[Asterisk-Users] Audio cuts off 10 minutes into calls
Roar Lorentzen, IP-Telefoni as
roar at iptelefoni.no
Wed Jul 7 11:42:35 MST 2004
Hello list,
We run Asterisk CVS-HEAD-06/02/04-11:25:18 built by root at Gate01 on a i686
running Linux.
All works fine except Audio is lost 10minutes into the call. This happens
for every call
PSTN-SIP, SIP-PSTN, SIP-SIP
Example of one call setup using Snom200 and Grandstream 486:
-- Executing Macro("SIP/xxxx1251-d638", "CFW|xxxx1251|SIP/xxxx1253") in new
stack
-- Executing DBget("SIP/xxxx1251-d638", "temp=CFIM/xxxx1253") in new
stack
-- DBget: varname=temp, family=CFIM, key=xxxx1253
-- DBget: Value not found in database.
-- Executing Goto("SIP/xxxx1251-d638", "s|4") in new stack
-- Goto (macro-CFW,s,4)
-- Executing Dial("SIP/xxxx1251-d638", "SIP/xxxx1253|30|t") in new stack
-- Called xxxx1253
-- SIP/xxxx1253-c5dc is ringing
-- SIP/xxxx1253-c5dc answered SIP/xxxx1251-d638
-- Attempting native bridge of SIP/xxxx1251-d638 and SIP/xxxx1253-c5dc
-- Attempting native bridge of SIP/xxxx1251-d638 and SIP/xxxx1253-c5dc
PSTN to SIP(Grandstream):
-- Executing Macro("Zap/15-1", "CFW|xxxx1253|SIP/xxxx1253") in new stack
-- Executing DBget("Zap/15-1", "temp=CFIM/xxxx1253") in new stack
-- DBget: varname=temp, family=CFIM, key=xxxx1253
-- DBget: Value not found in database.
-- Executing Goto("Zap/15-1", "s|4") in new stack
-- Goto (macro-CFW,s,4)
-- Executing Dial("Zap/15-1", "SIP/xxxx1253|30|t") in new stack
-- Called xxxx1253
-- Accepting call from 'xxxx6857' to 'xxxx1253' on channel 0/15, span 1
-- SIP/xxxx1253-bc29 is ringing
-- SIP/xxxx1253-bc29 answered Zap/15-1
I have set verbose 5, and nothing else is reported when audio is lost, when
I hang up the call some time after audio is lost
this is reported:(For PSTN-SIP(Grandstream)
Spawn extension (macro-CFW, s, 4) exited non-zero on 'Zap/15-1' in macro
'CFW'
== Spawn extension (default, xxxx1253, 1) exited non-zero on 'Zap/15-1'
-- Hungup 'Zap/15-1'
The call is not hung up, just loss of audio.
I have searched the archives and google without any luck.
Could someone pls give me a pointer of what may be the cause of this
problem.
We use TE410 PRI card, and the SIP clients are: Grandstream HandyTone 486,
Snom 200, Zyxel P2000W
With regards
Roar
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