I have 2 asterisk boxes: asterisk-alpha (running 1.0.3) and dev-asterisk
(running latest CVS).
I am the only SIP user on dev, everyone else in the office is on alpha.
If someone dials my extension, it should go IAX to the dev server and the
dev server should ring me.
Here is what I see on the dev machine's console:
-- Accepting AUTHENTICATED call from 192.168.1.25, requested format = 256,
actual format = 256
-- Executing Dial("IAX2/asterisk-alpha@asterisk-alpha/2",
"SIP/3044|30") in
new stack
-- Called 3044
Dec 27 14:14:06 WARNING[9194]: channel.c:2137 ast_channel_make_compatible:
No path to translate from SIP/3044-520a(4) to
IAX2/asterisk-alpha@asterisk-alpha/2(256)
== Spawn extension (all-incomming, 3044, 1) exited non-zero on
'IAX2/asterisk-alpha@asterisk-alpha/2'
-- Hungup 'IAX2/asterisk-alpha@asterisk-alpha/2'
I have the following in sip.conf:
[general]
port = 5060
bindaddr = 192.168.1.26
context = all-incomming
tos=lowdelay
maxexpirey = 3600
defaultexpirey = 120
promiscredir=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[3044]
type=friend
host=dynamic
nat=yes
disallow=all
allow=g729
allow=alaw
allow=ulaw
canreinvite=yes
This seems to be a codec issue but my phone is set for g729 and it appears
that the IAX call is comming in as g729. It seems that asterisk is b0rking
on the fact that I have no G729 licenses installed on the dev box. But that
shouldn't make a difference since asterisk is just passing thru g729. Why
would I need a license to go from IAX-G729 to SIP-G729?
Thanks,
Matthew
