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

Jeremy McNamara Wed, 30 Jul 2003 20:08:24 -0700

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" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
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/[EMAIL PROTECTED]
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" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
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" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
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
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

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

Reply via email to