[asterisk-users] re: putting 2 SIP channels together - hangup issues

[Yair Hakak]

Hello all,
Hoping someone can help me with an issue...I have i .call file which calls
out on a SIP channel and connects to an extension which dials another SIP
channel. (both via voip providers) - both to PSTN.

Problem is, hanging up the POTS phone doesn't release the channel (either
one - hanging up the calling channel or the destination doesn't do it).

Using IAX instead of SIP works better; it releases the voip channels but not
the POTS channels (i.e. the POTS phones don't immediately go back to a dial
tone or fast busy).

I'm using asterisk 1.4
here are the relevant bits:

the extension:

exten=> 157,1,Answer
exten=> 157,2,Dial(IAX2/myusername at voipjet/PSTNnumberToCall2, 60)
exten=> 157,3,Hangup

the .call file:

# Create the call on group 2 dial lines and set up
#  some re-try timers
#
Channel: IAX2/username at voipjet/PSTNnumberToCall1
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Extension: 157
Priority: 1

if anyone can shed some light on this i'd be eternally grateful, first of
all, why if i glue 2 SIP channels together hanging up the POTS phone doesn't
release the SIP channels, and second why if i glue two IAX channels together
it doesnt release the POTS lines.

thanks,

 yair
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