similar to: How to play audio to callee when a fax is detected ? [SOLVED]

Hello, Let say Alice and Bob both have a sip phone connected to the same asterisk 11 box. Alice has T.38 enabled softphone. When Alice sends a fax to Bob extension, the following happens on my system: - Bob phone starts to ring - Bob answers - asterisk sends the incoming call to appropriate fax extension - Bob is hearing nothing at all: no tone, no sound at all. I want to play an audio file

The easiest way is to just run the Dial() command to forward the call to the hard fax without ever Answer()-ing the call. Without an Answer() on the call, Asterisk can't listen for fax detection (because the call hasn't been set up and there is no audio leg yet). Thank you, Noah Engelberth -----Original Message----- From: asterisk-users-bounces at lists.digium.com

2012/1/11, Jos? Pablo M?ndez Soto <auxcri at gmail.com>: > Im using the one that comes with Ubuntu Server 10.10 (0.0.6~pre12-1): > > http://packages.ubuntu.com/search?keywords=libspandsp&searchon=names&suite=maverick&section=all > > And having a sweet time with T.38 gateway. Oneiric already offers latest > pre18. T.38/T.30 gatewaying can tricky enough to

On Sat, Jun 18, 2011 at 6:40 AM, Larry Moore <lmoore at starwon.com.au> wrote: > On 18/06/2011 5:36 AM, Matteo Campana wrote: > >> >> Inviato da iPhone >> >> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling<EWieling at nyigc.com> >> ha scritto: >> >> We experience the same thing. The solution we use is to not change >>>

Hi! Please is it possible for Cisco 5XX phones to use asterisk/FreePBX phone directories, and if so, how? Thanks in advance! On Feb 17, 2013 6:40 PM, <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

I solved the issue by not answering the call as I assume others have done. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Wednesday, February 18, 2015 12:50 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Res_fax - FAXOPT(faxdetect) Hello Le

Hi, Is there a way to "announce" a call to the callee? For instance, I've answered an incoming call, collected some info and now want to ring an extension, and make an accouncement to that extension before connecting the caller through. Thanks, Steve

thank you but could you please tell me how can i put it thanks and regards 2015-03-12 18:19 GMT+00:00 Administrator TOOTAI <admin at tootai.net>: > Hi, > > Le 12/03/2015 17:28, Salaheddine Elharit a ?crit : > >> hello list, >> >> i use the code below >> >> [macro-chanspy] >> exten => s,1,Authenticate(${ARG1}) >> exten =>

Hi, I still have the same problem trying to configure ITSP failover in extensions.conf for a connected PRI. Any comments thoughts or direction would be greatly appreciated. I sympathize with wanting inbound DID failover. If we have a client with multiple DIDs we will spread them across two or three ITSPs so that all inbound connectivity will not be lost if one of them has an issue. I

Hmm the calls are made during the day (and sometimes very early in the morning). Right now it looks like someone actually made these calls. If that is the case it's somewhat comforting to know the system wasn't compromised. However, the $25,000 phone bill still remains. Yikes. $6.25 per minute to Cambodia seems quite steep to me. On Wed, Jan 28, 2015 at 6:07 PM, Duncan Turnbull <duncan

On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <admin at tootai.net> wrote: > Le 18/03/2016 16:20, Trey Hilyard a ?crit : > > I am trying to set up my Asterisk server so that it will recognize an > > incoming call to the Asterisk's own Location Routing Number (LRN), > > validating the "rn" in the INVITE and then using the Called Number from >

Do you have DISA setup? We're seeing lots of attackers running scripts that send digits until they strike a DISA, misconfigured mailbox, etc. (Assuming it wasn't a stupid employee forwarding an inbound call to a 9xxxxxxx number etc). Have a look at SecAst (www.generationd.com) - it detects callers sending too many digits, monitors digit dialing speeds, etc. to help identify and block

2013/2/12 Doug Lytle <support at drdos.info> > >> non-standard locations such as /usr/local/sbin > > If compiling from source, it'd normally be specified by the --prefix > option: > > ./configure --prefix=/usr/local > > Doug > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary >

2013/9/19 Matthew Jordan <mjordan at digium.com> > > On Thu, Sep 19, 2013 at 9:02 AM, Olivier <oza_4h07 at yahoo.fr> wrote: > >> Hi, >> >> Asterisk 11 doc says CDR(src) value is read-only (see >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR). >> >> For various reasons, I would appreciate to change its value so that it

2012/1/10, Olivier <oza_4h07 at yahoo.fr>: > Hi, > > 1. This patch didn't correct the issue but I'm far from certain that I > correctly applied the patch. I was right to suspect I was wrong : now, after correctly applying the DAHLIN-275 patch, it's working OK (with the EchoCan module plugged-in). Thanks for your lighting fast correction !! > 2. I took the

Hi, I'm configuring a system so that end user can receive phone and calls using the same extension and DID. At the moment, fax are correctly detected but I'm trying to improve end user experience. Relevant dialplan (from extensions.ael) is : fax => { Verbose(0,Incoming fax from ${CALLERID(num)});

OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > Its really weird working with OpenSuse. I am not sure how others are using > with OpenSuse. Through Yast also I tried to install Asterisk package, it > didn't find. > > Now I am clueless to work with OpenSuse. > >

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n

penSuse 12.1. Lets check with OpenSuse 12.1.&nbsp; <div><br /> </div> <div>Regards.</div> <div><br /> <br /> <div class=3D"gmail_quote">On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan = N <span dir=3D"ltr">&lt;<a href=3D"mailto:gopalakrishnan.an at gmail.com" targ=

Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With SIP 3.2.X firmware (available on the Polycom download site) and Asterisk 1.6.1, Polycom phones now support a full featured BLF showing statuses of Ringing, Inuse and Online and one touch directed call pickup. On the asterisk side all that needs to be done is to add a hint to the extension and enable directed pickup.