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Messages - pw44

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46
Users / Re: FreePbx
« on: October 25, 2012, 09:44:27 pm »
Ja, mein Kommandant! Werde da nachschauen!

47
Users / Re: [SOLVED] Prefix, dialplan in 1004
« on: October 25, 2012, 09:25:52 pm »
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


48
Users / Re: FreePbx (SOLVED).
« on: October 25, 2012, 08:34:14 pm »
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.

49
Users / Re: FreePbx
« on: October 25, 2012, 08:13:05 pm »
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

50
Users / Re: FreePbx
« on: October 25, 2012, 07:57:01 pm »
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

51
Users / Re: FreePbx
« on: October 25, 2012, 07:50:56 pm »
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

Quote

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.

Quote
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.

52
Users / Re: Another commercial LinuxMCE variant
« on: October 24, 2012, 07:27:53 pm »
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

53
Users / Re: FreePbx
« on: October 24, 2012, 07:15:56 pm »
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: [Select]
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:

Quote
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

54
Users / Re: FreePbx
« on: October 23, 2012, 10:52:22 pm »
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: [Select]
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: [Select]
[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  :'(

55
Users / Re: Another commercial LinuxMCE variant
« on: October 23, 2012, 10:04:22 pm »
I did contact Merten and they told me not yet on the market and that the announcement was made to early...

56
Users / Re: FreePbx
« on: October 22, 2012, 03:03:41 am »
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

57
Users / Re: FreePbx
« on: October 21, 2012, 11:53:45 pm »
Well, something new to report:
having 2 sccp extensions and 1 sip extension, the results are:
calling from outside to my voipcheap trunk, sip extension rings but not the sccp (yes, all are configured to ring)
calling from outside to my sipgate trunk, sip extension rings, but not the sccp (yes, all are configured to ring).
Answering the call from the sip extension, vice both ways (perfect).
Calling from any sccp extension to the sip, voice one way only.
Anyone with a mixed environment (sccp and sip extensions)?
And i did find out that the /usr/share/asterisk/sounds/pluto are wrong (no voicemail)......

58
Users / Re: FreePbx
« on: October 21, 2012, 11:25:49 pm »
Microbrain,
thx for the answer, and now to the details:

Pw44,

The first thing you need to do is get all the "401" & "403" response codes fixed.

I'm seeing that your VoIp provider is not rejecting(401 code) then accepting. Educate me again on some things:

How does you machine communicate with the "outside" world? From LMCE to a router, to a DSL/Cable modem and are you using a static IP? If dynamic IP have you registered with someone like dyndns.org or are you just dependant on your Internet Service Provider?

LMCE box - external nic -> router -> adsl modem
IP setting dynamic and i also have dyndns setted on the external router.
LMCE firewall:
tcp    ipv4    443                  core_input       Delete
tcp    ipv4    2000          core_input       Delete
udp    ipv4    2000          core_input       Delete
udp    ipv4    4569          core_input       Delete
udp    ipv4    5060          core_input       Delete
udp    ipv4    10001 to 20000    core_input       Delete

Both external sip providers (sipgate and voipcheap) uses udp 5060.

Quote

On the spa, if I am correct I'm seeing 401 & 403s, you need to check that the user name & password are the same as what LMCE has entered in its database. It could be that it is and the database is not storing the passwords the same. I have had this issue before in the past, a corrupt database.....

I'm also seeing that the spa is trying to communicate on port 5061, is this the only port in the spa that is assigned, if so you need to have it at beginning at 5060 and end with blank or at least 5061.
I'm assuming that it is set for 5061, asterisk normally starts with 5060....


The spa3102 is connected in the internal network, as the log shows (192.168.80.30).
The spa configuration has two parts: pstn and line 1.
Line 1 is defined as extension, and registers as a sip phone.
The pstn is defined with port 5061 and user and password were checked and are the same. The whole configuration of the spa3102 is the same as the wiki, http://wiki.linuxmce.org/index.php/Linksys_SPA3102, including username and password
Subscriber Settings

    Display Name - Unknown Caller (this is what is displayed when a call comes in without caller ID info)
    UserID - spa3102 (this is what you set when creating the Phone Line in the LinuxMCE webmin)
    Password - lmce (this is what you set when creating the Phone Line in the LinuxMCE webmin)

In the 8.10 release, it worked well, with the new release, it does not authenticates, and i'm not finding out the reason, as userid, password, port and settings are the same.

