Hi

I was asked by our development departement to setup asterisk in a
manner that if someone calls an extension in the department that was
was only configured, but a handset was never attached to it to fall
back to a default extension. For example: Someone calls extension
2408, but there’s no phone attached to 2408 it should fall back and
ring at 2400..

How do I setup asterisk to find out if there’s a phone attached to an
internal number if not ring another extension?

TIA
Paolo

Hi all,

I’ve done a basic install of 10.1.2 to have a play with the new
ConfBridge application and have noticed high latency when in a
conference. It’s to the order of 900ms or so which is just too much for
a conference to work well.

I can account for about 120ms of that latency, but not the rest.

With two SIP handsets in the conference, worst case latency from handset
to handset (Handset –> Local Asterisk –> Gateway Asterisk –> PSTN –>
Gateway Asterisk –> Conference Asterisk –> Gateway Asterisk –> PSTN

I’ve got some users reporting an odd problem.

Once in a while, their Polycom phones ring and they are unable to answer
them… any of them.

When they pick up the handset, all of the phones continue to ring. Same thing
happens if they grab a different phone. They aren’t reporting any out-bound
issues.

I don’t see anything in the logs, and the phones remain registered.

They can’t reproduce the symptoms on demand, but it seems like it’s happening
more frequently, lately.

Any ideas?

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,

On a brand new system, I met an issue I’ve never met before.

My setup is :
debian 6.0.3
asterisk 1.8.8.1
dahdi 2.6.0
libpri 1.4.12
freepbx 2.9.0.4
TE420FB (with hardware EC)

This is the very first time I’m using Freepbx and the whole
configuration was first generated by a “make samples” command.

When a SIP hardphone dials another SIP hardphone, everything is OK.
When the same SIP hardphone dials an external number, there is a quite
loud noise that can be heard within the handset from the moment the
digit sequence have been sent to the moment any party hangs up. The
other party doesn’t hear this noise.
(It’s not easy for me to describe this noise)

I don’t have any idea of where to look at.
What would you suggest or recommend ?

Regards