[Asterisk-Users] Audio cuts off 10 minutes into calls

[Roar Lorentzen, IP-Telefoni as]

Hello list,

 

We run Asterisk CVS-HEAD-06/02/04-11:25:18 built by [EMAIL PROTECTED] 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