Re: [Asterisk-Users] How to make Asterisk to generate and terminate calls
After some search in wiki I was able to do what I wanted. Here is how it is, The .call file should appear something like this and it has to be placed in /var/spool/asterisk/outgoing of asterisk-1, Channel: local/[EMAIL PROTECTED] ; Any extension can be called using local/extension@context MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: sip Extension: 2001 Priority: 1 In asterisk-1 we should have the following entries in extensions.conf file, [sip] exten = 3001,1,MyOriginateScript() exten = 3001,2,Hangup In asterisk-2 we should have the following entries in extensions.conf file, [sip] exten = 2001,1,MyTerminateScript() exten = 2001,2,Hangup We can do whatever we want in our MyoriginateScript/MyTerminateScript. The features provided by asterisk is simply amazing !!! Long live asterisk cheers, Ravi Ravi Shankar wrote: Shawn, Thanks for info that would solve the problem of manually making calls and connecting the phones at the either ends. But my requirement is slightly different. I've the following .call file in the /var/spool/asterisk/outgoing directory of asterisk-1 asterisk-1 - SIP - asterisk-2 Channel: SIP/3001 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: sip Extension: Priority: 1 So Asterisk-1 bridges 3001 and (which is in asterisk-2). Since is the terminating side I can have an AGI script handle the call and do whatever I wanted and I don't need a real IP Phone. On the other hand on the originating side 3001 has to be a real SIP Phone. My question is on the originating side, can a AGI script answer the call instead of real IP Phone. This way I can simulate multiple IP Phones without having them physically available. I know this is not the intended usage of asterisk but it would serve to test bulk deployments and find out the capacity of the asterisk without having so many real phones. thanks, Ravi Shawn Porter wrote: Ravi, Take a look here. http://www.voip-info.org/wiki-Asterisk+auto-dial+out I would think that for what you are doing use a cron job and a shell script. Shawn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ravi Shankar Sent: Friday, December 23, 2005 8:41 AM To: Asterisk Users Subject: [Asterisk-Users] How to make Asterisk to generate and terminatecalls Hi, I would like to connect two linux machines running asterisk and then originate SIP calls from one asterisk and terminate it on the other asterisk. Terminating the call is not a problem because I can give the call handle to say AGI application on the terminating asterisk. How do i originate a call from the asterisk ? Is this possible using AGI ? Any pointers in this regard would be of great help. This type of application can be used two simulate bulk calls and find out what is the maximum limit for the asterisk in terms of CPU utilization, memory, etc. before it can be deployed in production environment. thanks, Ravi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to make Asterisk to generate and terminate calls
Hi, I would like to connect two linux machines running asterisk and then originate SIP calls from one asterisk and terminate it on the other asterisk. Terminating the call is not a problem because I can give the call handle to say AGI application on the terminating asterisk. How do i originate a call from the asterisk ? Is this possible using AGI ? Any pointers in this regard would be of great help. This type of application can be used two simulate bulk calls and find out what is the maximum limit for the asterisk in terms of CPU utilization, memory, etc. before it can be deployed in production environment. thanks, Ravi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make Asterisk to generate and terminate calls
This is the how long is a piece of string question. It all depends on the hardware Asterisk sits on, the codecs in use, the dialtone provider (SIP vs IAX vs T1/E1) etc. Do a wiki search and you'll find some examples of what folks have found. As for originate on one and terminat on another; thats doable. Your phone device will have settings for an outbound proxy. Set this for the outbound Asterisk server. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Ravi Shankar wrote: Hi, I would like to connect two linux machines running asterisk and then originate SIP calls from one asterisk and terminate it on the other asterisk. Terminating the call is not a problem because I can give the call handle to say AGI application on the terminating asterisk. How do i originate a call from the asterisk ? Is this possible using AGI ? Any pointers in this regard would be of great help. This type of application can be used two simulate bulk calls and find out what is the maximum limit for the asterisk in terms of CPU utilization, memory, etc. before it can be deployed in production environment. thanks, Ravi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make Asterisk to generate and terminate calls
Shawn, Thanks for info that would solve the problem of manually making calls and connecting the phones at the either ends. But my requirement is slightly different. I've the following .call file in the /var/spool/asterisk/outgoing directory of asterisk-1 asterisk-1 - SIP - asterisk-2 Channel: SIP/3001 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: sip Extension: Priority: 1 So Asterisk-1 bridges 3001 and (which is in asterisk-2). Since is the terminating side I can have an AGI script handle the call and do whatever I wanted and I don't need a real IP Phone. On the other hand on the originating side 3001 has to be a real SIP Phone. My question is on the originating side, can a AGI script answer the call instead of real IP Phone. This way I can simulate multiple IP Phones without having them physically available. I know this is not the intended usage of asterisk but it would serve to test bulk deployments and find out the capacity of the asterisk without having so many real phones. thanks, Ravi Shawn Porter wrote: Ravi, Take a look here. http://www.voip-info.org/wiki-Asterisk+auto-dial+out I would think that for what you are doing use a cron job and a shell script. Shawn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ravi Shankar Sent: Friday, December 23, 2005 8:41 AM To: Asterisk Users Subject: [Asterisk-Users] How to make Asterisk to generate and terminatecalls Hi, I would like to connect two linux machines running asterisk and then originate SIP calls from one asterisk and terminate it on the other asterisk. Terminating the call is not a problem because I can give the call handle to say AGI application on the terminating asterisk. How do i originate a call from the asterisk ? Is this possible using AGI ? Any pointers in this regard would be of great help. This type of application can be used two simulate bulk calls and find out what is the maximum limit for the asterisk in terms of CPU utilization, memory, etc. before it can be deployed in production environment. thanks, Ravi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
