Hi Matt,
As required, please find DEBUG trace for datetime function. I have used
this function in Dialplan to capture DEBUG trace. I hope, this can help
us in resolving the issue.
[Jul 2 15:54:44] DEBUG[2698] chan_sip.c: Checking device state for peer
1001
[Jul 2 15:54:44] DEBUG[2698] devicestate.c: Changing state for SIP/1001
- state 2 (In use)
[Jul 2 15:54:44] DEBUG[2698] devicestate.c: device 'SIP/1001' state '2'
[Jul 2 15:54:44] DEBUG[2737] pbx.c: Launching 'Answer'
[Jul 2 15:54:44] VERBOSE[2737] pbx.c: -- Executing [6666@avhan:1]
Answer("SIP/1001-00000000", "") in new stack
[Jul 2 15:54:44] DEBUG[2698] devicestate.c: No provider found, checking
channel drivers for SIP - 1001
[Jul 2 15:54:44] DEBUG[2698] chan_sip.c: Checking device state for peer
1001
[Jul 2 15:54:44] DEBUG[2698] devicestate.c: Changing state for SIP/1001
- state 2 (In use)
[Jul 2 15:54:44] DEBUG[2698] devicestate.c: device 'SIP/1001' state '2'
[Jul 2 15:54:44] DEBUG[2737] chan_sip.c: SIP answering channel:
SIP/1001-00000000
[Jul 2 15:54:44] DEBUG[2737] res_rtp_asterisk.c: Setting the marker bit
due to a source update
[Jul 2 15:54:44] DEBUG[2737] chan_sip.c: Setting framing from config on
incoming call
[Jul 2 15:54:44] DEBUG[2737] chan_sip.c: ** Our capability: 0x4 (ulaw)
Video flag: True Text flag: True
[Jul 2 15:54:44] DEBUG[2737] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Jul 2 15:54:44] DEBUG[2737] chan_sip.c: -- Done with adding codecs to SDP
[Jul 2 15:54:44] DEBUG[2737] chan_sip.c: Done building SDP. Settling
with this capability: 0x4 (ulaw)
[Jul 2 15:54:44] DEBUG[2737] chan_sip.c: Trying to put 'SIP/2.0 200'
onto UDP socket destined for 192.168.2.18:7490
[Jul 2 15:54:44] DEBUG[2734] app_queue.c: Device 'SIP/1001' changed to
state '2' (In use) but we don't care because they're not a member of any
queue.
[Jul 2 15:54:44] DEBUG[2734] app_queue.c: Device 'SIP/1001' changed to
state '2' (In use) but we don't care because they're not a member of any
queue.
[Jul 2 15:54:44] DEBUG[2734] app_queue.c: Device 'SIP/1001' changed to
state '2' (In use) but we don't care because they're not a member of any
queue.
[Jul 2 15:54:44] DEBUG[2722] chan_sip.c: = Looking for Call ID:
YjNlMjU5YTJlMmQ5Njc3YjQ1MDgyMDg3ZjI1ZDViMmY. (Checking From) --From tag
226b515a --To-tag as6e727cd7
[Jul 2 15:54:44] DEBUG[2722] chan_sip.c: **** Received ACK (6) -
Command in SIP ACK
[Jul 2 15:54:44] DEBUG[2722] chan_sip.c: Stopping retransmission on
'YjNlMjU5YTJlMmQ5Njc3YjQ1MDgyMDg3ZjI1ZDViMmY.' of Response 2: Match Found
[Jul 2 15:54:44] DEBUG[2737] pbx.c: Launching 'DateTime'
[Jul 2 15:54:44] VERBOSE[2737] pbx.c: -- Executing [6666@avhan:2]
DateTime("SIP/1001-00000000", "1365120000,,YBd") in new stack
[Jul 2 15:54:44] DEBUG[2737] app_playback.c: string
<datetime:YBd:201304050530.00-5- 94> depth <0>
[Jul 2 15:54:44] DEBUG[2737] app_playback.c: try
<datetime:YBd:201304050530.00-5- 94> in <en>
[Jul 2 15:54:44] DEBUG[2737] pbx.c: Launching 'Hangup'
[Jul 2 15:54:44] VERBOSE[2737] pbx.c: -- Executing [6666@avhan:3]
Hangup("SIP/1001-00000000", "") in new stack
[Jul 2 15:54:44] DEBUG[2737] pbx.c: Spawn extension (avhan,6666,3)
exited non-zero on 'SIP/1001-00000000'
[Jul 2 15:54:44] VERBOSE[2737] pbx.c: == Spawn extension (avhan,
6666, 3) exited non-zero on 'SIP/1001-00000000'
[Jul 2 15:54:44] DEBUG[2737] channel.c: Soft-Hanging up channel
'SIP/1001-00000000'
[Jul 2 15:54:44] DEBUG[2737] channel.c: Hanging up channel
'SIP/1001-00000000'
[Jul 2 15:54:44] DEBUG[2737] chan_sip.c: Hangup call SIP/1001-00000000,
SIP callid YjNlMjU5YTJlMmQ5Njc3YjQ1MDgyMDg3ZjI1ZDViMmY.
