTMF(*,500); same =>n,Verbose(1, Sent *); same