Messages

1

Installation issues / Re: Schedule Reboot Core

« on: June 26, 2010, 07:17:38 am »

Thks Sambuca,
Just to make sure the core will not "hang" after long period running...

cyf4746

2

Installation issues / Schedule Reboot Core

« on: June 25, 2010, 12:31:37 pm »

Is anyone know how to set the core to reboot at a specific time, such as "reboot at every Sunday 10.00 am"?

3

Users / Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)

« on: July 15, 2009, 02:19:41 pm »

It looks like setting the regional settings to Trapezoid, 90Volts, and 20Hz is working for most people...wasn't able to find specifics on Malaysia though. 

Hi los93sol,
I missed out something. The analog phone extension not ring. I already go to the wizard-device-phonelines and check "House Line" to ring in all scenario but the analog phone still not ring.
Please help.

4

Users / Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)

« on: July 15, 2009, 07:40:53 am »

It looks like setting the regional settings to Trapezoid, 90Volts, and 20Hz is working for most people...wasn't able to find specifics on Malaysia though. 

Hi los93sol,
I managed to fixed the problem. Apparently it's due to the "disconnect tone" not properly setup in SAP3000. I manage to get Malaysia PSTN tone info and fill in the correct tone. It's work now.
Thanks for your helps in this.

Chin

5

Users / Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)

« on: July 15, 2009, 04:54:46 am »

It looks like setting the regional settings to Trapezoid, 90Volts, and 20Hz is working for most people...wasn't able to find specifics on Malaysia though. 

Hi los93sol,
Here is the status of the asterisk status:

