Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces
Thanks Steve, I got the picture :) THANK!!! But my doubt is about the cable, what cable should i use? i have a Sangoma A108D in one machine (one machine with one card). What cable should i do? how can i make it? Best Regards! 2010/10/5 Steve Murphy m...@parsetree.com On Tue, Oct 5, 2010 at 1:02 PM, Danny Dias ing.diasda...@gmail.comwrote: Hello my friend Ingmar, I would like to know the cable you used? how was the connection? i'm using this one: http://wiki.sangoma.com/Pinouts#A108 Loop Back Is this ok? what should i do my friend, my problems are understand the fisicall connection :( Best Regards!!! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge *Van:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD I set up two machines with T1 interfaces, and connected the two with an appropriate t1 cable. One was acting as a network (master), the other as a subscriber (slave) (for timing). wrote two dialplans, one for each machine, that would answer an incoming call on one dahdi line, and call to the next numbered line on the other machine. The other machine was similarly outfit. I'd define the extension for the first line on the t1, and call it with any phone you desire. That call will cascade into 23 separate interlinked calls. If you are clever, the last call in should dial another real phone you have on-hand. You get the picture... right? Phone A dials the exten to call the first exten on the other machine. The dialplan should use the first channel on the t1 to place a call to the first exten on the other machine. On the other machine, the incoming call on channel 1 is answered, and then a dial to the second extension on the first machine, over the 2nd t1 channel. The first machine answers, and uses the 3rd channel to call the other machine and so on till all channels are being used. The last exten answers and dials a phone (dahdi or SIP, no matter) that you pick up. Such a looped call should probably be awful, but it's going thru 23 t1 channels! If you have two t1 interaces in a single card (or two cards), then you do this on one machine. Another approach: set up equal numbers of FZO and FXS lines, and similarly loop s single call thru all the channels.This would require just one machine. Other approaches would involve running multiple threads to call an extension and then hang up and repeating this over and over again on all channels to ascertain the load placed just by call setup and tear-down. This kind of load is different than when all lines are just shoveling data back and forth. Watch your load averages, your %cpu, your swap, etc, as the tests are in full swing. murf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Murphy ParseTree Corp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces
The Loop Back Plug on the link you provided is correct. You take a few inches of CAT5 and remove the outer jacket. Loop the wires into the RJ-45 connector like the diagram shows and then crimp. Ryan On Tue, Oct 5, 2010 at 3:02 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friend Ingmar, I would like to know the cable you used? how was the connection? i'm using this one: http://wiki.sangoma.com/Pinouts#A108 Loop Back Is this ok? what should i do my friend, my problems are understand the fisicall connection :( Best Regards!!! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Dias Verzonden: vrijdag 24 september 2010 11:05 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces
Hello my friend Ingmar, I would like to know the cable you used? how was the connection? i'm using this one: http://wiki.sangoma.com/Pinouts#A108 Loop Back Is this ok? what should i do my friend, my problems are understand the fisicall connection :( Best Regards!!! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge *Van:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces
On Tue, Oct 5, 2010 at 1:02 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friend Ingmar, I would like to know the cable you used? how was the connection? i'm using this one: http://wiki.sangoma.com/Pinouts#A108 Loop Back Is this ok? what should i do my friend, my problems are understand the fisicall connection :( Best Regards!!! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge *Van:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD I set up two machines with T1 interfaces, and connected the two with an appropriate t1 cable. One was acting as a network (master), the other as a subscriber (slave) (for timing). wrote two dialplans, one for each machine, that would answer an incoming call on one dahdi line, and call to the next numbered line on the other machine. The other machine was similarly outfit. I'd define the extension for the first line on the t1, and call it with any phone you desire. That call will cascade into 23 separate interlinked calls. If you are clever, the last call in should dial another real phone you have on-hand. You get the picture... right? Phone A dials the exten to call the first exten on the other machine. The dialplan should use the first channel on the t1 to place a call to the first exten on the other machine. On the other machine, the incoming call on channel 1 is answered, and then a dial to the second extension on the first machine, over the 2nd t1 channel. The first machine answers, and uses the 3rd channel to call the other machine and so on till all channels are being used. The last exten answers and dials a phone (dahdi or SIP, no matter) that you pick up. Such a looped call should probably be awful, but it's going thru 23 t1 channels! If you have two t1 interaces in a single card (or two cards), then you do this on one machine. Another approach: set up equal numbers of FZO and FXS lines, and similarly loop s single call thru all the channels.This would require just one machine. Other approaches would involve running multiple threads to call an extension and then hang up and repeating this over and over again on all channels to ascertain the load placed just by call setup and tear-down. This kind of load is different than when all lines are just shoveling data back and forth. Watch your load averages, your %cpu, your swap, etc, as the tests are in full swing. murf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Murphy ParseTree Corp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces
Hi DD, We usually use loopback cables and use the open source SIP test tool SIPp to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Dias Verzonden: vrijdag 24 september 2010 11:05 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces
ummm but how do you do that? SIPp is only for SIP calls...i need to check in some way the dahdi driver, i need in someway stress de card, is that possible? may be it has no sence at all :( Thanks! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge *Van:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces
As the previous poster said use the sip software to make test calls. Have the number it dials go out of the sangoma card and back into another port via a crossover cable to an extension which answers and plays back a file for a second or so before hanging up. You can then make lots of calls which constantly make outgoing calls on 4 ports and incoming calls on another 4 ports. By being able to change the diration of the call to can load the box very well. Danny Dias wrote: ummm but how do you do that? SIPp is only for SIP calls...i need to check in some way the dahdi driver, i need in someway stress de card, is that possible? may be it has no sence at all :( Thanks! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl mailto:i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge *Van:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces
Garet, MANY thanks my friend...can you believe that my brain was stucked :( So simple ;) THANKS for your valuable help! DD 2010/9/24 Gareth Blades list-aster...@skycomuk.com As the previous poster said use the sip software to make test calls. Have the number it dials go out of the sangoma card and back into another port via a crossover cable to an extension which answers and plays back a file for a second or so before hanging up. You can then make lots of calls which constantly make outgoing calls on 4 ports and incoming calls on another 4 ports. By being able to change the diration of the call to can load the box very well. Danny Dias wrote: ummm but how do you do that? SIPp is only for SIP calls...i need to check in some way the dahdi driver, i need in someway stress de card, is that possible? may be it has no sence at all :( Thanks! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl mailto:i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge *Van:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
