Author Topic: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)  (Read 16680 times)

tschak909

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #75 on: March 04, 2009, 10:05:53 am »
You make device templates for them.

Look at the device templates for the existing phones. You'll find configuration and setup scripts for them, which basically do all the manual work of setting up the extension, loading any firmware required, setting settings on the phone automatically, basically anything you need to do to make the installation automatic.

Couple this with pnp records corresponding to the manufacturer MAC addresses for the phone (not an easy thing to find, you often have to guess, or contact the manufacturer), and you're set. Often you can infer possible device ranges if you have more than one of the unit, and eyeball the mac addresses.

-Thom

jondecker76

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #76 on: March 05, 2009, 12:53:22 am »
Thom - while on the subject...  What detection methods are available for PCI cards?


tschak909

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Re: Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)
« Reply #77 on: March 05, 2009, 02:18:46 am »
Basically, we have one that we normally use. We talk to HAL.

HAL provides a device map of everything in the system, and we use this to detect PCI, and USB devices currently. There is a special case where we search for entire categories of devices, such as generic devices like optical disk drives. This is why when we had the HAL bug in 0810 a month ago, almost all pnp detection was affected.

But basically, the DHCPDevice table (which is what is filled when you fill a pnp section of a device template) acts as a filter. As devices are detected, they are sent a whole set of parameters as part of the event, some of them are filled, some are not..such as PCI or USB vendor ID/Model #, MAC address, etc... And the entries in the DHCPDevice table are matched against the incoming event as a conditional, anything matching is returned. If one device is returned, it merely asks to add the device. If multiple devices are returned, then a screen asking the user to pick a device is shown. The user selects a device, and the installation continues.

But technically, anything can fire a New Device Detected event, with a device template, to get the system to go into the routine of asking about and subsequently installing the device.

-Thom

dlewis

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Sure, I am planning on doing just that, I was hoping someone would run through the instructions I provided first so we can determine what changes if any need to go in.  I wrote these instructions from memory of my install a few months ago.

los93sol, any updates on the x100p integration? How about the wiki site?

los93sol

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No updates, I have been waiting for confirmation that my directions posted in this thread have worked for others.  Once I have that I will do a wiki, just don't want to put the information out there all willy nilly if it causes problems for people.  I wrote from memory of what I had done several months earlier.

dlewis

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No updates, I have been waiting for confirmation that my directions posted in this thread have worked for others.  Once I have that I will do a wiki, just don't want to put the information out there all willy nilly if it causes problems for people.  I wrote from memory of what I had done several months earlier.

Are you currently using the x100p?

los93sol

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I switched to the Sipura but still have the x100p installed in my system just disabled for now.

dlewis

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how's the sipura? Any echo?

gonzamen

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Hi everyone, last friday I've received a SPA3102 and during the weekend following jon's wiki I was able to make asterisk work with my PSTN for incoming calls but not for outgoing ones.
This is what I see in Asterisk's log.

05      05/31/09 21:33:53.594           NotifyRing from channel SIP/202-cc001fc0, callerid 202 to channel SIP/House Line-0080dec0, callerid  <0x41802950>
05      05/31/09 21:33:53.618           NotifyHangup channel SIP/House Line-0080dec0, reason Unallocated (unassigned) number <0x41802950>

Any help please?

Thanks,
Gonzalo

cyf4746

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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.

los93sol

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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.

cyf4746

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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.

los93sol

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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.

cyf4746

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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.

los93sol

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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.