Hi,
I asked some questions here about G.729 earlier in the week, and it
looks like it would fit the bill for compressing audio between my *
server in colocation and sip phone at home.
This is what I want my setup to look like.
(Wont make sense unless you are using a fixed width font)
[my phone] [asterisk] [third parties]
Snom 360 <----------> v 1.4 <-------------> ???
SIP IAX/SIP
G.729 Don't care (probably something
other than G.729, my preferred
supplier today likes ulaw and
alaw)
My phone sees the * box over a relatively slow consumer connectivity
link. The * box is colocated and has excellent connectivity.
Therefore the tighter compression between * and my phone is
important, hence why I want to use g.729 here.
The config for my phone, and my preferred voice supplier looks like
this :
[[[from sip.conf]]]
[andydesk]
type=friend
context=default
secret=xxx
host=dynamic
dtmfmode=rfc2833
username=andydesk
mailbox=1001
vmexten=500
disallow=all
allow=g729
allow=alaw
allow=ulaw
allow=gsm
regexten=1001
allowreinvite=no
[[[from iax.conf]]]
[thing]
type=friend
host=dynamic
username=thing
secret=xxx
trunking=off
bridging=on
context=thing
disallow=all
allow=ulaw
allow=alaw
allow=gsm
When I place a call, the other party's line rings as normal. When
the other party answers, I get a sip 'denied' packet, and the call is
aborted. Asterisk says : No path to translate from SIP/mydeskphone
to IAX/myprovider and Had to drop call because I couldn't make SIP/
mydeskphone ompatible with IAX/myprovider.
This looks similar to this bug :
http://bugs.digium.com/view.php?id=8781&nbn=4
What I would expect to happen, is that Asterisk would transcode
between the ulaw/alaw party, and me, wanting to listen via g729. Is
this what *should* happen ? Worth noting that my provider does not
support G.729. Is what is happening a bug ? Any patches I can try
to see if they work ? Or is it my config which is broken ?
Inbound calls work ok, I guess this is because they are presented as
alaw and asterisk is just passing them through (which of course isn't
what i really want).
Thanks for any suggestions,
Andy
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
