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<DIV dir=ltr align=left><SPAN class=392203018-03062005><FONT face=Arial
color=#0000ff size=2>The first error is probably because you donīt have licenses
for 729 and you are trancoding the audio.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=392203018-03062005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=392203018-03062005><FONT face=Arial
color=#0000ff size=2>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.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=392203018-03062005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=392203018-03062005><FONT face=Arial
color=#0000ff size=2>LTenorio</FONT></SPAN></DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Matias
G.<BR><B>Sent:</B> Friday, June 03, 2005 1:02 PM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> [Asterisk-Users] oh-323 /
Cisco AS5300 problem<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><FONT face=Arial size=2>Hi i'm trying to connect to the PSTN in the
following way</FONT></DIV>
<DIV><FONT face=Arial size=2>sip ATA -> * -> gnugk -> Cisco AS5300
-> PSTN</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I'm using asterisk CVS-HEAD-06/01/05-14:33:15
running over RH EL3 </FONT></DIV>
<DIV><FONT face=Arial size=2>Asterisk-Oh323 0.7.2 pre1</FONT></DIV>
<DIV><FONT face=Arial size=2>Open H323 v1.13.5</FONT></DIV>
<DIV><FONT face=Arial size=2>pwlib v1.6.6</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>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.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>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:</FONT></DIV>
<DIV> -- Executing Dial("SIP/D0103-3a43",
"OH323/80152392522|50")<BR> -- H.323 call to 80152392522 with
codec(s) g729<BR> -- Called 80152392522<BR>
-- OH323/80152392522-ef38 is ringing<BR>Jun 3 10:09:05 NOTICE[27744]:
channel.c:1873 set_format: Unable to find <BR>a path from ulaw to
g729<BR>Jun 3 10:09:05 NOTICE[27744]: channel.c:1873 set_format: Unable to
find <BR>a path from ulaw to g729<BR>OH323/80152392522-ef38: Format changed to
ulaw (native g729).<BR>Jun 3 10:09:05 NOTICE[27744]: channel.c:1873
set_format: Unable to find <BR>a path from g729 to ulaw<BR>Jun 3 10:09:05
WARNING[27744]: app_dial.c:583 wait_for_answer: Unable <BR>to forward
voice<BR> -- Hungup 'OH323/80152392522-ef38'<BR> == No
one is available to answer at this time (1:0/0/0)<BR> -- H.323
call 'ip$localhost/32125' cleared, reason 1 (Cleared by <BR>local
user)<BR><BR><FONT face=Arial size=2>but if i try g711 ulaw on both the ata
and the oh323 i get the following error</FONT></DIV>
<DIV> -- Executing Dial("SIP/D0103-d16b",
"OH323/00541152392522|50")<BR> -- H.323 call to 00541152392522
with codec(s) ulaw<BR> -- Called
00541152392522<BR> -- H.323 call 'ip$localhost/26583' cleared,
reason 24 (Call ended <BR>with Q.931 cause)<BR> --
OH323/00541152392522-d872 is busy<BR> -- Hungup
'OH323/00541152392522-d872'<BR> == Everyone is busy/congested at this time
(1:1/0/0)<BR></DIV>
<DIV><FONT face=Arial size=2>I found a message about this error from
January but there is no follow up</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>any clues? </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>thanks a lot in advance.</FONT></DIV>
<DIV><FONT face=Arial size=2>Matias</FONT></DIV></BODY></HTML>