*************************************************************************
Asterisk 1.4.10, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.10 currently running on dcerouter (pid = 6415)
Verbosity is at least 16
    -- Executing [03-6280-9300@from-trunk:1] Set("SIP/House Line-08212700", "__FROM_DID=03-6280-9300") in new stack
    -- Executing [03-6280-9300@from-trunk:2] GotoIf("SIP/House Line-08212700", "1 ?cidok") in new stack
    -- Goto (from-trunk,03-6280-9300,4)
    -- Executing [03-6280-9300@from-trunk:4] NoOp("SIP/House Line-08212700", "CallerID is "PSTN Call" <House Line>") in new stack
    -- Executing [03-6280-9300@from-trunk:5] Goto("SIP/House Line-08212700", "custom-linuxmce|103|1") in new stack
    -- Goto (custom-linuxmce,103,1)
    -- Executing [103@custom-linuxmce:1] AGI("SIP/House Line-08212700", "pluto-gethousemode.agi") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/pluto-gethousemode.agi
    -- AGI Script Executing Application: (Set) Options: (HOUSEMODE=1)
    -- AGI Script pluto-gethousemode.agi completed, returning 0
    -- Executing [103@custom-linuxmce:2] Goto("SIP/House Line-08212700", "103-hm1|1") in new stack
    -- Goto (custom-linuxmce,103-hm1,1)
    -- Executing [103-hm1@custom-linuxmce:1] Dial("SIP/House Line-08212700", "Local/200@trusted|15") in new stack
    -- Called 200@trusted
    -- Executing [200@trusted:1] Macro("Local/200@trusted-7519,2", "exten-vm|novm|200") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("Local/200@trusted-7519,2", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] NoOp("Local/200@trusted-7519,2", "user-callerid: PSTN Call House Line") in new stack
    -- Executing [s@macro-user-callerid:2] Set("Local/200@trusted-7519,2", "AMPUSER=House Line") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("Local/200@trusted-7519,2", "1?report") in new stack
    -- Goto (macro-user-callerid,s,13)
    -- Executing [s@macro-user-callerid:13] NoOp("Local/200@trusted-7519,2", "TTL:  ARG1: novm") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("Local/200@trusted-7519,2", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:15] Set("Local/200@trusted-7519,2", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:16] GotoIf("Local/200@trusted-7519,2", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,23)
    -- Executing [s@macro-user-callerid:23] NoOp("Local/200@trusted-7519,2", "Using CallerID "PSTN Call" <House Line>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("Local/200@trusted-7519,2", "FROMCONTEXT=exten-vm") in new stack
    -- Executing [s@macro-exten-vm:3] Set("Local/200@trusted-7519,2", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("Local/200@trusted-7519,2", "EXTTOCALL=200") in new stack
    -- Executing [s@macro-exten-vm:5] Set("Local/200@trusted-7519,2", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("Local/200@trusted-7519,2", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("Local/200@trusted-7519,2", "RT=") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("Local/200@trusted-7519,2", "record-enable|200|IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("Local/200@trusted-7519,2", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("Local/200@trusted-7519,2", "recordingcheck|20090714-193153|1247625113.66") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck
  recordingcheck|20090714-193153|1247625113.66: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] NoOp("Local/200@trusted-7519,2", "No recording needed") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("Local/200@trusted-7519,2", "dial||tr|200") in new stack
    -- Executing [s@macro-dial:1] GotoIf("Local/200@trusted-7519,2", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("Local/200@trusted-7519,2", "dialparties.agi") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Starting New Dialparties.agi
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  dialparties.agi: Caller ID name is 'PSTN Call' number is 'House Line'
  dialparties.agi: USE_CONFIRMATION:  'FALSE'
  dialparties.agi: RINGGROUP_INDEX:   ''
  dialparties.agi: Methodology of ring is  'none'
    --  dialparties.agi: Added extension 200 to extension map
    --  dialparties.agi: Extension 200 cf is disabled
    --  dialparties.agi: Extension 200 do not disturb is disabled
       >  dialparties.agi: extnum 200 has:  cw: 0; hascfb: 0 [] hascfu: 0 []
       >  dialparties.agi: ExtensionState: 0
  dialparties.agi: Extension 200 has ExtensionState: 0
    --  dialparties.agi: Checking CW and CFB status for extension 200
    --  dialparties.agi: DbDel CALLTRACE/200 - Caller ID is not defined
  == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:10] Dial("Local/200@trusted-7519,2", "SIP/200||tr") in new stack
    -- Called 200
    -- Local/200@trusted-7519,1 is ringing
    -- SIP/200-0821b638 is ringing
    -- Nobody picked up in 15000 ms
    -- Executing [103-hm1@custom-linuxmce:2] Goto("SIP/House Line-08212700", "103-hm1-NOANSWER|1") in new stack
    -- Goto (custom-linuxmce,103-hm1-NOANSWER,1)
    -- Executing [103-hm1-NOANSWER@custom-linuxmce:1] Goto("SIP/House Line-08212700", "voice-menu-pluto-custom|s|1") in new stack
    -- Goto (voice-menu-pluto-custom,s,1)
    -- Executing [s@voice-menu-pluto-custom:1] Answer("SIP/House Line-08212700", "") in new stack
    -- Executing [s@voice-menu-pluto-custom:2] Wait("SIP/House Line-08212700", "1") in new stack
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/200@trusted-7519,2' in macro 'dial'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/200@trusted-7519,2' in macro 'exten-vm'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/200@trusted-7519,2'
    -- Executing [h@macro-dial:1] Macro("Local/200@trusted-7519,2", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("Local/200@trusted-7519,2", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("Local/200@trusted-7519,2", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("Local/200@trusted-7519,2", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("Local/200@trusted-7519,2", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("Local/200@trusted-7519,2", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("Local/200@trusted-7519,2", "") in new stack
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/200@trusted-7519,2' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/200@trusted-7519,2'
    -- Executing [s@voice-menu-pluto-custom:3] AGI("SIP/House Line-08212700", "pluto-callersforme.agi") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/pluto-callersforme.agi
    -- AGI Script Executing Application: (NoOp) Options: (Finding if  is a priority call for somebody)
    -- AGI Script pluto-callersforme.agi completed, returning 0
    -- Executing [s@voice-menu-pluto-custom:4] BackGround("SIP/House Line-08212700", "pluto/pluto-default-voicemenu") in new stack
    -- <SIP/House Line-08212700> Playing 'pluto/pluto-default-voicemenu' (language 'en')
    -- Executing [s@voice-menu-pluto-custom:5] Set("SIP/House Line-08212700", "TIMEOUT(digit)=10") in new stack
    -- Digit timeout set to 10
    -- Executing [s@voice-menu-pluto-custom:6] Set("SIP/House Line-08212700", "TIMEOUT(response)=20") in new stack
    -- Response timeout set to 20
dcerouter*CLI>

6

Users / Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)

« on: July 14, 2009, 03:40:14 pm »

If you're still having trouble after that let me know and I can provide some remote assistance to help you get this going.

