17 Jun
2005
17 Jun
'05
7:36 p.m.
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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