Quote

Had to look your post over pretty quick as I have to leave for church but will check it again when I return but in the passing time check for all your 401 & 403 and try to correct them, as I say it's a authentication issue. I have a website that will give you the sip responses at: http://en.wikipedia.org/wiki/List_of_SIP_response_codes and you can always go to asterisk web site for additional info.

microbrain

The other issues are:
cisco sccp extension 203 calling sip on spa3102 extension 204: 204 rings, i can hear from 203, but 203 does not hears from 204 :(
sip on spa3102 extension 204 calling cisco sccp extension 203: always busy.

Well, any help is welcome in order to solve it all.

BTW, where should i define the dialplans according to the trunk?

Best regards and thx again.

Paulo

59
Users / Re: FreePbx
« on: October 21, 2012, 04:49:32 pm »
Pw44,
Before you do anything, turn the firewall off and try all your registrations and see if they work. The process of elimination so to speak. Then

Do a "sip set debug on" then do a "sip show peers". Post the results here or send to my email I will try to see what's happening with SPA.

Also, I see that you have listed sip.conf and sip_additional_conf. Have you actually modified these files and added what you needed on the new machine with 10.04? On newer versions of * sip.conf and others like it are rewritten when you reload *, older versions didn't do that. Check to make sure that your sip.con doesn't say at the beginning like "to modify another conf file as this file is overwritten when reloading * . I will read through you conf files posted tonight when I get free and see if I can tell what's going on.

I am planning tonight on seeing if it is just possible to set LMCE up as a client to a external * machine and be done with trying to make * in LMCE work.

microbrain

Hi Microbrain,
thx for yor offer in analyze the debug.
As told, i do have: 2 sccp devices (cisco 7970), extensions 201 and 203, two media directors, extensions 200 and 202 and a spa-3102, where line 1 is set as extension 204. spa-3102 is connected in the inside network (192.168.80.0), with not access from the external network. Is only a pstn to voip gateway to enable the use of the old pstn.
As sip trunks, i have voipcheap, but sound only one way, and the pstn line from spa-3102 does not registers on asterisk.
I did enable nat, and the needed parameters at the database tables, disabled the firewall, but no way to have it working.
sip set debug on gave the following results, and i hope that some can see what i'm not being able to.
Best regards and thx again.
Paulo

Code: [Select]
dcerouter*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
[2012-10-21 12:31:35] NOTICE[13062]: chan_sip.c:13059 sip_reregister:    -- Re-registration for  pwollny@sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK14bcecd4;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as4bbc681b
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1580 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2083991781", response="ecf5a4df5c2bfe63effba1a4d47aca3f"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK14bcecd4;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as4bbc681b
To: <sip:pwollny@sip.voipcheap.com>
Contact: sip:77.72.169.134:5060
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1580 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm="sip.voipcheap.com",nonce="2084097281",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4ae0babf;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as7af43c56
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1581 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084097281", response="3a495d3194a17245a7c7a27c1715cf7b"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4ae0babf;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as7af43c56
To: <sip:pwollny@sip.voipcheap.com>
Contact: <sip:2062036594@192.168.80.1:5060>;expires=120
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1581 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' in 32000 ms (Method: REGISTER)
[2012-10-21 12:31:35] NOTICE[13062]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for sip.voipcheap.com is 120 sec (Scheduling reregistration in 105 s)
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->

