<br><font size=2 face="sans-serif">Verify communication between protocols. SIP ou IAX2.</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td>
<br><font size=1 face="sans-serif"><b>Jason Frisch <jfrisch@tsukaeru.net></b></font>
<br><font size=1 face="sans-serif">Enviado Por: asterisk-users-bounces@lists.digium.com</font>
<br><font size=1 face="sans-serif">13/12/05 00:13</font>
<br><font size=1 face="sans-serif">Favor responder a Asterisk Users Mailing List - Non-Commercial Discussion</font>
<br>
<td><font size=1 face="Arial"> </font>
<br><font size=1 face="sans-serif"> Para: asterisk-users@lists.digium.com</font>
<br><font size=1 face="sans-serif"> cc: </font>
<br><font size=1 face="sans-serif"> Assunto: [Asterisk-Users] No outgoing sound...sometimes</font></table>
<p><font size=1 color=#000080 face="sans-serif"> - </font>
<p>
<br>
<br><font size=2 face="Courier New">Hi All,<br>
<br>
I have been having trouble with my asterisk box since last week. It<br>
was going fine until then and I can't remember changing anything..<br>
nothing that I haven't put back anyway.<br>
<br>
The issue is with that about half of the calls received or placed,<br>
the outside party cannot hear my voice; I can hear the<br>
other end fine. I have checked the logs and nothing is different<br>
for the calls that fail. I thought it was the phones, but the messages<br>
played from asterisk<br>
itself also have the same problem.<br>
<br>
The "native bridge" in the below sections seems strange as I though this<br>
was disabled with "canreinvite=no".<br>
<br>
denwa*CLI><br>
-- Executing Goto("SIP/10.129.46.102-0853ec38", "sip|1000|1") in new stack<br>
-- Goto (sip,1000,1)<br>
-- Executing SetVar("SIP/10.129.46.102-0853ec38",<br>
"CALLFILENAME=000-20051213-110514") in new sta<br>
ck<br>
-- Executing GotoIfTime("SIP/10.129.46.102-0853ec38",<br>
"18:00-10:00|mon-fri|*|*?24hour|s|1") in n<br>
ew stack<br>
-- Executing GotoIfTime("SIP/10.129.46.102-0853ec38",<br>
"*|sat-sun|*|*?24hour|s|1") in new stack<br>
-- Executing Dial("SIP/10.129.46.102-0853ec38",<br>
"SIP/2201&SIP/2202|180|tTH") in new stack<br>
-- Called 2201<br>
-- Called 2202<br>
-- SIP/2201-afc3 is ringing<br>
-- SIP/2202-4367 is ringing<br>
-- SIP/2201-afc3 answered SIP/10.129.46.102-0853ec38<br>
-- Attempting native bridge of SIP/10.129.46.102-0853ec38 and SIP/2201-afc3<br>
== Spawn extension (sip, 1000, 4) exited non-zero on<br>
'SIP/10.129.46.102-0853ec38'<br>
<br>
-------------------------<br>
<br>
conf file:<br>
<br>
sip.conf<br>
[general]<br>
port=5060<br>
realm=ocn.ne.jp<br>
context=sip<br>
register=number@ocn.ne.jp:secret:LNRTKR4U@voip-ca35323.ocn.ne.jp/number<br>
disallow=all<br>
allow=ulaw<br>
<br>
[number]<br>
type=friend<br>
host=voip-ca35323.ocn.ne.jp<br>
username=username<br>
secret=secret<br>
fromuser=number<br>
fromdomain=ocn.ne.jp<br>
port=5060<br>
dtmfmode=inband<br>
disallow=all<br>
allow=ulaw<br>
nat=yes<br>
canreinvite=no<br>
context=sip<br>
<br>
[snip]<br>
<br>
If anybody has any idea where I should look, it would be most appreciated.<br>
<br>
Jason<br>
<br>
_______________________________________________<br>
--Bandwidth and Colocation provided by Easynews.com --<br>
<br>
Asterisk-Users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
http://lists.digium.com/mailman/listinfo/asterisk-users<br>
</font>
<br>
<br>