Hi los93sol,
I manage to get the MD alert when incoming call. But there are 2 problems here:
1) The MD don't drop call when I disconnect call from my mobile
2) The analog phone don;t ring when incoming call.

Are there due to my regional ringging voltage setting? But, sadly. I am not able to find what is my country's ringging voltage. I did call up Malaysia telecom, but they failed to give me a correct answer.
And, how to enable remote assistance? From the forum i know the LMCE remote access server is currently ddown. Please advice.

7

Users / Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)

« on: July 14, 2009, 03:57:13 am »

Did you attempt making a call to your house while the Asterisk CLI was open?  It doesn't appear the call is ever even making it to your asterisk box which probably means your Sipura is not configured correctly.  Verify that your settings are correct and that you Sipura is on your network and double check that your trunk is configured correctly in FreePBX.

If you need additional help beyond this point I can assist remotely, but will need access to your box and your Sipura.

Hi los93sol ,
I will redo everything again. Will let you know the status.
Thanks for your advice.

Chin

8

Users / Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)

« on: July 13, 2009, 06:09:38 am »

Did you verify that the calls are not making it to your linuxmce core by using the asterisk cli as I suggested or are you just listening?  You need to be more specific in exactly what is happening for me to help you troubleshoot.

I did as you told. And here is the result from terminal:
************************************************************************
Asterisk 1.4.10, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.10 currently running on dcerouter (pid = 7251)
    -- Remote UNIX connection
Verbosity is at least 16
dcerouter*CLI>
***************************************************************************************************
On second though. Is the problems due to my regional setting in SPA3000? I am in Malaysia.

9

Users / Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)

« on: July 11, 2009, 09:05:30 am »

First fire up asterisk cli with asterisk -vvvvvvvvvvvvvvvvr and make sure that incoming calls are even being routed to the analog extension.  If they are then check the ring voltage settings on the Sipura and make sure they are set appropriately for your country.

The incoming call not working. Both MD and analog phone not ringing when I did a test call from my mobile. Form my mobile I hear 2 ringing tone then the call disconnected.
The outgoing call workd but with some problem. The call made to my mobile didn't hang-up / drop when I hang up the call after receiving it.

Please help.

10

Users / Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)

« on: July 10, 2009, 02:42:21 pm »

Hi,
I install Sipura 3000 with an analog phone plug-in. It works in outgoing call with some problem. I made successfull call to my mobile from a MD. But the call not cutt off or drop even I have disconnect the call from my mobile. And, my analog phone don't ring when there is an incoming call to my phone line.
Please help.

11

Users / Re: IP Camera Pan/Tilt Functions

« on: April 28, 2009, 04:17:57 pm »

Thom,
It's too expensive for me. I didn;t mean your effort didn't worth it. I just thought I can get this cameras functions work and share with others... Thanks anyway...

12

Users / Re: IP Camera Pan/Tilt Functions

« on: April 27, 2009, 03:27:23 am »

Thom,
If I sent the IP camera to you. Do you have time to integrate all the functions into LMCE?

13

Users / Re: IP Camera Pan/Tilt Functions

« on: April 26, 2009, 10:51:13 am »

Thom,
How about the audio functions? Panasonic template don;t have the audio functions right?

14

Users / Re: iPhone/iPod Touch Orbiter?

« on: April 25, 2009, 10:16:58 am »

Any plans for Orbiter for Win CE 5.0 and 6.0? Please take note I mean WinCE not Win Mobile? The existing Orbiter download from web admin is not work for Win CE 5.0 and 6.0. And, the WEBPAD 366 orbiter develop by Thom is a good choice for users. But the device already face off from the manufacturer.  Why don;t we develop an Orbiter for most of the MID can use? So that users from Iphone, Winmobile, WinCE, and ArmLinux can use it. Just a suggestion....  

15

Users / IP Camera Pan/Tilt Functions

« on: April 25, 2009, 10:09:28 am »

I got an IP camera from China and it is with pan/tilt functions and audio fucntions. I can get it work in LMCE but the pan/tilt and audio functions not working. If i got the SDK from the manufacturer. How to configure it in LMCE? Please help!