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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