find the inline comment...
On 07/29/2011 12:11 AM, Ishwar Sridharan wrote:
The dialplan is very simple. When the call comes in, we hand the call
over to adhearsion.
This is how the dialplan looks:
;group 0 will be used for incoming calls
EXOIN = DAHDI/g0
;group 11 for outgoing
EXOOUT = DAHDI/G11
;This will be used by adhearsion
EXOCID=xxxxxxxx
[general]
autofallthrough = yes ;really?
clearglobalvars = no
[frompstn]
;Send everything to adhearsion
exten => _X.,1,Ringing
exten => _X.,n,AGI(agi://127.0.0.1 <http://127.0.0.1>)
exten => _X.,n,Hangup() ; Please try this.
; End dialplan
The rest of the logic happens in adhearsion.
--
Thanks,
Ishwar.
On Thu, Jul 28, 2011 at 6:33 PM, Nikhil <[email protected]
<mailto:[email protected]>> wrote:
Can you share the dialplan ,where SIP call is dialing...
Thanks
Nikhil
On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
Hello everybody,
We have an asterisk 1.8.4.1 setup, connected to a PRI line.
We're currently facing an issue where asterisk does not recognise
the event when the called party declines/cuts the call. This
happens specifically over calls on a PRI line. For calls over
SIP, call decline event is captured properly.
I wasn't able to find a solution on the asterisk-users mailing
list archive. Any suggestions/help would be much appreiciated :)
I can share the relevant parts of the configuration files, if needed.
Here's an excerpt from asterisk logs for a SIP call.
-- SIP/xxxxx-00000000 requested special control 16, passing
it to SIP/xxxxx-00000001
-- Started music on hold, class 'default', on SIP/xxxxx-00000001
-- SIP/xxxxx-00000000 requested special control 20, passing
it to SIP/xxxxx-00000001
-- Got SIP response 603 "Decline" back from 127.0.0.1:5063
<http://127.0.0.1:5063/>
-- SIP/xxxxx-00000001 is busy
-- Stopped music on hold on SIP/xxxxx-00000001
As you can see, on a SIP call, a call reject event is identified.
For a call over the PRI, on the other hand, this event is not
recognised. Here's an excerpt from asterisk log for a call over PRI.
Call from yyyy to xxxx.
-- Requested transfer capability: 0x10 - 3K1AUDIO
-- Called G11/xxxxx
-- Started music on hold, class 'default', on DAHDI/i1/yyyyy
-- DAHDI/i1/xxxxx-18f8 is proceeding passing it to DAHDI/i1/yyyyy
-- DAHDI/i1/xxxxx-18f8 is ringing
# At this point in time, xxxxx rejects the call. The event that's
logged in asterisk is the following:
-- DAHDI/i1/xxxxx-18f8 is making progress passing it to
DAHDI/i1/yyyyy
# And the call times out after the default 30s.
-- Nobody picked up in 30000 ms
Is there a reason why asterisk doesn't recognise the "call
decline", and does it need any configuration changes to enable this?
Thanks for your help.
--
Cheers,
Ishwar.
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