Verify communication between protocols. SIP ou IAX2.
Jason Frisch [EMAIL PROTECTED]
Enviado Por: [EMAIL PROTECTED]
13/12/05 00:13
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Para:asterisk-users@lists.digium.com
cc:
Assunto:[Asterisk-Users] No outgoing sound...sometimes
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Hi All,
I have been having trouble with my asterisk box since last week. It
was going fine until then and I can't remember changing anything..
nothing that I haven't put back anyway.
The issue is with that about half of the calls received or placed,
the outside party cannot hear my voice; I can hear the
other end fine. I have checked the logs and nothing is different
for the calls that fail. I thought it was the phones, but the messages
played from asterisk
itself also have the same problem.
The native bridge in the below sections seems strange as I though this
was disabled with canreinvite=no.
denwa*CLI
-- Executing Goto(SIP/10.129.46.102-0853ec38, sip|1000|1) in new stack
-- Goto (sip,1000,1)
-- Executing SetVar(SIP/10.129.46.102-0853ec38,
CALLFILENAME=000-20051213-110514) in new sta
ck
-- Executing GotoIfTime(SIP/10.129.46.102-0853ec38,
18:00-10:00|mon-fri|*|*?24hour|s|1) in n
ew stack
-- Executing GotoIfTime(SIP/10.129.46.102-0853ec38,
*|sat-sun|*|*?24hour|s|1) in new stack
-- Executing Dial(SIP/10.129.46.102-0853ec38,
SIP/2201SIP/2202|180|tTH) in new stack
-- Called 2201
-- Called 2202
-- SIP/2201-afc3 is ringing
-- SIP/2202-4367 is ringing
-- SIP/2201-afc3 answered SIP/10.129.46.102-0853ec38
-- Attempting native bridge of SIP/10.129.46.102-0853ec38 and SIP/2201-afc3
== Spawn extension (sip, 1000, 4) exited non-zero on
'SIP/10.129.46.102-0853ec38'
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conf file:
sip.conf
[general]
port=5060
realm=ocn.ne.jp
context=sip
[EMAIL PROTECTED]:secret:[EMAIL PROTECTED]/number
disallow=all
allow=ulaw
[number]
type=friend
host=voip-ca35323.ocn.ne.jp
username=username
secret=secret
fromuser=number
fromdomain=ocn.ne.jp
port=5060
dtmfmode=inband
disallow=all
allow=ulaw
nat=yes
canreinvite=no
context=sip
[snip]
If anybody has any idea where I should look, it would be most appreciated.
Jason
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