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

[Chee Foong]

Hi Jeremy,

Ok, still learning to get the backtrace. will post a trace next.

When I issues a dial command on console, ex

dial H323/6031334000

I get seg fault also, this only happen if it involve dialing through H323
channels

Thank for your reply

Foong

----- Original Message -----
From: "Jeremy McNamara" <jj at indie.org>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, July 31, 2003 11:07 AM
Subject: Re: [Asterisk-Users] sample.call + chan_h323 gives seg fault


> 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
> >>>
> >>>
> >>_______________________________________________
> >>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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