[Asterisk-Users] Record call from switch using serviceobserve? (execute command after dial?)

[Garry Adkins]

Hmmm. Now I have another problem...

After the call goes to the extension 100 in this example, I get a jump to
the "t" extension for the context.  I can't find a way to make the call not
time out, and asterisk acts like it needs to do something after the record
start (i.e. the 100,2 in  your example).  When it jumps to the non-existing
"t" (for timeout) it hangs up.

This is kind of kludgey, but I sent it to a "quiet" (no announcements)
meetme room.

Any better way to handle this?

Otherwise it works fine!

Thanks!

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com 
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
> Adam Goryachev
> Sent: Thursday, June 24, 2004 7:51 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Record call from switch using 
> serviceobserve? (execute command after dial?)
> 
> Use the call file, and set the channel to something like:
> 
> Zap/1/160w<extension>
> 
> then set the extension/context/etc to point to something like this:
> exten => 100,1,playback("Now call will be recorded") exten => 
> 100,2,Record("some file") exten => 100,3,playback("beep")
> 
> Now, to stop recording you have two choices, either record to 
> the end of the call, or else use the manager interface to 
> signal a soft hangup. 
> 
> Actually, better is to use the manager interface to transfer 
> the x100P to another extension like this:
> exten => 101,1,playback("Call no longer being recorded") 
> exten => 101,2,StopRecord exten => 101,3,playback("beep") 
> exten => 101,4,softhangup
> 
> Check the proper applications for stoprecording and softhangup....
> 
> Regards,
> Adam
> 
> On Fri, 2004-06-25 at 04:54, Garry Adkins wrote:
> > Hi,
> >  
> > I am working on a project to record agent calls when completing 
> > specific transactions with customers.
> >  
> > Since these calls do not go through the asterisk box (They 
> go through 
> > a lucent G3), we're thinking that service observe would be 
> the easiest 
> > way to accomplish our goal.
> >  
> > Here's what I need:
> > On demand, I need to be able to attach to the switch, dial 
> the service 
> > observe code, make an announcement, record.
> > On the second event, I need to make an announcement, stop the 
> > recording, and hang-up the channel to the switch.
> >  
> > 
> > Here's my plan:
> >  
> > 1)  Agent software calls a CGI on the asterisk box.  This passes 
> > extension the agent is talking on.
> > 2)  CGI program somehow makes asterisk call to the switch, dials 
> > 160w<extension> which does a service observe (i.e. attaches the 
> > <extension> audio to our channel)
> > 3)  Asterisk play recording about transaction being recorded
> > 4)  Start recording
> > 5)  Software calls CGI again to notify asterisk to stop the 
> recording.
> > 6)  Asterisk plays recording that the transaction is recorded
> > 7)  Asterisk disconnects channel.
> >  
> > 
> > Eventually I will have a T1 interface into the switch, but 
> for testing 
> > I'm just using the X100P and an analog port on the switch.
> >  
> > The two communicate properly, I can call the asterisk box 
> and have it 
> > answer, and I can generate a call to the switch from a different 
> > extension on the Asterisk box.
> >  
> > Here's my attempted solutions:
> >  
> > 1) When I try to generate the call from a SIP phone, it 
> works fine.  
> > The extensions.conf contains a dial(zap/1/160w<extension>)
> >  
> > 
> > 2) When I try to generate the call from the manager interface, I 
> > cannot do it without having a different input.
> > action: originate
> > channel: zap/1
> > exten: 555
> > context: default
> > priority: 1
> >  
> > Extension 555 does a dial(zap/1/160w<extension>)
> >  
> > Three problems:
> >    a) The problem is I have no other channels but the ZAP 
> channel for 
> > the X100p.  I can't connect both ends to the same channel.
> >    b) Also, I cannot send audio to this channel from the manager 
> > channel (for the announcement of the recording)
> >    c)  Dial doesn't exit until hang-up, so I cannot 
> background() the 
> > audio to the channel.
> >  
> > 
> > 
> > 
> > 
> > 3)  When I try to dial by generating a call file in the proper 
> > outbound call directory, I still get stuck on the dial command.
> >  
> > 
> > Any ideas?  Am I just not understanding something critical?
> >  
> > 
> > Thanks for any help!  I've search the archives and the WIKI 
> for about 
> > 3 days.  I'm stumped!
> >  
> > -G
> > 
> > 
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> --
>  --
> Adam Goryachev
> Website Managers
> Ph:  +61 2 9345 4395                        
> adam at websitemanagers.com.au
> Fax: +61 2 9345 4396                        www.websitemanagers.com.au
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
>