Re: [asterisk-users] asterisk go to holiday extension but hoiday is not defined
what is the pipe before jul I don't understand the utility of that | from the doc. On 01/05/2012 17:15, Joseph wrote: When a call comes in asterisk is forwarding the call to holiday extension, even though the holiday is not defined. Here is my dial plan exten = 4,1,GotoIfTime(*,*,1,jan?holiday,s,1) ; new years day exten = 4,n,GotoIfTime(*,*,6,apr?holiday,s,1) ; easter holiday exten = 4,n,GotoIfTime(*,*,23,may?holiday,s,1) exten = 4,n,GotoIfTime(*,*,1,|jul?holiday,s,1) ; canada day exten = 4,n,GotoIfTime(*,*,1,aug?holiday,s,1) ; long weekend ... Today is May 1, so why is it going to holiday extension? Is there another dial plan that holidays are defined? I'm using asterisk 1.8.10 -- _ -- 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] asterisk go to holiday extension but hoiday is not defined
they say daynames and monthnames are not case-sensitive. the pipe should be used to concatenate several values, it could get the |jul as everything|jul since an empty value before the pipe doesn't have sense? On 01/05/2012 17:23, Doug Lytle wrote: exten = 4,n,GotoIfTime(*,*,1,|jul?holiday,s,1) ; canada day I don't use gotoiftime for holiday matching, but the COMMA PIPE stands out on your example. Doug -- _ -- 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
[asterisk-users] AMI Originate double call
Hi, probably is a problem already solved, but I cannot find a solution
anywhere.
so:
I tried to connect to Asterisk AMI using php and telnet, but the problem
is there anyway.
1. I call just 18 and a playback start.
2. then open a telnet connection and authenticate
3. originate one new call
4. two calls are originated ???
you could see it in the 4th and 5th line of asterisk cli:
-- Executing [play@system:1] Answer(Local/play@system-763a;2, )
in new stack
-- Executing [play@system:1] Answer(Local/play@system-763a;1, )
in new stack
someone know how to solve?
thanks.
extensions.conf:
[from-sip]
exten = 18,1,Answer()
exten = 18,n,PlayBack(catania)
exten = 18,n,Hangup()
[system]
exten = play,1,Answer()
exten = play,n,Set(__destinatario=${destinatario})
exten = play,n,Dial(Local/in@system,3,A(beep)L(3000))
exten = play,n,Hangup()
exten = in,1,Answer()
exten = in,n,ChanSpy(${destinatario},qsWE)
exten = in,n,Hangup()
telnet console:
Action: Originate
Channel: Local/play@system
Variable: destinatario=SIP/Work-0001
Async: true
Response: Success
Message: Originate successfully queued
asterisk cli:
== Using SIP RTP CoS mark 5
-- Executing [18@from-sip:1] Answer(SIP/Work-0001, ) in
new stack
-- Executing [18@from-sip:2] Playback(SIP/Work-0001,
catania) in new stack
-- SIP/Work-0001 Playing 'catania.gsm' (language 'en')
-- Executing [play@system:1] Answer(Local/play@system-763a;2,
) in new stack
-- Executing [play@system:1] Answer(Local/play@system-763a;1,
) in new stack
-- Executing [play@system:2] Set(Local/play@system-763a;1,
__destinatario=SIP/Work-0001) in new stack
-- Executing [play@system:3] Dial(Local/play@system-763a;1,
Local/in@system,3,A(beep)L(3000)) in new stack
-- Setting call duration limit to 3.000 seconds.
-- Called Local/in@system
-- Executing [in@system:1] Answer(Local/in@system-9d0b;2, )
in new stack
-- Local/in@system-9d0b;1 answered Local/play@system-763a;1
-- Local/in@system-9d0b;1 Playing 'beep.gsm' (language 'en')
-- Executing [in@system:2] ChanSpy(Local/in@system-9d0b;2,
SIP/Work-0001,qsWE) in new stack
== Spying on channel SIP/Work-0001
-- Executing [play@system:2] Set(Local/play@system-763a;2,
__destinatario=SIP/Work-0001) in new stack
-- Executing [play@system:3] Dial(Local/play@system-763a;2,
Local/in@system,3,A(beep)L(3000)) in new stack
-- Setting call duration limit to 3.000 seconds.
