I'll throw you even a bone if i go a feed the animals in the back... :p

I did love it!!!!!!!!!!!!!!!!!!!

It's all in this wiki: http://wiki.linuxmce.org/index.php/PPTP_server
I created it two years ago.

Append it to /usr/pluto/bin/Network_Firewall.sh

For ppp, you need to enable the protocol 47, only opening port 1723 will not work.
Insert the following iptables rules:

iptables --append FORWARD -o ppp+ --protocol tcp --tcp-flags SYN,RST SYN --jump TCPMSS --clamp-mss-to-pmtu
iptables --append INPUT  --protocol 47 --jump ACCEPT
iptables --append OUTPUT --protocol 47 --jump ACCEPT

Ja, mein Kommandant! Werde da nachschauen!

Hi,
my current dialplan is:

mysql> select * from extensions where context='outbound-allroutes'
    -> ;
+------+--------------------+-------+----------+-------+---------------------------------------------+
| id   | context            | exten | priority | app   | appdata                                     |
+------+--------------------+-------+----------+-------+---------------------------------------------+
| 5936 | outbound-allroutes | 190   |        1 | Macro | dialout-trunk,/,${EXTEN},,                  |
| 5937 | outbound-allroutes | 190   |        2 | Macro | outisbusy,                                  |
| 5938 | outbound-allroutes | 193   |        1 | Macro | dialout-trunk,/,${EXTEN},,                  |
| 5939 | outbound-allroutes | 193   |        2 | Macro | outisbusy,                                  |
| 5976 | outbound-allroutes | _7.   |        1 | Macro | dialout-trunk,SIP/054138594676,${EXTEN:1},, |
| 5977 | outbound-allroutes | _7.   |        2 | Macro | outisbusy,                                  |
| 5958 | outbound-allroutes | _8.   |        1 | Macro | dialout-trunk,SIP/2122498618,${EXTEN:1},,   |
| 5959 | outbound-allroutes | _8.   |        2 | Macro | outisbusy,                                  |
| 5940 | outbound-allroutes | _9.   |        1 | Macro | dialout-trunk,SIP/2062036594,${EXTEN:1},,   |
| 5941 | outbound-allroutes | _9.   |        2 | Macro | outisbusy,                                  |
+------+--------------------+-------+----------+-------+---------------------------------------------+

The trunks:
1 sipgate (dialout-trunk,SIP/054138594676)
2 spa3102 (dialout-trunk,SIP/2122498618)
3 voipcheap (dialout-trunk,SIP/2062036594)

The plans (by trunk)
1 - sipgate
dial rules:
00+XXXXXXX.
outbound:
900|XXXXXXX.
trunk sequence: voipceap, sipgate


2 - spa3102
dial rules:
XXXXXXXX
08+08|00XXXXX.
005521|XXXXXXXX
031+0055|XXXXXXXXXX
031+0|XXXXXXXXXX
031+XXXXXXXXXX
outbound:
121|XXXXXXXX
19X
1|XXXXXXXXXX
9|0055ZXXXXXXXXX
9|0800XXXXX.
9|0ZXNXXXXXXX
9|NXXXXXXX
9|ZXX
trunk sequence: spa3102

3 - voipcheap
dial rules:
00+XXXXXXX.
outbound:
800|XXXXXXX.
900|XXXXXXX.
trunk sequence: voipcheap, sipgate

Ok, how do i insert it in the table, defining the rules, outbound and trunk order, please?
I want to use the prefix 9 for all. For now, as i'm not finding out how, i defined 3 prefixes (uggly ).....

Best regards,

Paulo

For the PSTN line setup a phone line in lmce as spa with the phone number (1234567890) as the user and in spa3102 config under pstn line

Line Enable:   yes      
 
SIP Settings
SIP Port:   5061      
 
Proxy and Registration
Proxy:   192.168.80.1
Register:   yes   Make Call Without Reg:   yes
Register Expires:   300   Ans Call Without Reg:   yes
  
Subscriber Information
Display Name:   Unknown  Caller   User ID:   phone number
Password:   password   Use Auth ID:   yes
Auth ID:   phone number   


With those settings the spa3102 pstn line registers for me.

