Hi!
Are the GIPS codecs now implemented with the Asterisk?
If I need more analog lines, say around 30, what's the
easiest way doing it? I checked the Mediatrix box with
24 connections, maybe that would be a good (and rel. cheap)
way to go? Any other suggestions? The ports has to
support fax
Hi
Isn't this exactly what we _don't_ wanna do?! =) I suppose
TDM and VoIP is supposed to interconnect not to be
separated.
i think it's nice with a busy list, it means some real hot
stuff is happening, and that's good!
rgds
/staffan
-Ursprungligt meddelande-
Från: Luciano Ramos
I have an interesting problem now. I use asterisk to connect
to both FWD and a sip provider here in sweden. suddenly, (i know
my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try
to make a call using this provider. FWD still works fine, and I can call directly
Ok
Yea, cause I used both Kphone and Windows messenger, and they
successfully registered (and subscribed i think) towards asterisk. Using
Kphone I even get a online status on all other users on the asterisk but
no interaction with status or IM. So maybe there is some quasi presence
avaible? I
Hi
Is there any work being done on implementing IM/SIMPLE support
for SIP on Asterisk? Like a presence server?
rdgs,
/Staffan Kerker
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Hi
Is it possible (or recommended) to run both Asterisk and
say SER on the same physical machine? How about port conflicts?
Maybe the easiest way is to change the default SIP port on Asterisk?
But how will that work if I register some SIP accounts directly
from asterisk (like my SIP provider)
Hi
I know this is a bit off topic, but still pretty interesting.
I'm running Asterisk on my Linux router/NAT/FW connected via
cable (1mbit/200kbit) to the internet.
Now, I wanna do local QoS implementation. Just very simple to
give RTP (UDP) highest priority on my outbound interface. So,
Hi!
Does * do DNS-lookups when outgoing calls are placed to a different
SIP domain? Can I call from sip:[EMAIL PROTECTED] to sip:[EMAIL PROTECTED]?
Can * work as a regular SIP proxy in that aspect?
Can * handle SIP URI:s that are complete SIP URI:s (sip:[EMAIL PROTECTED]) instead
of numbers only?
Hi
I'm running exactly the same setup. Asterisk is running on my FW/NAT/Router
with two interfaces. My local phones are situated behind the NAT and connects
to the outer interface of the */FW/NAT/Router. * is then connected to my
SIP providers (since I'm only using the SIP-part of *, PSTN
