[Serusers] 30s calls

Greger V. Teigre greger at teigre.com
Fri Jun 17 17:36:09 CEST 2005


In order for the call to not disconnect, the final ACK from caller (the ACK 
to the OK sent by the callee) must be received by the callee (Asterisk). So, 
your problem may be caused by your ser.cfg not handling ACKs properly.  I 
have experienced myself that removing the loose route handling (or moving it 
too far down in the script) will create this symptom.  Have a look at the 
ngrep output on the Asterisk box and see if you receive the ACK.  If not, 
have a look at the reference configs at http://onsip.org/ to see how ACKs 
are handled.
g-)

Jerlique Ban wrote:
> Hello,
>
> I have configured SER and asterisk to allow me to make calls to the
> PSTN network, however on my voip phone (behind NAT) I am having
> issues with the voip tx audio dropping out after 30 seconds.  Now I'm
> guessing it's a nat issue but even that doesn't really make sense!!
>
> Why, well because the only the TX of the voip phone drops out (ie the
> PSTN phone cannot hear what is said on the voip phone). The PSTN
> phone can still transmit audio to the voip phone (through the nat).
>
> Anyway in SER, I have set the natping_interval to 5 seconds, and this
> still doesn't resolve the issue. Strangely at the time that the audio
> disconnects Asterisk is sending my phone an INVITE message. Why would
> it do this mid call?
>
> I'm using SER0.9.0+Asterisk as my platform.
>
> Any pointers??
>
> JB
>
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