it is probably not what you are looking for.
but simply use a conference room of asterisk for those 1 line phones.
pamela
Rilawich Ango wrote:
> That's easy if phone supports 3 ways call. However, phones in my
> company only have 1 line without join function. Is it possible to
> implement 3 way
Hello group,
I've encountered some problems getting Beronets app_bundle installed
with Asterisk 1.4.0-beta2.
The output messages I get while running make are:
cc -ggdb -Wall -D_GNU_SOURCE -fPIC
-DAST_CONFIG_DIR=\"/usr/local/asterisk/etc/asterisk/\"
-DPTYSPOOLDIR=\"/usr/local/asterisk/var/spo
hello,
there are two options to do this:
1. if you retrieve your voicemails via your phone you will played back
some option like changing your busy and unavailable message (after
dialing 0 - just follow the instructions).
or
2. you just record your soundfiles with your favourite recorder and
ch
hello,
i wanted to use g729 in asterisk (iax to sip) as passthrough. Has anyone
got experience with configuring this or does someone know if this is
possible at all?
At the moment asterisk2 is always transcoding to alaw but but this
results in horrible voice quality.
phone1 behind NAT (sip) ->
Hello group,
I've a rather strange problem with my Snom200 telephone.
I'm using it in combination with SER, asterisk and rtpproxy.
The telephone is behind NAT and connects to SER. It can be called
without any problem from any Client on asterisk or SER.
But whenever I make a call to asterisk or oth
Hi Kevin,
no you didn't miss the reply and I've not resolved it yet.
Have you got similar problems?
Pamela
Kevin Fjelsted wrote:
Pamela,
Did you resolve the problems you described?
I didn't see a reply on the list but I may have missed it.
-Kevin
-Original Message-----
Fr
I also have another question to asterisk and NAT:
o) If one asterisk machine and the telephones are behind NAT, do I need
a proxy to get the speech through, or should asterisk work this out on
its own?
Any help with my problem will be greatly appreciated. Thanks in advance.
Pamela Weis
__