<--- 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-67afa790
From: Line 1 <sip:204@192.168.80.1>;tag=aa9309622215e998o0
To: Line 1 <sip:204@192.168.80.1>
Call-ID: 73534f17-489cd6b0@192.168.80.30
CSeq: 52204 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-67afa790;received=192.168.80.30;rport=5060
From: Line 1 <sip:204@192.168.80.1>;tag=aa9309622215e998o0
To: Line 1 <sip:204@192.168.80.1>;tag=as45e5cbbc
Call-ID: 73534f17-489cd6b0@192.168.80.30
CSeq: 52204 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="532a4994"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '73534f17-489cd6b0@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-b84734c9
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=daea7b1299b5a3a8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>
Call-ID: 59e95fec-ea4373df@192.168.80.30
CSeq: 47241 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-b84734c9;received=192.168.80.30;rport=5061
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=daea7b1299b5a3a8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>;tag=as00018c3a
Call-ID: 59e95fec-ea4373df@192.168.80.30
CSeq: 47241 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="77ef4f55"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '59e95fec-ea4373df@192.168.80.30' in 32000 ms (Method: REGISTER)
[2012-10-21 12:31:53] NOTICE[13062]: 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 '59e95fec-ea4373df@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-a2c02b85
From: Line 1 <sip:204@192.168.80.1>;tag=aa9309622215e998o0
To: Line 1 <sip:204@192.168.80.1>
Call-ID: 73534f17-489cd6b0@192.168.80.30
CSeq: 52205 REGISTER
Max-Forwards: 70
Authorization: Digest username="204",realm="asterisk",nonce="532a4994",uri="sip:192.168.80.1",algorithm=MD5,response="aa826026d9f08657d89505d667fdd596"
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-a2c02b85;received=192.168.80.30;rport=5060
From: Line 1 <sip:204@192.168.80.1>;tag=aa9309622215e998o0
To: Line 1 <sip:204@192.168.80.1>;tag=as45e5cbbc
Call-ID: 73534f17-489cd6b0@192.168.80.30
CSeq: 52205 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: Sun, 21 Oct 2012 14:31:53 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '73534f17-489cd6b0@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-359ccda2
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=daea7b1299b5a3a8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>
Call-ID: 59e95fec-ea4373df@192.168.80.30
CSeq: 47242 REGISTER
Max-Forwards: 70
Authorization: Digest username="spa3102",realm="asterisk",nonce="77ef4f55",uri="sip:192.168.80.1",algorithm=MD5,response="01a7d37fe11ebbe27edc873f8e69200e"
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-359ccda2;received=192.168.80.30;rport=5061
From: Unknown Caller <sip:spa3102@192.168.80.1>;tag=daea7b1299b5a3a8o1
To: Unknown Caller <sip:spa3102@192.168.80.1>;tag=as00018c3a
Call-ID: 59e95fec-ea4373df@192.168.80.30
CSeq: 47242 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


<------------>
[2012-10-21 12:31:53] NOTICE[13062]: 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 '59e95fec-ea4373df@192.168.80.30' in 32000 ms (Method: REGISTER)
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
Really destroying SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' Method: REGISTER

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
Really destroying SIP dialog '73534f17-489cd6b0@192.168.80.30' Method: REGISTER
Really destroying SIP dialog '59e95fec-ea4373df@192.168.80.30' Method: REGISTER

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
REGISTER sip:dcerouter SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5061;rport;branch=z9hG4bK538045933
From: <sip:200@dcerouter>;tag=1389717865
To: <sip:200@dcerouter>
Call-ID: 1582944178
CSeq: 300 REGISTER
Contact: <sip:200@192.168.80.1:5061;line=8b7c960eff2e4e2>
Authorization: Digest username="200", realm="asterisk", nonce="4dfdcb82", uri="sip:dcerouter", response="45322023e5915e9823507c25531e1e80", algorithm=MD5
Max-Forwards: 70
User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
Expires: 600
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.80.1:5061 (NAT)

<--- Transmitting (NAT) to 192.168.80.1:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.160:5061;branch=z9hG4bK538045933;received=192.168.80.1;rport=5061
From: <sip:200@dcerouter>;tag=1389717865
To: <sip:200@dcerouter>;tag=as3f131c59
Call-ID: 1582944178
CSeq: 300 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="05b0c029"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1582944178' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.1:5061 --->
REGISTER sip:dcerouter SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5061;rport;branch=z9hG4bK372606350
From: <sip:200@dcerouter>;tag=1389717865
To: <sip:200@dcerouter>
Call-ID: 1582944178
CSeq: 301 REGISTER
Contact: <sip:200@192.168.80.1:5061;line=8b7c960eff2e4e2>
Authorization: Digest username="200", realm="asterisk", nonce="05b0c029", uri="sip:dcerouter", response="e707cd6421b91229218a42c6b76b6235", algorithm=MD5
Max-Forwards: 70
User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
Expires: 600
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.80.1:5061 (NAT)

<--- Transmitting (NAT) to 192.168.80.1:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.160:5061;branch=z9hG4bK372606350;received=192.168.80.1;rport=5061
From: <sip:200@dcerouter>;tag=1389717865
To: <sip:200@dcerouter>;tag=as3f131c59
Call-ID: 1582944178
CSeq: 301 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:200@192.168.80.1:5061;line=8b7c960eff2e4e2>;expires=600
Date: Sun, 21 Oct 2012 14:32:56 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1582944178' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
[2012-10-21 12:33:20] NOTICE[13062]: chan_sip.c:13059 sip_reregister:    -- Re-registration for  pwollny@sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6c40f5fe;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as08137794
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1582 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084097281", response="3a495d3194a17245a7c7a27c1715cf7b"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK6c40f5fe;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as08137794
To: <sip:pwollny@sip.voipcheap.com>
Contact: sip:77.72.169.134:5060
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1582 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm="sip.voipcheap.com",nonce="2084202765",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4c780ecd;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as2e9816e3
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1583 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084202765", response="4db53d6f0c9fc539adbc8baef3beb139"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4c780ecd;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as2e9816e3
To: <sip:pwollny@sip.voipcheap.com>
Contact: <sip:2062036594@192.168.80.1:5060>;expires=120
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1583 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' in 32000 ms (Method: REGISTER)
[2012-10-21 12:33:21] NOTICE[13062]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for sip.voipcheap.com is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog '1582944178' Method: REGISTER

