Re: [asterisk-users] bug in 1.2.24
Thank you for the example Isaac. I did as you mentioned and now it seems to
be working perfectly.
Saludos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Isaac Xiao
Sent: jueves, 13 de septiembre de 2007 10:33 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] bug in 1.2.24
Here is our dial plan. You need to avoid double recording as well when
you transfer the call to other extension.
exten = 7141,3,Set(CALLFILENAME=q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten = 7141,4,Set(__FROM-EXT-QUEUES=ext-queues)
exten = 7141,5,MixMonitor(${CALLFILENAME}.gsm|b)
exten = 7141,6,Playback(custom/None)
exten = 7141,7,Queue(7141|t|||7200)
Here is the CLI log.
-- Executing Playback(Zap/9-1, monitoring) in new stack
-- Playing 'monitoring' (language 'md')
-- Executing Playback(Zap/9-1, press-1-to-msg) in new stack
-- Playing 'press-1-to-msg' (language 'md')
-- Executing Goto(Zap/9-1, ext-queues|7141|1) in new stack
-- Goto (ext-queues,7141,1)
-- Executing NoOp(Zap/9-1, do not answer call before entering
queue) in new stack
-- Executing SetCIDName(Zap/9-1, CN) in new stack
-- Executing Set(Zap/9-1,
CALLFILENAME=q7141-20070914-132445-1189740177.10324) in new stack
-- Executing Set(Zap/9-1, __FROM-EXT-QUEUES=ext-queues) in new
stack
-- Executing MixMonitor(Zap/9-1,
q7141-20070914-132445-1189740177.10324.gsm|b) in new stack
-- Executing Playback(Zap/9-1, custom/None) in new stack
-- Executing Queue(Zap/9-1, 7141|t|||7200) in new stack
So Yes. As long as Zap/9-1 channel (customer's channel) not hangs up, it
will be always recorded.
Isaac
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Re: [asterisk-users] bug in 1.2.24
On 13:33, Fri 14 Sep 07, Isaac Xiao wrote:
Here is our dial plan. You need to avoid double recording as well when
you transfer the call to other extension.
exten = 7141,3,Set(CALLFILENAME=q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten = 7141,4,Set(__FROM-EXT-QUEUES=ext-queues)
exten = 7141,5,MixMonitor(${CALLFILENAME}.gsm|b)
exten = 7141,6,Playback(custom/None)
exten = 7141,7,Queue(7141|t|||7200)
Here is the CLI log.
-- Executing Playback(Zap/9-1, monitoring) in new stack
-- Playing 'monitoring' (language 'md')
-- Executing Playback(Zap/9-1, press-1-to-msg) in new stack
-- Playing 'press-1-to-msg' (language 'md')
-- Executing Goto(Zap/9-1, ext-queues|7141|1) in new stack
-- Goto (ext-queues,7141,1)
-- Executing NoOp(Zap/9-1, do not answer call before entering
queue) in new stack
-- Executing SetCIDName(Zap/9-1, CN) in new stack
-- Executing Set(Zap/9-1,
CALLFILENAME=q7141-20070914-132445-1189740177.10324) in new stack
-- Executing Set(Zap/9-1, __FROM-EXT-QUEUES=ext-queues) in new
stack
-- Executing MixMonitor(Zap/9-1,
q7141-20070914-132445-1189740177.10324.gsm|b) in new stack
-- Executing Playback(Zap/9-1, custom/None) in new stack
-- Executing Queue(Zap/9-1, 7141|t|||7200) in new stack
So Yes. As long as Zap/9-1 channel (customer's channel) not hangs up, it
will be always recorded.
Can you confirm this bug is present in 1.4 as well?
1.2 has gone into security-patches-only mode so no bugfixes
will be made for it.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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[asterisk-users] bug in 1.2.24
Here is our dial plan. You need to avoid double recording as well when
you transfer the call to other extension.
exten = 7141,3,Set(CALLFILENAME=q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten = 7141,4,Set(__FROM-EXT-QUEUES=ext-queues)
exten = 7141,5,MixMonitor(${CALLFILENAME}.gsm|b)
exten = 7141,6,Playback(custom/None)
exten = 7141,7,Queue(7141|t|||7200)
Here is the CLI log.
