Author Topic: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)  (Read 17210 times)

cyf4746

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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

los93sol

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

cyf4746

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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.

los93sol

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For remote assistance it would be through giving telnet access and web admin access temporarily to your box rather than through the Remote Assistance feature of LMCE.

Now that your MD is ringing you should fire up the Asterisk CLI again and call your home.  You should see in the CLI where your MD's extension is ringing and your analog extension.  If you see the analog extension ringing now then we need to figure out your ring voltage.  I had to adjust mine to get a normal ring on my phones since my phone almost sounded like it was trying to do a distinctive ring since the voltage was set too high by default.

I will do some digging and see if I can figure out anything for you as to how this should be set.

los93sol

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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. 

cyf4746

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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>

cyf4746

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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

cyf4746

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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.

pw44

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #98 on: March 06, 2010, 10:45:15 pm »
Hia,
i'm trying to setup the pstn analog line using the linksys spa-3102, using as reference the
http://wiki.linuxmce.org/index.php/Sipura/Linksys_spa3000_pstn_interface page.
But when following the following section, i got trouble:
Code: [Select]
Next, go to Wizard->Devices->Phone Lines (on the left pane in the LMCE web admin, not in FreePBX). We are going to add a dummy line (NOTE: this is a temporary hack for now! I won't go into too many details other than saying that it will allow you to use the "Settings" link next to the listing to do some call routing on your pstn line!) Use the dropdown to select broadvoice. Once you do this, you will see a form to fill in some data. Just put whatever you want in the fields, they won't be used with this hack! After you are done, you will see it listed as a phone line - use the "settings" link next to it to do call routing depending on security mode! (Note: After creating this "dummy" phone line, go back to the FreePBX admin, and look at the Incoming Route for the broadvoice line. Look at the Custom App option at the bottom of the page. It should contain that same custom-linuxmce,102,1. If it does not, go back to the Incoming Route for the House Line, and change its custom app line to be the same as this one! From my tests, it should be the same (though the 3 digit number can change, so check this to be sure).
Using the settings lik next to to it to do call routing... there is no Custom App option.
As this wiki was modified jan 26, 2009, i guess there were changes.
Does anyone knows how to use the spa-3102 with analog lines only and having the pap2t-na as extensions (using analog phone devices as softphones)?
TIA,
Paulo