Hi,

Freepbx includes a fax_process.pl which convert TIF files into PDF
files before sending by email.

I’m used to use sSMTP with Asterisk.
I’m certain ssmtp is correctly configured in my (Debian Squeeze) setup
as I’m correctly receiving voicemails in email box.

Is it possible to tell fax_process.pl to use ssmtp when sending emails out ?
If positive, any hint on how to configure this ?
If negative, which smtp software shall I replace ssmtp with ?

Regards

Hello,

I can do simple, “yum install asterisk18-*” and it installs Asterisk and
Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and
smooth.

However, if I want to compile dahdi-linux on the same openvz then I get the
error, *”You do not appear to have the source for the 2.6.32-4-pve kernel
installed”.*
*
*
1- Based on above error and Google search I have concluded that dahdi-linux
module should be installed on mother node. So, I am puzzled. How does
Digium yum repository achive this without acessing the mother node?

2- Do I even need Dahdi, if the server doesn’t connect to PSTN at all and
it’s all SIP? If yes, what do I need it for?

Any feedback is much appreciated.

Thanks

Is it possible to make Asterisk jump into action and play a sound file as soon as a handset is lifted, instead of providing a Dial Tone and waiting for the user to dial an extension.

With analog phones (chan_dahdi) – you just have to set ‘immediate = yes’ in chan_dahdi.conf , with a SIP phone: that’s something to configure the handset for, as it only sends out a call once you “dialed”.

Thanks to: Tzafrir Cohen

Hi all,
please help me.
how we can configure between call add the IVR.
My scenarios is
“A” get the incomming call from “C”.In between them I need to one side IVR
play for “C”, “C” enter the some DTMF inputs and “A” should be on hold.
after finish “C” input will complete again they want talk each other .This
is the scenario.

Can anybody help to me how can I add this IVR in between those call….,
and how my asterisk will detect the DTMF input….

Best Regards,
Mahesh Katta

Hi,

A calls B and B has it’s phone forwarded to C. So the call rings at C.
Is there any way to inform A about that forwarding? Best way would be
to update the called name so A has “B forwarded to C” in his display.
Any chance to get this?
I tried “Set(REDIRECTING(to-name)=…)”. This sends a “SIP/2.0 181
Call is being forwarded” to the calling phone, but with no information
about the new callee name.

Regards,
Gunnar