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
Really destroying SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' Method: REGISTER
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
[2012-10-21 12:35:06] NOTICE[13062]: chan_sip.c:13059 sip_reregister:    -- Re-registration for  pwollny@sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4d847b9f;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as7cfc2f3b
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1584 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084202765", response="4db53d6f0c9fc539adbc8baef3beb139"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK4d847b9f;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as7cfc2f3b
To: <sip:pwollny@sip.voipcheap.com>
Contact: sip:77.72.169.134:5060
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1584 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm="sip.voipcheap.com",nonce="2084308234",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK137e616f;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as3018dd41
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1585 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084308234", response="0d2c10f7fb91def4d10a03d2a95009b0"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK137e616f;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as3018dd41
To: <sip:pwollny@sip.voipcheap.com>
Contact: <sip:2062036594@192.168.80.1:5060>;expires=120
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1585 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' in 32000 ms (Method: REGISTER)
[2012-10-21 12:35:06] NOTICE[13062]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for sip.voipcheap.com is 120 sec (Scheduling reregistration in 105 s)
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
Really destroying SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' Method: REGISTER

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
dcerouter*CLI>
dcerouter*CLI>
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
[2012-10-21 12:36:51] NOTICE[13062]: chan_sip.c:13059 sip_reregister:    -- Re-registration for  pwollny@sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK5b478817;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as769b2582
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1586 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084308234", response="0d2c10f7fb91def4d10a03d2a95009b0"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK5b478817;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as769b2582
To: <sip:pwollny@sip.voipcheap.com>
Contact: sip:77.72.169.134:5060
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1586 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm="sip.voipcheap.com",nonce="2084413718",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name sip.voipcheap.com
       > doing dnsmgr_lookup for 'sip.voipcheap.com'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 77.72.169.134:5060:
REGISTER sip:sip.voipcheap.com SIP/2.0
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK5c7fc405;rport
Max-Forwards: 70
From: <sip:pwollny@sip.voipcheap.com>;tag=as0ce7d75e
To: <sip:pwollny@sip.voipcheap.com>
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1587 REGISTER
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="pwollny", realm="sip.voipcheap.com", algorithm=MD5, uri="sip:sip.voipcheap.com", nonce="2084413718", response="1366aec33526c429e259e206276a95be"
Expires: 120
Contact: <sip:2062036594@192.168.80.1:5060>
Content-Length: 0


---

<--- SIP read from UDP:77.72.169.134:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.80.1:5060;branch=z9hG4bK5c7fc405;rport
From: <sip:pwollny@sip.voipcheap.com>;tag=as0ce7d75e
To: <sip:pwollny@sip.voipcheap.com>
Contact: <sip:2062036594@192.168.80.1:5060>;expires=120
Call-ID: 2c36108a47f4f63e44344df47dd84129@192.168.80.1
CSeq: 1587 REGISTER
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' in 32000 ms (Method: REGISTER)
[2012-10-21 12:36:52] NOTICE[13062]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for sip.voipcheap.com is 120 sec (Scheduling reregistration in 105 s)
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!
Really destroying SIP dialog '2c36108a47f4f63e44344df47dd84129@192.168.80.1' Method: REGISTER
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Accepted connection from 192.168.80.131
    -- SCCP: Using ip 192.168.80.1
    -- SCCP: Using 245760 memory for this thread
  == SEP00137FFD944B: Crossover device registration!

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->
dcerouter*CLI> sip show peers
Name/username              Host                                    Dyn Forcerport ACL Port     Status     Realtime
200/200                    192.168.80.1                             D   N             5061     Unmonitored Cached RT
204/204                    192.168.80.30                            D   N             5060     Unmonitored Cached RT
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]

<--- SIP read from UDP:192.168.80.1:5061 --->
jaK
<------------->

60
Users / Re: FreePbx
« on: October 20, 2012, 05:16:22 pm »

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