Thank you. It worked. spa3102 registered. This is a difference between the old asterisk (used in release 8.10) and the new one (used in release 10.04).

Again, big THX!!!!!!!!!!!!

Wiki updated with the info.

I'm not blaming not having freepbx, but the lack of documentation and information, and that's not your fault. I know it's a volunteer effort and i also understand it's not easy to make it easy for end users.
I would gladly help, if i had the time and the knowledge for it. Some very little contribution i gave (fail2ban, and some voip trunk settings).  

Well, if i can suggest, how about a little tutorial about, comparing the freepbx settings (trunk - peer detail, user detail), outbound route, dialplan according to trunk and register, or a hidden panel where we could tune it.

For me, at least following directives are missing.
* 83    0    18    0    sip.conf    general    alwaysauthreject    yes
* 85    0    18    0    sip.conf    general    nat                            yes
* 86    0    60    0    sip.conf    general    externhost            mydyndns.homeunix.org
* 87    0    5    0    sip.conf    general    externrefresh            5
* 88    0    60    0    sip.conf    general    localnet                   192.168.80.0/255.255.255.0
* 89    0    9    0    sip.conf    general    allow                    g729
* 90    0    10    0    sip.conf    general    allow                    g723
* 91    0    101    0    sip.conf    general    register                    pwollny:XXXXXXXXX@sip.voipcheap.com/2062036594


I don't know if i did it right, because i could not find what are cat_metric and var_metric for.
I'm not asking to have someone doing it for me, but i wish to know where to put what i need, and i'm not finding out

Best regards to all,

Paulo

For the PSTN line setup a phone line in lmce as spa with the phone number (1234567890) as the user and in spa3102 config under pstn line

Line Enable:   yes      
 
SIP Settings
SIP Port:   5061      
 
Proxy and Registration
Proxy:   192.168.80.1
Register:   yes   Make Call Without Reg:   yes
Register Expires:   300   Ans Call Without Reg:   yes
 
Subscriber Information
Display Name:   Unknown  Caller   User ID:   phone number
Password:   password   Use Auth ID:   yes
Auth ID:   phone number   


With those settings the spa3102 pstn line registers for me.

Gbutters,

thank you for your email. I will try this configuration later and will report back with results..

Best regards,

Paulo

Pw44,

Your problem is within the authentication of your PSTN line., but you know that already.

Three things I would look at: (make note of any changes you make)

1: Did you set the spa for dhcp or assign it a dynamic ip? If you have it as dhcp, change it to static and try that then change the #2 next, if it still don't work put it back.

spa is setted to a fixed ip address 192.168.80.30

2: in the spa under "Proxy and Registration" is it set for yes, if so try to set it to off and then check if it works. If it is set to on it tries to tell LMCE what your spa address is, and since you already assigned an ip you don't need to register.

Proxy and Registration: Register is set to YES. I will give a try with NO.

3: as much as I hate to say this, go back to LMCE 8.10 because the current 10.04 will not work without the ability to setup this box with something like (here it comes, wait, wait,) FREEPBX...... and don't feel too bad as you won't be the only one having to do it.

I myself was going to try and set LMCE up to be an extension of my asterisk box, but after studying how LMCE communicates with asterisk I can't do that either nor can I make the asterisk box work as a in/out trunk to the LMCE because of the changes they made in how the LMCE deals with asterisk. I'm still not sure if I fully understand why the powers to be had to change up asterisk/lmce. If it truly was an issue (as I have read on some other post) that too many people were breaking LMCE using FreePbx I would have thought it would have been easier to just make FreePBX an add-on and not offered help when they screwed it up, just my opinion.

microbrain

Yes, would be prefereable do have something like freepbx, but it's gone, so learning all again from ground zero (where and what, beside the criptic syntax, which was hidden by freepbx). But, as said, it's gone.

