Hi,
I have a problem where calling "out" of asterisk when the call is answered
dahdi hangs up immediately.
For example: Sip Client A calls external number. Route: SIP -> FXO -> GSM
Gateway ->External Landline.
However when that external landline answers the call dahdi hangs up
immediately .
Going the other way is fine (External Landline -> GSM Gateway -> FXO ->
SIP).
I've tried multiple different internet searches and can't seem to find any
information on this problem.
Below are my config files.
Sip.conf
[office-phone](!)
type=friend
context=sipofficephone
host=dynamic
nat=yes
#secret=xxxx
dtmfmode=auto
disallow=all
;allow=ulaw
allow=alaw
allow=GSM
[lewisphone](office-phone);lewis mobile
secret=xxxx
Chan_dahdi.conf
[channels]
signalling=fxs_ks
context=pstnincomming
group=0
channel => 1
Extensions.conf
[sipofficephone]
exten => _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})
same => n,Dial(DAHDI/1/${EXTEN})
same => n,Hangup()
[pstnincomming]Diamon
exten => s,1,Answer()
same => n,Dial(SIP/lewisphone)
same => n,Hangup()
Asterisk CLI Output (Verbose 3)
My comments bold.
== Using SIP RTP CoS mark 5
-- Executing [xxxx@sipofficephone:1] Verbose("SIP/lewisphone-0000000a",
"2,Call from VoIP network to xxxx") in new stack
== Call from VoIP network to xxxx
-- Executing [xxxx@sipofficephone:2] Dial("SIP/lewisphone-0000000a",
"DAHDI/1/xxxx") in new stack
-- Called DAHDI/1/xxxx
-- DAHDI/1-1 answered SIP/lewisphone-0000000a GSM Gateway Answering Call
then Sending it out.
-- Hanging up on 'DAHDI/1-1' Dest answering call to which DAHDI hangs up
-- Hungup 'DAHDI/1-1'
== Spawn extension (sipofficephone, xxxx, 2) exited non-zero on
'SIP/lewisphone-0000000a'
Best Regards
Lewis
digitalselect-e
www.Digital-Select.com <http://www.digital-select.com/>
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