[Asterisk-Users] sample.call + chan_h323 gives seg fault

[Jeremy McNamara]

Send me the backtrace and console output, off list.

That's a pretty crazy extension.   I bet your trying to make some kind 
of crazy callback system :)



Jeremy McNamara




Chee Foong wrote:

>I dumped the following test.call file into /var/spool/asterisk/outgoing
>gives me segmentation fault :(
>
>Channel: H323/0143126544
>MaxRetries: 2
>RetryTime: 60
>WaitTime: 30
>Context: voip-test
>Extension: 90324324433
>Priority: 1
>
>same thing happend if I execute dial command on console.
>
>I figure out that this happen only if I dial through a H323 channel. I am
>using chan_h323.
>
>Any one experience the same thing?
>
>Foong
>
>----- Original Message -----
>From: "Andy Powell" <andy at beagles-den.demon.co.uk>
>To: <asterisk-users at lists.digium.com>
>Sent: Wednesday, July 30, 2003 6:56 PM
>Subject: Re: [Asterisk-Users] Call Transfer
>
>
>  
>
>>Foong
>>
>>Take a look at the sample.call file, modifying the settings in there and
>>    
>>
>copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial
>the call.. an example config is below
>  
>
>>Channel: SIP/1000 at mysipcontext
>>MaxRetries: 2
>>RetryTime: 60
>>WaitTime: 30
>>Context: mysipcontext2
>>Extension: 2000
>>Priority: 1
>>
>>This will make asterisk dial exten 1000 in the context mysipcontext when
>>    
>>
>it's answered it will then call exten 2000 in mysipcontext2..
>  
>
>>All you need is a script to lookup in the database and generate the script
>>    
>>
>file for you and it's done.
>  
>
>>HTH
>>
>>Andy
>>
>>
>>*********** REPLY SEPARATOR  ***********
>>
>>On 30/07/2003 at 16:30 Chee Foong wrote:
>>
>>    
>>
>>>Hello Dan,
>>>
>>>Thanks for you reply.
>>>
>>>Base on you recomendation using the 'T' argument. I manage to do call
>>>transfer an it works really well.
>>>
>>>My problem comes when my boss comes out with a superb idea where the
>>>transfering process is automated without involving a human :(
>>>
>>>Say asterisk get 2 numbers (from database, text file, etc), one belongs
>>>party A and the other belongs to party B. Asterisk will calls both
>>>      
>>>
>parties
>  
>
>>>and do the tranfer automatically. In another words, asterisk is
>>>      
>>>
>resposible
>  
>
>>>to 'press' the '#' to do the transfer. I don't this can be achieve in the
>>>extension.conf not matter how you structure you dial plan.
>>>
>>>Perhaps, the only way is to write a apps and plug it into asterisk like
>>>      
>>>
>all
>  
>
>>>the asterisk modules such as Meetme.
>>>
>>>Any ideas?
>>>
>>>
>>>Foong
>>>
>>>----- Original Message -----
>>>From: "Dan" <dtoma at fx.ro>
>>>To: <asterisk-users at lists.digium.com>
>>>Sent: Wednesday, July 30, 2003 3:42 PM
>>>Subject: Re: [Asterisk-Users] Call Transfer
>>>
>>>
>>>      
>>>
>>>>Hi,
>>>>
>>>>It works if you put the 'T' switch in the dial line.
>>>>
>>>>You can then transfer the call from the caller.
>>>>I have tested it in the folllowing configuration and it works:
>>>>Call from a Cisco 7960 to an ATA 186.
>>>>Select 'Transfer" on 7960
>>>>Call another extension (X-Lite)
>>>>Select again transfer on 7960.
>>>>The call remain between ATA and X-Lite.
>>>>
>>>>This is what you need?
>>>>
>>>>BR,
>>>>Dan
>>>>
>>>>----- Original Message -----
>>>>From: "Chee Foong" <cheefoong at inovas.com>
>>>>To: <asterisk-users at lists.digium.com>
>>>>Sent: Wednesday, July 30, 2003 7:08 AM
>>>>Subject: [Asterisk-Users] Call Transfer
>>>>
>>>>
>>>>Hello all,
>>>>
>>>>I am in a situation where I need to use asterisk to call someone say
>>>>        
>>>>
>>>Party
>>>      
>>>
>>>>A. After the call to Party A got through, asterisk will put Party A on
>>>>        
>>>>
>>>hold,
>>>      
>>>
>>>>then asterisk will call Party B. If call to Party B got through,
>>>>        
>>>>
>asterisk
>  
>
>>>>will transfer Party A to Party B.
>>>>
>>>>I wonder if this features is implemented into asterisk. I have found a
>>>>        
>>>>
>>>post
>>>      
>>>
>>>>in asterisk mailing list:
>>>>http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
>>>>
>>>>but that doesn't help much.
>>>>
>>>>If this features is not implemented, can anyone give me some point on
>>>>        
>>>>
>how
>  
>
>>>to
>>>      
>>>
>>>>implement this in asterisk? Do I need to write an app like the Dial
>>>>        
>>>>
>apps
>  
>
>>>for
>>>      
>>>
>>>>asterisk to load at start up?
>>>>
>>>>
>>>>thanks
>>>>
>>>>Foong
>>>>
>>>>
>>>>_______________________________________________
>>>>Asterisk-Users mailing list
>>>>Asterisk-Users at lists.digium.com
>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>>        
>>>>
>>>_______________________________________________
>>>Asterisk-Users mailing list
>>>Asterisk-Users at lists.digium.com
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>      
>>>
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>>
>>    
>>
>
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