Thx for your trying to help Let's see if we get it, mostly by trial and error, due lack of documentation.

de-merten-infoline@de.schneider-electric.com
   
23. Okt (vor 1 Tag)
      
Sehr geehrter Herr Wollny,

leider ist dieses Produkt mit dem Namen "UNIQ" immer noch in einer frühen Entwicklungsphase und eine Markteinführung noch nicht ansatzweise in Sicht.
Die verfügbaren Prospekte wurden leider zu früh in Umlauf gebracht.

Sorry!


Mit freundlichen Grüßen / best regards

i. A.

Uwe Sczepanski

___________________________________________________________________________________________

Uwe Sczepanski  |  Schneider Electric  |  Global Operations  |   Germany  |   Teamleader InfoLine
Phone: +49 2261 702 ext. 235  |  Fax: +49 2261 702 680
Email: infoline.merten@schneider-electric.com  |  Site: www.schneider-electric.com  |  Address: Schneider Electric GmbH, c/o Merten, Fritz-Kotz-Straße 8, 51674 Wiehl
Sitz der Gesellschaft: Ratingen  |  Amtsgericht Düsseldorf  |  HRB 47852  |  USt-IdNr. DE225673854
Geschäftsführer: Dipl.-Ing. Rada Rodriguez (Vorsitzende der Geschäftsführung), Dipl.-Ing. Clemens Blum  |  Vorsitzender des Aufsichtsrats: Marc Coroler
*** Please consider the environment before printing this e-mail






Paulo Wollny

22.10.2012 18:11
   
An
   DE-Merten-Infoline@Europe
Kopie
   
Thema
   Re: Merten - Neue Kontaktanfrage


Sehr geehrter Herr Sczepanski,
ich meine die Lösung die ich unter http://forum.linuxmce.org/index.php/topic,12933.0.html und was ich auch in Ihre Webseite gesehen habe.
Darüber möchte ich weitere Informationen.
Mit freundliche Grüsse,
Paulo Wollny

Am 22. Oktober 2012 10:16 schrieb <de-merten-infoline@de.schneider-electric.com>:
Sehr geehrer Herr Wollny,

das genannte MCE - Plutohome - System ist mir bekannt, aber leider mit keinem unserer Produkte vergleichbar.

Sorry!


Mit freundlichen Grüßen / best regards

i. A.

Uwe Sczepanski

sip debug for asp3102 ata ip:
Please note that the spa3102 is serving fro two purposes:
1 - pstn line as a pots trunk - no way no register.
2 - line 1 as a sip extension 204 - this one registers and works.

Code:

dcerouter*CLI> sip set debug ip 192.168.80.30
SIP Debugging Enabled for IP: 192.168.80.30
[2012-10-24 15:08:34] NOTICE[2494]: chan_sip.c:13059 sip_reregister:    -- Re-registration for  pwollny@sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
[2012-10-24 15:08:34] NOTICE[2494]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for sip.voipcheap.com is 120 sec (Scheduling reregistration in 105 s)


<--- SIP read from UDP:192.168.80.30:5060 --->
REGISTER sip:192.168.80.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.30:5060;branch=z9hG4bK-30f57ef2
From: Line 1 <sip:204@192.168.80.1>;tag=a085e89ca30903a8o0
To: Line 1 <sip:204@192.168.80.1>
Call-ID: e37c19f1-dc99f9a0@192.168.80.30
CSeq: 34238 REGISTER
Max-Forwards: 70
Contact: Line 1 <sip:204@192.168.80.30:5060>;expires=600
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.80.30:5060 (NAT)