-- Called Local/in@system
-- Local/play@system-763a;2 requested special control 20,
passing it to Local/in@system-9c17;1
-- Executing [in@system:1] Answer(Local/in@system-9c17;2, )
in new stack
-- Local/in@system-9c17;1 answered Local/play@system-763a;2
-- Local/in@system-9c17;1 Playing 'beep.gsm' (language 'en')
-- Executing [in@system:2] ChanSpy(Local/in@system-9c17;2,
SIP/Work-0001,qsWE) in new stack
== Spying on channel SIP/Work-0001
== Spawn extension (system, play, 3) exited non-zero on
'Local/play@system-763a;1'
== Spawn extension (system, play, 3) exited non-zero on
'Local/play@system-763a;2'
== Done Spying on channel SIP/Work-0001
-- Stopped spying due to the spied-on channel hanging up.
== Spawn extension (system, in, 2) exited non-zero on
'Local/in@system-9d0b;2'
== Done Spying on channel SIP/Work-0001
-- Stopped spying due to the spied-on channel hanging up.
== Spawn extension (system, in, 2) exited non-zero on
'Local/in@system-9c17;2'
== Spawn extension (from-sip, 18, 2) exited non-zero on
'SIP/Work-0001'
--
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Re: [asterisk-users] concurrent channels limit
no, it's a set of script that I'm supposed to update. However the result will be similar. On 02/04/2012 17:42, Israel Gottlieb wrote: are you by chance using the a2billing script? -- _ -- 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] concurrent channels limit
No, I don't do transcoding, I've disabled all the codec except for the g729. But in my last test I've found out what is the problem (not yet how to solve it) I make all my calls through a php agi, this old script works well on asterisk 1.4 and I want to move on 1.8. Just for test I've created three different (simplest) scripts: 1 - stream a file codified in g729 2 - make some mysql queries and stream the file 3 - make an http hit and stream the file I stream an audio file to create calls that last some minute and test also the audio quality, I don't know if there's a better way. Anyway, if I use one of this 3 agi (also randomly) I'm able to establish up to 2500 channels with a perfect audio. If I use my old agi I could establish just 74 channels. I'm going mad on this because the number is not variable, is not one time 80 and the other 70 and sometimes 88, it's always 74. The old agi script is a little longer than my test scripts, but it make the same things. I could accept the loss of some channels, but from 2500 to 74 there is a difference a little too big. On 02/04/2012 00:37, Matt Riddell wrote: How many g729 licenses do you have? You sure you're not transcoding? -- _ -- 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] Limit Call ?
have you tried the L parameter in the dial command? * *L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. Numbers must be integers- beware of AGI scripts that may return long integers in scientific notation (esp PHP 5.2.56) The following special variables are optional for limit calls: (pasted from app_dial.c) o *LIMIT_PLAYAUDIO_CALLER* - yes|no (default yes) - Play sounds to the caller. o *LIMIT_PLAYAUDIO_CALLEE* - yes|no - Play sounds to the callee. o *LIMIT_TIMEOUT_FILE* - File to play when time is up. o *LIMIT_CONNECT_FILE* - File to play when call begins. o *LIMIT_WARNING_FILE* - File to play as warning if 'y' is defined. If *LIMIT_WARNING_FILE* is not defined, then the default behaviour is to announce (You have [XX minutes] YY seconds). http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial On 02/04/2012 16:01, Olivier CALVANO wrote: Thanks but i read: ; The maximum number of concurrent calls you want to allow Not limit the duration of a call ;=) Le 2 avril 2012 16:55, Bakkoasannu...@gmail.com a écrit : Hi, look at maxcalls parameter on the asterisk.conf file. regards El 02/04/2012 16:46, Olivier CALVANO escribió: Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier -- _ -- 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 -- _ -- 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] concurrent channels limit
