Static IP LinuxMCE Server to Static IP SIP Trunk Provider

[steven.herr]

Hello,

I have started my first attempt at creating a LinuxMCE box.  I am running LinuxMCE-810 RC1.  My difficulty is in attempting to configure FreePBX.  I live in a rural area where I am fortunate enough to have a local provider that will provide a SIP trunk with a local phone number.  Unfortunately, Vianet, my local provider, is not one of the templates.

Vianet provided me with an IP address of their server and said I do not require a username or password because I too have a static IP.  They said all I have to do is send my trunk to their static IP.

I tried to edit /usr/pluto/bin/create_amp_vianet.pl but I couldn't make enough sense out of it.

I am hoping someone could give me a little direction on how to set up a SIP trunk to point to a static IP address that does not require a username or password.

Thank-you in advance,

Steven

[Aviator]

I was able to add support for my provider by following this:
http://wiki.linuxmce.org/index.php/How_to_properly_add_support_for_SIP_providers

It takes some trial and error, but I hope it helps.  Please be sure and open a ticket and attach you provider_list.txt and create_amp_vianet.pl so it can be added into the system.

[fibres]

Hi Steven.herr

You would configure this in the same way you would a sip trunk that uses username/password. Just leave the username/password fields blank or put anything in them.

Also make sure anywhere that talks about a register line you leave this blank.

I am running 1004 which has a non working asterisk so not sure if the files Aviator talks about are on my core. It is currently shutdown as I am going on holiday on friday.

If your still struggling when I get back in a weeks time Ill have a look and try and help you further.

Regards

[steven.herr]

Thanks,

I did try the wiki.  I ultimately discovered I didn't need that approach found in the wiki and simply needed to configure within FreePBX.  My system was actually working and responding to incoming calls by playing the "ss-noservice" message.  Here is a CLI output:


  -- Executing [XXXXXXXXXXX@from-sip-external:1] NoOp("SIP/69.60.226.14-08afe638", "Received incoming SIP connection from unknown peer to XXXXXXXXXX") in new stack
    -- Executing [XXXXXXXXXXX@from-sip-external:2] Set("SIP/69.60.226.14-08afe638", "DID=XXXXXXXXXXX") in new stack
    -- Executing [XXXXXXXXXXX@from-sip-external:3] Goto("SIP/69.60.226.14-08afe638", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/69.60.226.14-08afe638", "0?from-trunk|XXXXXXXXXXX|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/69.60.226.14-08afe638", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2011-10-14 04:26:11 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/69.60.226.14-08afe638", "") in new stack
    -- Executing [s@from-sip-external:4] Wait("SIP/69.60.226.14-08afe638", "2") in new stack
    -- Executing [s@from-sip-external:5] Playback("SIP/69.60.226.14-08afe638", "ss-noservice") in new stack
    -- <SIP/69.60.226.14-08afe638> Playing 'ss-noservice' (language 'en')

Does anyone understand why incoming calls won't go to the IVR which I configured?

Thanks,

Steve

[fibres]

It seems that it is not matching the incomming calll to the peer you createrd in FreePBX.

Check the settings you setup for the peer.

Regards