<--- Transmitting (NAT) to 192.168.80.30:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.30:5060;branch=z9hG4bK-30f57ef2;received=192.168.80.30;rport=5060
From: Line 1 <sip:204@192.168.80.1>;tag=a085e89ca30903a8o0
To: Line 1 <sip:204@192.168.80.1>;tag=as2d7251a9
Call-ID: e37c19f1-dc99f9a0@192.168.80.30
CSeq: 34238 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48da0ee2"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'e37c19f1-dc99f9a0@192.168.80.30' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.30:5061 --->
REGISTER sip:192.168.80.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.30:5061;branch=z9hG4bK-fc1374d1
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=77c8a0fc9a3ea6c8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>
Call-ID: 7fe3a044-958a6f69@192.168.80.30
CSeq: 57993 REGISTER
Max-Forwards: 70
Contact: Unknown Caller <sip:spa3102@192.168.80.30:5061>;expires=300
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.80.30:5061 (NAT)

<--- Transmitting (NAT) to 192.168.80.30:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.30:5061;branch=z9hG4bK-fc1374d1;received=192.168.80.30;rport=5061
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=77c8a0fc9a3ea6c8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>;tag=as20bcbad4
Call-ID: 7fe3a044-958a6f69@192.168.80.30
CSeq: 57993 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0534bc94"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7fe3a044-958a6f69@192.168.80.30' in 32000 ms (Method: REGISTER)
[2012-10-24 15:08:43] NOTICE[2494]: chan_sip.c:24933 handle_request_register: Registration from 'Unknown Caller <sip:spa3102@192.168.80.1>' failed for '192.168.80.30:5061' - No matching peer found
Scheduling destruction of SIP dialog '7fe3a044-958a6f69@192.168.80.30' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.30:5060 --->
REGISTER sip:192.168.80.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.30:5060;branch=z9hG4bK-268220b5
From: Line 1 <sip:204@192.168.80.1>;tag=a085e89ca30903a8o0
To: Line 1 <sip:204@192.168.80.1>
Call-ID: e37c19f1-dc99f9a0@192.168.80.30
CSeq: 34239 REGISTER
Max-Forwards: 70
Authorization: Digest username="204",realm="asterisk",nonce="48da0ee2",uri="sip:192.168.80.1",algorithm=MD5,response="b46a6784f5334d0ea28b614c869a1d13"
Contact: Line 1 <sip:204@192.168.80.30:5060>;expires=600
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.80.30:5060 (NAT)
    -- Registered SIP '204' at 192.168.80.30:5060

<--- Transmitting (NAT) to 192.168.80.30:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.80.30:5060;branch=z9hG4bK-268220b5;received=192.168.80.30;rport=5060
From: Line 1 <sip:204@192.168.80.1>;tag=a085e89ca30903a8o0
To: Line 1 <sip:204@192.168.80.1>;tag=as2d7251a9
Call-ID: e37c19f1-dc99f9a0@192.168.80.30
CSeq: 34239 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 600
Contact: <sip:204@192.168.80.30:5060>;expires=600
Date: Wed, 24 Oct 2012 17:08:43 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'e37c19f1-dc99f9a0@192.168.80.30' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.30:5061 --->
REGISTER sip:192.168.80.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.80.30:5061;branch=z9hG4bK-5349da3a
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=77c8a0fc9a3ea6c8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>
Call-ID: 7fe3a044-958a6f69@192.168.80.30
CSeq: 57994 REGISTER
Max-Forwards: 70
Authorization: Digest username="spa3102",realm="asterisk",nonce="0534bc94",uri="sip:192.168.80.1",algorithm=MD5,response="6daeb6df71cbff698c5bce7e1f3c98a3"
Contact: Unknown Caller <sip:spa3102@192.168.80.30:5061>;expires=300
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.80.30:5061 (NAT)

