Re: [Asterisk-Users] sample.call + chan_h323 gives seg fault
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 exte
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
Re: [Asterisk-Users] sample.call + chan_h323 gives seg fault
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] sample.call
Guys, I have some answer about sample.call 1. Can we use sample.call to test (or simulated) asterisk (in a predetermined scenario) to accept calls simultaneously?. 2. How many calls can be simulated? 3. Can we used the result as a basis on how many simultaneous calls can handled by asterisk? 4. What channel can be tested using this scheme? Regards, Herry Sitepu Clarisense Digital Media www.clarisense.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
