similar to: AbsoluteTimeout Inside A Macro

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ

This probably came up before, but I have a faxing question for everyone. I have a simple extension setup to use rxfax to receive faxes sent to asterisk. It is: exten => s,1,Answer() exten => s,n,AbsoluteTimeout(300) exten => s,n,Set(FAXFILE=/var/spool/asterisk/fax/${ARG1}_${CALLERIDNUM}_${UNIQUEID}.tif) exten => s,n,rxfax(${FAXFILE}) exten => s,n,System(/usr/bin/mailfax

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

Is AbsoluteTimeout designed to work with canreinvite=yes? If not, are the any other options for disconnecting a call after a predefined duration when using canreinvite=yes? Thanks! David

I have no idea what this really should be called, so for lack of a better name, I called it "reverse hold". Hopefully someone else can make use of it, or even make it better, as its the first thing of its kind I've made for asterisk. Like most people, I'm very busy, so when I call other companies, sitting on hold really sucks. If you have speaker phone, its not so bad, but then

I'm using the M() option to Dial() and having problems. In the following dialplan example ANY digit exits the macro. When the callee presses 1 the Noop(Reset AbsoluteTimeout(0)) does not get run. Does anyone have any ideas as to what I'm doing wrong? Asterisk 1.2.x [extensions] exten => 2998,1,Dial(Zap/1/5551212,,wM(answer-confirmation^20)) [macro-answer-confirmation] exten

The attendant attempts an attended call transfer (all phones are IP501). The attendant pushes "hold", "transfer", dials the extension and announces the call. When the attendant pushes "transfer" the second time, the original call is lost. The reason this is a big problem is that the PRI channel for the call remains busy. Subsequent inbound calls on that

A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers. The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the ?more? soft key, then the ?Transfer? soft key, then enter the extension number they want to transfer to, and hit the

I just discovered an 18 hour call to Brazil that was 60 seconds of an employee calling a customer, then 18 hours and 47 minutes of background noise in their office. The Cisco 7960's have an issue where you sometime don't realize the phone is still off hook as was the case for this call. I'd like to use some options like AbsoluteTimeout, which would jump in the middle of a call every

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile

Hi Steve, > .. I'll try to sort all this out, and then I'll attack this > problem. Hopefully, I get it all into svn before the next release of > 1.4...! Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling. I for one

Dear All, I have a problem with call transfer. When I dial a number and then I want to transfer current call to an extension, I'm getting disconnected. Transfering incoming call works fine. I'm using macro for dialing. extensions.conf: [from-internal] ignorepat => 9 exten => 200,1,Macro(stdexten,200,SIP/dzalewski) [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})})

Does anyone know if AbsoluteTimeout is working completely? As far as I can see on my systems, I'm still getting occasional hung SIP channels, even though there should be nothing over my setting... cheers Sherwood -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050804/31a0dac0/attachment.htm

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP

I am using Asterisk 1.06 Stable. When I dial my Mobile Number to check Voice Mail or my Bank Account Phone Access Number, the IVR System on the other end asks me to enter *2378 to transfer to an attendant. But When I press *2378, Asterisk tells me that it cannot transfer the calls and gives an error on CLI saying Extension '' does not exist in the dial plan. What is the trick to make

Skipped content of type multipart/alternative-------------- next part -------------- Index: pbx.c =================================================================== RCS file: /usr/cvsroot/asterisk/pbx.c,v retrieving revision 1.14 diff -u -r1.14 pbx.c --- pbx.c 19 Apr 2003 02:41:22 -0000 1.14 +++ pbx.c 21 Apr 2003 02:27:43 -0000 @@ -713,6 +713,8 @@ { char *first,*second; char tmpvar[80] =

Hi All, I've been having a hard time getting the AbsoluteTimeout function to work. Is this Function working in for SIP? I've search all the messages in the news letters and tried what was suggested and still have not gotten it to work. Below is a portion of my extensions.conf. I've also been running these test on ver 0.5.0 exten => _X.,1,Absolutetimeout(20) exten =>

Greetings list, I've been playing around this afternoon with Flash Operator Panel, trying to get it to do attended transfers. I am running the latest version. Has anyone managed to get this working reliably, and if so, would you mind sharing how you did it please? Alternatively, are there any other GUIs (free or commercial) that reliably support attended transfers? I'm trying to