<--- Transmitting (NAT) to 192.168.80.30:5061 --->
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 192.168.80.30:5061;branch=z9hG4bK-5349da3a;received=192.168.80.30;rport=5061
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=77c8a0fc9a3ea6c8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>;tag=as20bcbad4
Call-ID: 7fe3a044-958a6f69@192.168.80.30
CSeq: 57994 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2012-10-24 15:08:43] NOTICE[2494]: chan_sip.c:24933 handle_request_register: Registration from 'Unknown Caller <sip:spa3102@192.168.80.1>' failed for '192.168.80.30:5061' - No matching peer found
Scheduling destruction of SIP dialog '7fe3a044-958a6f69@192.168.80.30' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog 'e37c19f1-dc99f9a0@192.168.80.30' Method: REGISTER
Really destroying SIP dialog '7fe3a044-958a6f69@192.168.80.30' Method: REGISTER
dcerouter*CLI> sip set debug off
SIP Debugging Disabled
dcerouter*CLI> quit
Executing last minute cleanups
dcerouter_1031272:

Also while your at it, copy and paste your "sip.conf" file here, you can find it on the LMCE server /etc/asterisk/sip.conf, oh wait a minute, there is no sip.conf on the LMCE machine I have under etc/asterisk..... Either it is located somewhere else and I'm not aware of or found yet, or, it's been renamed to something else, or, maybe someone forgot to include it in the build....

sip.conf is stored in the mysql database.

Thx Microbrain for your offer.

BR,

Paulo

The most frustrating is that is almost impossible to get help and support....  no one knows and the few that knows are silent... in the asterisk forum, etc... If there was any decent documentation... but this is also a no show.
my spa3102 is there, as nmap shows:

Code:

dcerouter_1031272:/home/paulo#  nmap -p 5061 -sU 192.168.80.30

Starting Nmap 5.00 ( http://nmap.org ) at 2012-10-23 18:19 BRST
Interesting ports on 192.168.80.30:
PORT     STATE         SERVICE
5061/udp open|filtered sip-tls
MAC Address: 00:0E:08:C0:97:55 (Cisco Linksys)

Nmap done: 1 IP address (1 host up) scanned in 7.05 seconds
dcerouter_1031272:/home/paulo#  nmap -p 5060 -sU 192.168.80.30

Starting Nmap 5.00 ( http://nmap.org ) at 2012-10-23 18:20 BRST
Interesting ports on 192.168.80.30:
PORT     STATE         SERVICE
5060/udp open|filtered sip
MAC Address: 00:0E:08:C0:97:55 (Cisco Linksys)

Nmap done: 1 IP address (1 host up) scanned in 7.01 seconds
dcerouter_1031272:/home/paulo# nmap -p 5061 -sU 192.168.80.1

Starting Nmap 5.00 ( http://nmap.org ) at 2012-10-23 18:20 BRST
Interesting ports on dcerouter.localdomain (192.168.80.1):
PORT     STATE         SERVICE
5061/udp open|filtered sip-tls

Nmap done: 1 IP address (1 host up) scanned in 2.13 seconds

But asterisk claims no matching peer, but knows the peer is located at 192.168.80.30

Code:

[2012-10-23 18:12:14] NOTICE[2494]: chan_sip.c:24933 handle_request_register: Registration from 'Unknown Caller <sip:spa3102@192.168.80.1>' failed for '192.168.80.30:5061' - No matching peer found
    -- Registered SIP '204' at 192.168.80.30:5060
       > Saved useragent "Linksys/SPA3102-5.1.7(GW)" for peer 204
[2012-10-23 18:12:14] NOTICE[2494]: chan_sip.c:24933 handle_request_register: Registration from 'Unknown Caller <sip:spa3102@192.168.80.1>' failed for '192.168.80.30:5061' - No matching peer found

spa3102 = username = trunkname.
Password checked and rechecked.
What could be wrong? Me? I'm considering myself too stupid to understand what's going on 

I did contact Merten and they told me not yet on the market and that the announcement was made to early...

Microbrain,
you are right. On the previous version, 8.10, with freepbx, i got all running as you said.
With the new one, i'm not finding where and how to make it..... and maybe that's the problem: my ignorance and lack of documentation
Anyway, thx again for trying to help.
Best regards,
Paulo