[Jul 2 15:54:44] DEBUG[2737] chan_sip.c: Updating call counter for
incoming call
[Jul 2 15:54:44] DEBUG[2698] devicestate.c: No provider found, checking
channel drivers for SIP - 1001
[Jul 2 15:54:44] DEBUG[2698] chan_sip.c: Checking device state for peer
1001
[Jul 2 15:54:44] DEBUG[2698] devicestate.c: Changing state for SIP/1001
- state 1 (Not in use)
[Jul 2 15:54:44] DEBUG[2698] devicestate.c: device 'SIP/1001' state '1'
[Jul 2 15:54:44] DEBUG[2737] res_rtp_asterisk.c: Setting RTCP address
on RTP instance '0x98ac7f0'
[Jul 2 15:54:44] DEBUG[2737] netsock2.c: Splitting '192.168.2.18:7490'
into...
[Jul 2 15:54:44] DEBUG[2737] netsock2.c: ...host '192.168.2.18' and
port '7490'.
[Jul 2 15:54:44] DEBUG[2737] chan_sip.c: Trying to put 'BYE sip:100'
onto UDP socket destined for 192.168.2.18:7490
Thanks & Regards,
Amit Patkar
On 7/2/2013 5:15 PM, Matthew Jordan wrote:
On Tue, Jul 2, 2013 at 2:40 AM, Amit Patkar | ATPL <[email protected]
<mailto:[email protected]>> wrote:
Hello Matthew
I have pasted logs of the manager commands for the following
execution of the AGI Command and the result. As can be seen the
execution of the command replies "200 success" immediately without
executing the command. The date time is not played. Asterisk Logs
and AGI logs do not have anything of any significance , since we
use the asterisk manager.
This happens when we use mode=new , in say.conf ( Default file )
What we send through the manager commands is
Action: AGI
ActionId: 800
CommandId: 800
Channel: DAHDI/i1/115-1
Command: SAY DATETIME 1366934400 0 YBd
Response for the same is below
02-07-2013 11:35:43.578$Line: Event: AGIExec
02-07-2013 11:35:43.578$Line: Privilege: agi,all
02-07-2013 11:35:43.578$Line: SubEvent: Start
02-07-2013 11:35:43.578$Line: Channel: DAHDI/i1/115-1
02-07-2013 11:35:43.578$Line: CommandId: 456187281
02-07-2013 11:35:43.578$Line: Command: SAY DATETIME 1366934400
0 YBd
02-07-2013 11:35:43.578$Line: Event: AGIExec
02-07-2013 11:35:43.578$Line: Privilege: agi,all
02-07-2013 11:35:43.578$Line: SubEvent: End
02-07-2013 11:35:43.578$Line: Channel: DAHDI/i1/115-1
02-07-2013 11:35:43.578$Line: CommandId: 456187281
02-07-2013 11:35:43.578$Line: Command: SAY DATETIME 1366934400
0 YBd
02-07-2013 11:35:43.578$Line: ResultCode: 200
02-07-2013 11:35:43.578$Line: Result: Success
02-07-2013 11:35:43.578$Line: Event: AsyncAGI
02-07-2013 11:35:43.578$Line: Privilege: agi,all
02-07-2013 11:35:43.578$Line: SubEvent: Exec
02-07-2013 11:35:43.578$Line: Channel: DAHDI/i1/115-1
02-07-2013 11:35:43.578$Line: CommandID: 800
02-07-2013 11:35:43.578$Line: Result: 200%20result%3D0%0A
Please suggest next steps. We have to play date in different
voices and we are not able to do it because of this issue.Also we
can't implement different language due to this limitation.
Please don't reply to me directly.
When issues are discussed on the asterisk-users mailing list, the
process by which the issue is diagnosed and/or resolved helps to build
a knowledge base for the Asterisk community. If someone runs into a
similar issue, they can see how the issue was resolved - or if it is a
valid issue. That way, everyone in the community benefits!
The reason I asked for a pastebin of the DEBUG (and higher messages)
is because those messages will tell me the code paths your command is
taking. The AMI AsyncAGI responses unfortunately do not.
Matt
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