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

[Chee Foong]

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
> >>
> >
> >_______________________________________________
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> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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