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