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Jason Frisch

2005-Dec-12 19:13 UTC

[Asterisk-Users] No outgoing sound...sometimes

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/2201&SIP/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'

-------------------------

conf file:

sip.conf
[general]
port=5060
realm=ocn.ne.jp
context=sip
register=number@ocn.ne.jp:secret:LNRTKR4U@voip-ca35323.ocn.ne.jp/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

Jason Frisch

2005-Dec-12 19:55 UTC

head link

[Asterisk-Users] No outgoing sound...sometimes

Think I have it sorted :-)
For those that have the same trouble, try replacing your NIC.

Jason

Mario Evangelista-Silva

2005-Dec-13 04:20 UTC

head link

[Asterisk-Users] No outgoing sound...sometimes

Verify communication between protocols.  SIP ou IAX2.





Jason Frisch <jfrisch@tsukaeru.net>
Enviado Por: asterisk-users-bounces@lists.digium.com
13/12/05 00:13
Favor responder a Asterisk Users Mailing List - Non-Commercial Discussion

 
        Para:   asterisk-users@lists.digium.com
        cc: 
        Assunto:        [Asterisk-Users] No outgoing sound...sometimes
 - 


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/2201&SIP/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'

-------------------------

conf file:

sip.conf
[general]
port=5060
realm=ocn.ne.jp
context=sip
register=number@ocn.ne.jp:secret:LNRTKR4U@voip-ca35323.ocn.ne.jp/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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asterisk users - Dec 2005 - No outgoing sound...sometimes

[Asterisk-Users] No outgoing sound...sometimes

[Asterisk-Users] No outgoing sound...sometimes

[Asterisk-Users] No outgoing sound...sometimes