Asterisk says to process the call correctly: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [17000@sipp:1] Answer(SIP/sipp-005a, ) in new stack -- Executing [17000@sipp:2] Set(SIP/sipp-005a, rn=100) in new stack -- Executing [17000@sipp:3] Goto(SIP/sipp-005a, set100) in new stack -- Goto (sipp,17000,12) -- Executing [17000@sipp:12] Answer(SIP/sipp-005a, ) in new stack -- Executing [17000@sipp:13] BackGround(SIP/sipp-005a, you-seem-impatient) in new stack -- SIP/sipp-005a Playing 'you-seem-impatient.ulaw' (language 'en') -- Executing [17000@sipp:14] Wait(SIP/sipp-0055, 20) in new stack sipp says Aborting call on an unexpected BYE for call: 96-1956@192.168.200.185 asterisk -rx 'core show channels'|tail -n3 shows: 80 active channels- constant 80 active calls- constant 160 calls processed - increase every second the sipp command I use is ./sipp 192.168.200.64 -sn uac -i 192.168.200.185 -s 17000 -d 9 -l 1 -r 100 -rp 3 -t un that generate 100 calls every 30 seconds. every call last 90 seconds. I'm not trying to break the limit of 1 calls, I want just to have 200 or 300 calls. sip does not have setted any limit, and call-limit is deprecated in asterisk 1.8. On 30/03/2012 14:04, Danny Nicholas wrote: Check the sip.conf.sample file. I think it is the call-limit parameter that is getting you. The sample file should tell you what the default is. Another possibility is that your rtp range is set too low; the normal range is 1-2, which allows for 2500 calls(or 5000 if you set other things correctly). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes Sent: Friday, March 30, 2012 7:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] concurrent channels limit -- _ -- 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] concurrent channels limit
Ok, this was a stupid thing (my fault), with -r 1000 I get easily 1000 concurrent calls that terminate in 20 seconds. This calls just answer, play a file the first 2 seconds and then wait. Then sipp close because of two many errors, this is the log: sipp: The following events occured: 2012-03-30-15:17:07:081---1333117027.081757: Discarding message which can't be mapped to a known SIPp call: BYE sip:sipp@192.168.200.185:52281 SIP/2.0^M Via: SIP/2.0/UDP 192.168.200.64:5060;branch=z9hG4bK2a711f73;rport^M Max-Forwards: 70^M From: sut sip:17000@192.168.200.64:5060;tag=as4ad7b2e8^M To: sipp sip:sipp@192.168.200.185:52281;tag=2001SIPpTag0015^M Call-ID: 15-2001@192.168.200.185^M CSeq: 102 BYE^M User-Agent: Asterisk PBX 1.8.11.0^M X-Asterisk-HangupCause: Normal Clearing^M X-Asterisk-HangupCauseCode: 16^M Content-Length: 0^M ^M . 2012-03-30-15:17:07:580---1333117027.580847: Discarding message which can't be mapped to a known SIPp call: BYE sip:sipp@192.168.200.185:52281 SIP/2.0^M Via: SIP/2.0/UDP 192.168.200.64:5060;branch=z9hG4bK2a711f73;rport^M Max-Forwards: 70^M From: sut sip:17000@192.168.200.64:5060;tag=as4ad7b2e8^M To: sipp sip:sipp@192.168.200.185:52281;tag=2001SIPpTag0015^M Call-ID: 15-2001@192.168.200.185^M CSeq: 102 BYE^M User-Agent: Asterisk PBX 1.8.11.0^M X-Asterisk-HangupCause: Normal Clearing^M X-Asterisk-HangupCauseCode: 16^M Content-Length: 0^M ^M . 2012-03-30-15:17:07:982---1333117027.982422: Discarding message which can't be mapped to a known SIPp call: BYE sip:sipp@192.168.200.185:38844 SIP/2.0^M Via: SIP/2.0/UDP 192.168.200.64:5060;branch=z9hG4bK66a86c70;rport^M Max-Forwards: 70^M From: sut sip:17000@192.168.200.64:5060;tag=as43adc103^M To: sipp sip:sipp@192.168.200.185:38844;tag=2001SIPpTag009^M Call-ID: 9-2001@192.168.200.185^M CSeq: 102 BYE^M User-Agent: Asterisk PBX 1.8.11.0^M X-Asterisk-HangupCause: Normal Clearing^M X-Asterisk-HangupCauseCode: 16^M Content-Length: 0^M ^M . 2012-03-30-15:17:08:504---1333117028.504334: Unable to get a UDP socket (3). But if I change the dialplan, remove background and wait functions, add play with a g729 audio file instead, I could do again just 80 concurrent call. On 30/03/2012 14:50, Danny Nicholas wrote: Change --r 100 to --r 300. -- _ -- 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
