Author Topic: FreePbx  (Read 6841 times)

posde

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Re: FreePbx
« Reply #30 on: October 21, 2012, 10:38:23 am »
mcefan, if you find it useful to add a link to this thread, why didn't you

pw44

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Re: FreePbx
« Reply #31 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
<------------->

microbrain

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Re: FreePbx
« Reply #32 on: October 21, 2012, 10:41:11 pm »
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?

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....

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

pw44

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Re: FreePbx
« Reply #33 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

pw44

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Re: FreePbx
« Reply #34 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)......

microbrain

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Re: FreePbx
« Reply #35 on: October 22, 2012, 01:59:02 am »
Pw44,

"BTW, where should i define the dialplans according to the trunk?", that and other isses from what I am seeing with the way LMCE setup files with Asterisk this is the problem.

Where we had (under FreePBX) the ability to set the following:

General Settings, Dialed Rules, Outgoing Peer Details (including the ability to list what codecs to use), User Context & Details plus the Registration String....

we don't under the way LMCE has it setup now. Not all VoIp providers have a "Standard" of connection. It would be nice if they did as it would make life easier. That's where something like FreePBX comes in so that you can modify necessary information. Just like under "Phones" we don't have the ability to choose the codecs, the Outbound CID Number Alias, and a host of other options.

If LMCE could find it in their design to at least code the phone line page & phones page to resemble the same as is under FreePBX it would be great.

I, under the circumstances, don't feel I can help you any further as I see the problems being all within LMCE. It will take a rewrite of some code to fix.

Sorry I couldn't be of better help

microbrain

pw44

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Re: FreePbx
« Reply #36 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

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Re: FreePbx
« Reply #37 on: October 22, 2012, 03:55:19 am »
mcefan, if you find it useful to add a link to this thread, why didn't you

Because I had no clue where that was! Unfortunately, people assume others know. I for one, don't usually work with these things, so, this is a learning experience for me, and in the process,  I made up my mind to try to lower the entry curve for others. Not everyone interested in the project is a developer, and I really believe that a larger user base will help the project a great deal, and that's my goal. The entry level learning curve is ridiculously steep, so, every time you see me asking or commenting, I'm not necessarily thinking about myself.

When I do know, I do post the links.

Hope this helps clear things a bit.

posde

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Re: FreePbx
« Reply #38 on: October 22, 2012, 12:17:51 pm »
In trac just add a comment

mcefan

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Re: FreePbx
« Reply #39 on: October 22, 2012, 07:06:51 pm »
In trac just add a comment
I don't know what trac is...
but since the subject came up, I looked on the main site and found the link "Tracker" under "Developer", which takes me here.
Seeing that I am not a developer, I would not have ventured under that heading.
I suggest we add a simple paragraph to the wiki that will point people to it, with a simple explanation of when to do so, and links to the most important pages there.

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Re: FreePbx
« Reply #40 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  :'(

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Re: FreePbx
« Reply #41 on: October 23, 2012, 11:11:34 pm »
see if you can find anything useful using sip debug on:

sip set debug {on|off|ip|peer} Enable/Disable SIP debugging

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Re: FreePbx
« Reply #42 on: October 24, 2012, 12:33:38 am »
Pw44,

As I said earlier I think it's going to be a bug within LMCE and it's db. Please run the following command from a command line on the LMCE machine:

sip set debug ip 192.168.0.3  (put in your IP address of you LMCE machine that the SPA is trying to communicate on)

The post here so I can see what it is doing. Since there is no peer found running sip show peer won't show what's going on between the SPA and LMCE, using the "IP" will.

Asterisk is not finding the peer name you are trying to register with the SPA. There is either a user name or password not matching up with what Asterisk is looking for based on what Asterisk is pulling from it's db, normally Asterisk pulls this info from it's AstDB.

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....

I will be tied up till after ten tonight so I won't be able to get right back to you, but I will look at it tonight. Stop pulling your hair out, I'm sure there is a simple answer to it all.

microbrain



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Re: FreePbx
« Reply #43 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
« Last Edit: October 24, 2012, 07:21:29 pm by pw44 »

microbrain

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Re: FreePbx
« Reply #44 on: October 25, 2012, 02:45:51 am »
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.

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.

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
« Last Edit: October 25, 2012, 03:59:53 am by microbrain »