-- Executing Playback(Zap/9-1, monitoring) in new stack
-- Playing 'monitoring' (language 'md')
-- Executing Playback(Zap/9-1, press-1-to-msg) in new stack
-- Playing 'press-1-to-msg' (language 'md')
-- Executing Goto(Zap/9-1, ext-queues|7141|1) in new stack
-- Goto (ext-queues,7141,1)
-- Executing NoOp(Zap/9-1, do not answer call before entering
queue) in new stack
-- Executing SetCIDName(Zap/9-1, CN) in new stack
-- Executing Set(Zap/9-1,
CALLFILENAME=q7141-20070914-132445-1189740177.10324) in new stack
-- Executing Set(Zap/9-1, __FROM-EXT-QUEUES=ext-queues) in new
stack
-- Executing MixMonitor(Zap/9-1,
q7141-20070914-132445-1189740177.10324.gsm|b) in new stack
-- Executing Playback(Zap/9-1, custom/None) in new stack
-- Executing Queue(Zap/9-1, 7141|t|||7200) in new stack
So Yes. As long as Zap/9-1 channel (customer's channel) not hangs up, it
will be always recorded.
Isaac
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This e-mail and its attachments are not intended to constitute financial advice
or recommendation of, or an offer to buy or sell, any securities or other
financial products.
We recommend that you seek your own independent legal or financial advice
before proceeding with any investment decision.
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Re: [asterisk-users] bug in 1.2.24
Thank Isaac, Ill try it this way.. Im currently using this before entering
the queue so calls from the queue are recorded:
exten =
s,n,SetVar(MONITOR_FILENAME=/var/spool/asterisk/${TIMESTAMP}-${UNIQUEID}-${C
ALLERIDNUM}-Queue-Ventas)
exten = s,n,SetVar(TRANSFER_CONTEXT=internalphones)
So I could just run mixmonitor instead of those lines and thats it? Queue
call will be recorded and everything that happens afterwards if it is
transferred?
Saludos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Isaac Xiao
Sent: martes, 11 de septiembre de 2007 06:24 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] bug in 1.2.24
It is not a bug. attended Transfer is using Local channel, if you have a
look the debug log from CLI, you will see why it fails. To solve this
problem, enable recording before the calls go into the queue.
Exten = ,1,MixMonitor(...)
Exten = ,2,Goto(ext-queue, , 1)
This will ensure you to record the customer/caller's channel instead of
exten's channel. So no matter where you transfer the call and as long as
the caller not hangup the call, it will be always recorded.
By the way, 1.2.24 stable, we got problem with 1.2.21. 1.2.17 seems
stable.
Good luck,
Isaac Xiao
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[asterisk-users] bug in 1.2.24
GUys.. I dont know if this is a known bug or not but I just tested and replicated this one over and over again. It involves call transfer from calls that entered the pbx via a queue.. say a call comes in and its thrown in a queue, somebody answers the call but then wants to transfer the call to somebody else outside the queue, of course... the bug comes in here.. Im using mixmonitor to record calls and when this scenario happens, the recording of the call coming in is OK, the call when in the queue and taking to the agent is OK, but then, when the agent transfers the call using attended transfer, mixmonitor stops recording... this doesn't happen if the call is transfer using BLIND transfer, just when using ATTENDED. Anybody seen this? Any bug fix or patch for 1.2.24 for this? Thx guys ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bug in 1.2.24
It is not a bug. attended Transfer is using Local channel, if you have a look the debug log from CLI, you will see why it fails. To solve this problem, enable recording before the calls go into the queue. Exten = ,1,MixMonitor(...) Exten = ,2,Goto(ext-queue, , 1) This will ensure you to record the customer/caller's channel instead of exten's channel. So no matter where you transfer the call and as long as the caller not hangup the call, it will be always recorded. By the way, 1.2.24 stable, we got problem with 1.2.21. 1.2.17 seems stable. Good luck, Isaac Xiao WARNING - This e-mail and any attachments may be CONFIDENTIAL and are for the intended addressee only. If received in error, please delete and inform us by returning an email. Any unauthorized copying, disclosure or distribution of the material in this email is strictly prohibited. E-mail transmission cannot be guaranteed to be secure, error-free or virus-free. The sender therefore does not accept liability for any errors, omissions or consequences which arise as a result of e-mail transmission. This e-mail and its attachments are not intended to constitute financial advice or recommendation of, or an offer to buy or sell, any securities or other financial products. We recommend that you seek your own independent legal or financial advice before proceeding with any investment decision. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
