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
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] 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
