my problem is this. I have google forward the call to gizmo5. I have this line
in my sip file :
register => user:passw...@proxy01.sipphone.com
I believe this lines connects asterisk with gizmo5 so when it gets a call from
Google, asterisk will answer it?
At the end of my sip file i have this
[Calls-From-Gizmo-Network]
type=user
context=demo
disallow=all
allow=ulaw
allow=ilbc
allow=gsm
dtmfmode=rfc2833
host=proxy01.sipphone.com
insecure=very
username=user
secret=password
canreinvite=no
In my extentions i have this:
[fromgizmo]
exten => s,1,Wait(5)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(welcome)
exten => s,n,Playback(test)
exten => s,n,Playback(test2)
exten => s,n,Hangup
The odd thing is i would have thought the
context=demo line from sip.conf would play the demo in extensions?
Instead it plays default which i put a line in to direct to fromgizmo...
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => fromgizmo
why not play demo?
Anyways, The first caller goes through just fine but the 2nd caller just gets a
ringing. the output looks like this.
-- Executing [...@default:1] Wait("SIP/198.65.166.147-084fe8b0", "5") in
new s
tack
-- Executing [...@default:2] Answer("SIP/198.65.166.147-084fe8b0", "") in
new
stack
-- Executing [...@default:3] Wait("SIP/198.65.166.147-084fe8b0", "2") in
new s
tack
-- Executing [...@default:4] Playback("SIP/198.65.166.147-084fe8b0",
"welcome"
) in new stack
-- <SIP/198.65.166.147-084fe8b0> Playing 'welcome' (language 'en')
-- Executing [...@default:5] Playback("SIP/198.65.166.147-084fe8b0",
"test") i
n new stack
-- <SIP/198.65.166.147-084fe8b0> Playing 'test' (language 'en')
-- Executing [...@default:1] Wait("SIP/198.65.166.147-084fc2e0", "5") in
new s
tack
-- Executing [...@default:2] Answer("SIP/198.65.166.147-084fc2e0", "") in
new
stack
-- Executing [...@default:3] Wait("SIP/198.65.166.147-084fc2e0", "2") in
new s
tack
-- Executing [...@default:4] Playback("SIP/198.65.166.147-084fc2e0",
"welcome"
) in new stack
-- <SIP/198.65.166.147-084fc2e0> Playing 'welcome' (language 'en')
-- Executing [...@default:5] Playback("SIP/198.65.166.147-084fc2e0",
"test") i
n new stack
-- <SIP/198.65.166.147-084fc2e0> Playing 'test' (language 'en')
-- Executing [...@default:6] Playback("SIP/198.65.166.147-084fe8b0",
"test2") in new stack
-- <SIP/198.65.166.147-084fe8b0> Playing 'test2' (language 'en')
== Spawn extension (default, s, 5) exited non-zero on
'SIP/198.65.166.147-084fc2e0'
-- Executing [...@default:7] Hangup("SIP/198.65.166.147-084fe8b0", "") in
new stack
== Spawn extension (default, s, 7) exited non-zero on
'SIP/198.65.166.147-084fe8b0'
The
odd thing is that to asterisk it looks like both calls are taken right?
But whoever is the 2nd caller goes not get the call (it just rings and then
goes to google voice mail). One more thing to
note is that if i make one call online (from sip softphone) and the other
from a land line or cell it works! Its only when i try to two "phones" (cell
and/or land line) that it does not. How can i get two "phones" connected?
Thanks!
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