asterisk users - Mar 2004 - IAX drops calls exactly 5 secs into the call

Hi List,

Two boxes

A   has a PRI 

B   terminates SIP devices


A  <--IAX-->  B

Both on the same switch, same IP network.

Call from PSTN to A gets pushed via IAX to B -> Sip device
with no problems.

Call from Sip device -> B via IAX -> A -> PSTN 
will drop exactly 5 seconds after the call is answered.

I've built with 0.7.2, 1_0_Stable  devel    etc


Any clue / hints ??

thanks

> Two boxes
> 
> A   has a PRI 
> 
> B   terminates SIP devices
> 
> 
> A  <--IAX-->  B
> 
> Both on the same switch, same IP network.
> 
> Call from PSTN to A gets pushed via IAX to B -> Sip device
> with no problems.
> 
> Call from Sip device -> B via IAX -> A -> PSTN 
> will drop exactly 5 seconds after the call is answered.
> 
> I've built with 0.7.2, 1_0_Stable  devel    etc
> 
> 
> Any clue / hints ??
Not enough info to guess. The five second disconnect would suggest a sip
problem, but without "iax debug" or "sip debug" outputs (or
even some
CLI clues), some clue as to sip phones used, conf files, etc, no guess. 
Might check the /var/log/asterisk/messages file to see if anything 
provides a hint.

Rich

>
> Hi List,
>
> Two boxes
>
> A   has a PRI
>
> B   terminates SIP devices
>
>
> A  <--IAX-->  B
>
> Both on the same switch, same IP network.
>
> Call from PSTN to A gets pushed via IAX to B -> Sip device
> with no problems.
>
> Call from Sip device -> B via IAX -> A -> PSTN
> will drop exactly 5 seconds after the call is answered.
>
> I've built with 0.7.2, 1_0_Stable  devel    etc
>
>
> Any clue / hints ??
>
This strongly resembles a SIP problem I had with Grandstream SIP phones. The
phones did not like the codec and refused to ACK. The Invite protocol should
go; Phone sends INVITE
Asterisk sends 200 OK with preferred CODEC
Phone refuses to ACK (does not like CODEC)
Media exchange begins (call seems OK)
A few seconds later * dumps call cos no ACK

Do a dump of the protocol (Ethereal, tcpdump) and check it out. Might be a
similar problem.

Brian