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Could you send the trace?,
just guessing, but in the 53xx, your carrier could fix the
dial peer just to use g729, and give you a busy for anything
else.
Leandro,
I don�t have the licenses on porpuose, I shouldn't
need'em cause the phone I'm using has only g729 enabled, the oh323 has only
g729a enabled and the Cisco (over which I have no control at all) should
be able to manage g729... furthermore reading the trace from the oh323 (debug
level 5) I never found any reference to ulaw... just tenths of references to rtp
audio being sent in g729 and then suddenly a:
51:45.239
ClearCallT...d:b092b248 H323 Clearing connection
ip$localhost/32126 reason=EndedByLocalUser
on the second case, the phone is (99% sure) not
busy (it's my desk phone) and if I call using g729 I get it to ring the phone
and when I pick up I get the error, if I call using ulaw or alaw it doesn�t even
ring I get the busy msg and that's it... but what is strange is that it can't be
wrong rules in the 53xx cause when using g729 everything works fine till I
answer the call... so the dialing logic is ok, of course I dial the same number
when in g729 and g711...
any more clues?
thanks again.
M.
----- Original Message -----
Sent: Friday, June 03, 2005 3:32 PM
Subject: RE: [Asterisk-Users] oh-323 /
Cisco AS5300 problem
The first error is probably because you don�t have
licenses for 729 and you are trancoding the audio.
The second is well dialed and you get from the 5300 a
busy message, the reason could be user busy (as the message saids), wrong dial
peer config, wrong dialing rules in 53xx.
LTenorio
Hi i'm trying to connect to the PSTN in the
following way
sip ATA -> * -> gnugk -> Cisco AS5300
-> PSTN
I'm using asterisk CVS-HEAD-06/01/05-14:33:15
running over RH EL3
Asterisk-Oh323 0.7.2 pre1
Open H323 v1.13.5
pwlib v1.6.6
and I'm having a lot of trouble, gnugk and * both
have public ips and are not behind any type of firewall, the sip ATA is behind
a firewall and has a private ip but there's a stun server working and the
connection works fine.
If i try to dial out from the sip ATA through the
GK (gnugk) configuring g729 in the sip account and the oh323.conf I
get an error about trying to use ulaw:
-- Executing Dial("SIP/D0103-3a43",
"OH323/80152392522|50")
-- H.323 call to 80152392522
with codec(s) g729
-- Called
80152392522
-- OH323/80152392522-ef38 is
ringing
Jun 3 10:09:05 NOTICE[27744]: channel.c:1873 set_format:
Unable to find
a path from ulaw to g729
Jun 3 10:09:05
NOTICE[27744]: channel.c:1873 set_format: Unable to find
a path from ulaw
to g729
OH323/80152392522-ef38: Format changed to ulaw (native
g729).
Jun 3 10:09:05 NOTICE[27744]: channel.c:1873 set_format:
Unable to find
a path from g729 to ulaw
Jun 3 10:09:05
WARNING[27744]: app_dial.c:583 wait_for_answer: Unable
to forward
voice
-- Hungup 'OH323/80152392522-ef38'
== No
one is available to answer at this time (1:0/0/0)
--
H.323 call 'ip$localhost/32125' cleared, reason 1 (Cleared by
local
user)
but if i try g711 ulaw on both the
ata and the oh323 i get the following error
-- Executing Dial("SIP/D0103-d16b",
"OH323/00541152392522|50")
-- H.323 call to
00541152392522 with codec(s) ulaw
-- Called
00541152392522
-- H.323 call 'ip$localhost/26583'
cleared, reason 24 (Call ended
with Q.931 cause)
--
OH323/00541152392522-d872 is busy
-- Hungup
'OH323/00541152392522-d872'
== Everyone is busy/congested at this
time (1:1/0/0)
I found a message about this error from
January but there is no follow up
any clues?
thanks a lot in advance.
Matias
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