Re: [asterisk-users] MixMonitor and attended transfers [SOLVED]

[Ishfaq Malik]

On Tue, 2011-08-02 at 10:58 +0100, Ishfaq Malik wrote:
> Hi
> 
> I'm using asterisk 1.8.3.2 (with a couple of patches)
> 
> I have the following scenario...
> 
> SIP call comes in and gets answered by extension A (MixMonitor is
> executed as part of this inbound dial plan of the number being called)
> 
> Extension A puts call on hold and calls extension B
> 
> Extension A then does an attended transfer of incoming call to extension
> B
> 
> I'm finding that the recording only lasts up to the point that the
> transfer is made.
> 
> Is this the correct behaviour? Is there any way I could make this
> inbound call into a single continuous recording?
> 
> Thanks in advance
> 
> Ish

I got a resolution to this from Digium support. You need to add the
following after the MixMonitor step

Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)

And then the recording follows the whole call.

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] MixMonitor and attended transfers

[Ishfaq Malik]

On Tue, 2011-08-02 at 10:58 +0100, Ishfaq Malik wrote:
> Hi
> 
> I'm using asterisk 1.8.3.2 (with a couple of patches)
> 
> I have the following scenario...
> 
> SIP call comes in and gets answered by extension A (MixMonitor is
> executed as part of this inbound dial plan of the number being called)
> 
> Extension A puts call on hold and calls extension B
> 
> Extension A then does an attended transfer of incoming call to extension
> B
> 
> I'm finding that the recording only lasts up to the point that the
> transfer is made.
> 
> Is this the correct behaviour? Is there any way I could make this
> inbound call into a single continuous recording?
> 
> Thanks in advance
> 
> Ish

Here's part of the log for this procedure

[2011-08-02 13:47:13] VERBOSE[6475] rtp_engine.c: -- Locally bridging 
SIP/A-0049 and SIP/B-004a
[2011-08-02 13:47:20] VERBOSE[6475] rtp_engine.c: -- Locally bridging 
SIP/inbound-0047 and SIP/B-004a
[2011-08-02 13:47:20] VERBOSE[6463] pbx.c:   == Spawn extension (inbound, s, 4) 
exited non-zero on 'SIP/A-0049'
[2011-08-02 13:47:20] VERBOSE[6464] app_mixmonitor.c:   == MixMonitor close 
filestream
[2011-08-02 13:47:26] VERBOSE[6475] app_macro.c:   == Spawn extension 
(macro-stdexten, s, 1) exited non-zero on 'SIP/inbound-0047' in macro 
'stdexten'
[2011-08-02 13:47:26] VERBOSE[6475] pbx.c:   == Spawn extension (local, B, 1) 
exited non-zero on 'SIP/inbound-0047'
[2011-08-02 13:47:26] VERBOSE[6464] app_mixmonitor.c:   == End MixMonitor 
Recording SIP/inbound-0047

Obviously, I've obscured some of the more sensitive details in there

The thing to notice here though is that MixMonitor closes the filestream
when I hit the transfer button but actually Ends the recording 6 seconds
later when the whole call was ended.

This seems like inconsistent behaviour and more like an unintentional
consequence of changes rather than intended behaviour, i.e. why would
you close the filestream yet not end the recording?

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] MixMonitor and attended transfers

[Ishfaq Malik]

On Tue, 2011-08-02 at 07:51 -0400, Dan Journo wrote:
> > Hi
> 
> > 
> 
> > I'm using asterisk 1.8.3.2 (with a couple of patches) 
> 
> > I have the following scenario...
> 
> > SIP call comes in and gets answered by extension A (MixMonitor is
> 
> > executed as part of this inbound dial plan of the number being
> called)
> 
> > 
> 
> > Extension A puts call on hold and calls extension B
> 
> > 
> 
> > Extension A then does an attended transfer of incoming call to
> extension
> 
> > B
> 
> > 
> 
> > I'm finding that the recording only lasts up to the point that the
> 
> > transfer is made.
> 
> > 
> 
> > Is this the correct behaviour? Is there any way I could make this
> 
> > inbound call into a single continuous recording?
> 
>  
> 
> I've not used 1.8 yet, but in 1.4, you could send the incoming call
> through a LOCAL channel when the call comes in, and start the
> recording on the Local channel.
> 
> That way, the LOCAL channel should keep recording, even when you
> transfer the call.
> 
> You may need to add /n
> 
> http://www.voip-info.org/wiki/view/Asterisk+local+channels
> 
>  
> 
> Hope that helps. It's a little hard to explain, but try it out.
>  
> 
> Dan Journo
> 
> Kesher Communications (UK)
> 
> Business Phone Systems | Hosted PBX
> 

I didn't have this problem with 1.4, it just recorded the whole message as a 
matter of course.

I'll have a look into using local channels for this but I think it has
more to do with the way that 1.8 as treating attended transfers and how
it joins the 2 channels involved.

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] MixMonitor and attended transfers

[Dan Journo]

> Hi

>

> I'm using asterisk 1.8.3.2 (with a couple of patches)

>

> I have the following scenario...

>

> SIP call comes in and gets answered by extension A (MixMonitor is

> executed as part of this inbound dial plan of the number being called)

>

> Extension A puts call on hold and calls extension B

>

> Extension A then does an attended transfer of incoming call to extension

> B

>

> I'm finding that the recording only lasts up to the point that the

> transfer is made.

>

> Is this the correct behaviour? Is there any way I could make this

> inbound call into a single continuous recording?



I've not used 1.8 yet, but in 1.4, you could send the incoming call through a 
LOCAL channel when the call comes in, and start the recording on the Local 
channel.

That way, the LOCAL channel should keep recording, even when you transfer the 
call.



You may need to add /n



http://www.voip-info.org/wiki/view/Asterisk+local+channels



Hope that helps. It's a little hard to explain, but try it out.




Dan Journo
Kesher Communications (UK)
Business Phone Systems | Hosted 
PBX








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[asterisk-users] MixMonitor and attended transfers

[Ishfaq Malik]

Hi

I'm using asterisk 1.8.3.2 (with a couple of patches)

I have the following scenario...

SIP call comes in and gets answered by extension A (MixMonitor is
executed as part of this inbound dial plan of the number being called)

Extension A puts call on hold and calls extension B

Extension A then does an attended transfer of incoming call to extension
B

I'm finding that the recording only lasts up to the point that the
transfer is made.

Is this the correct behaviour? Is there any way I could make this
inbound call into a single continuous recording?

Thanks in advance

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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