Try this:
phone1=192.168.7.251
number1="1+0+1"
curl "http://$phone1/command.htm?key=$number1+ENTER" >/dev/null
2>/dev/null
sleep 10
curl "http://$phone2/command.htm?key=CANCEL" >/dev/null
2>/dev/null
Available keys:
#define KEY_CANCEL "CANCEL"
#define KEY_CLEAR "CLEAR"
#define KEY_ENTER "ENTER"
#define KEY_OFFHOOK "OFFHOOK"
#define KEY_ONHOOK "ONHOOK"
#define KEY_RIGHT "RIGHT"
#define KEY_LEFT "LEFT"
#define KEY_FUNCTION "FUNCTION" // below redial
#define KEY_MENU "MENU"
#define KEY_REDIAL "REDIAL"
#define KEY_ORG_F1 "F1"
#define KEY_ORG_F2 "F2"
#define KEY_ORG_F3 "F3"
#define KEY_ORG_F4 "F4"
#define KEY_SPEAKER "SPEAKER"
#define KEY_DISCONN "DISCONNECT"
#define KEY_RECALL "RECALL"
#define KEY_BREAK "BREAK"
#define KEY_TRANSFER "TRANSFER"
#define KEY_CONFERENCE "CONFERENCE"
#define KEY_HELP "HELP"
#define KEY_VOLUME_UP "VOLUME_UP"
#define KEY_VOLUME_DOWN "VOLUME_DOWN"
#define KEY_MUTE "MUTE"
#define KEY_HEADSET "HEADSET"
#define KEY_UP "UP"
#define KEY_DOWN "DOWN"
#define KEY_REC "REC"
#define KEY_RETRIEVE "RETRIEVE"
#define KEY_SETTINGS "SETTINGS"
#define KEY_PHONE_BOOK "PHONE_BOOK"
#define KEY_SNOM "SNOM"
#define KEY_DND "DND"
And yes, it is a good reason to set the password on your phone if you
dont want to use it only for testing! Giving everyone access to the web
server of the phone is not a good idea, not only for snom phones.
CS
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
> Anton Krall
> Sent: Saturday, August 13, 2005 7:46 PM
> To: asterisk-biz@thevoipconnection.com; 'Asterisk Users
> Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Load Testing
>
> Hi Michael.
>
> Are there any script already made for doing this? Sending
> calls from one asterisk to the one been tested? Something
> that would simulate your 1 phone scenario?
>
>
>
> |-----Original Message-----
> |From: asterisk-users-bounces@lists.digium.com
> |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf
> Of The VoIP
> |Connection
> |Sent: Viernes, 12 de Agosto de 2005 10:42 p.m.
> |To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> |Subject: RE: [Asterisk-Users] Load Testing
> |
> |Anton,
> |
> |A great tool for "ghetto" call capacity testing is a single
> snom phone.
> |There is no limit to how many calls a snom phone can make,
> just put it
> |on hold and dial again. So, with a single snom phone and a little
> |imagination you can test any number of scenarios. You can
> approximate
> |basic SIP capacity by creating an extension that plays the asterisk
> |test message and dialing it repeatedly until quality starts
> to degrade
> |or asterisk gives up.
> |To simulate actual call throughput you really need another
> |(faster) machine to connect to, but you can use the same technique.
> |
> |You can run "top" on the console while you are doing your
> tests to see
> |what resources you are using. Check your logs when you are
> done to see
> |what errors were generated when it came unglued. CPU is not
> always the
> |limiting resource, especially with Digium card interfaces
> which tend to
> |be bound by FSB speed, but echo cancellation and codec
> conversion will
> |burn a LOT of cycles.
> |
> |Michael Crown
> |Managing Partner
> |www.thevoipconnection.com
> |321.989.6728 ext. 611
> |sip:611@voiceserver.thevoipconnection.com
> |
> |> -----Original Message-----
> |> From: Anton Krall [mailto:akrall-lists@intruder.com.mx]
> |> Sent: Friday, August 12, 2005 9:56 PM
> |> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> |> Subject: [Asterisk-Users] Load Testing
> |>
> |> Guys.
> |>
> |> How and which tools to use to load test an asterisk install?
> |> Say for example, you need to see how many calls can be routed thru
> |> before losing quality and making the cpu jump to the roof?
> |>
> |>
> |>
> |
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>