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Messages - pw44

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46
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...

47
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

48
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)......

49
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

50
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
<------------->

51
Users / Re: FreePbx
« on: October 20, 2012, 05:16:22 pm »

52
Users / Re: FreePbx
« on: October 19, 2012, 02:36:24 am »
Ticket created. If something is needed, please inform and i will provide.
TIA.

53
Users / Re: FreePbx
« on: October 18, 2012, 09:41:50 pm »
Paulo,

to make things easier for everybody, please put the needed information into trac tickets. Forum posts get lost/ignored/whatever. A trac ticket that keeps all the information (and not links to other places) can easily be worked with.

Thanks.

All in one ticket or one ticket for each trunk?

54
Users / Re: FreePbx
« on: October 18, 2012, 09:39:44 pm »
I use the SPA3102 in 1004 as a phone line (PSTN -> IP adapter). I configured it using the phone lines page in the web admin. You have to select the SPA protocol. What took the most work was to set up the SPA itself.

And for the provider setups, if the setups had been integrated into LinuxMCE, they would probably have been considered when changing the Asterisk setup for 1004. Because they were only in a wiki page, they most probably were not.

br,
sambuca
The one in the mentioned wiki is the right one to make spa3102 acts like a trunk for pstn lines. It's all i need. And worked on 8.10.
On 10.04, defining as SPA, actually it will use SIP in the config databank and should use udp port 5061 instead of 5060,

55
Users / Re: FreePbx
« on: October 18, 2012, 03:07:52 am »
The spa3102 in located in the 192.168.80.0 network, no firewall, no nat, nothing. I do have firewall in the 192.168.0.0, which is the external network....
Spa3102 simple does not register. I will try to put it again to work and will debug all, from spa3102 and asterisk, and post the results.
The problem with one way voice is with voipcheap trunk - for this one i will disable firewall and see what happens.
Sipgate i did not give i try, because i don't want to add noise to what is not working.
Regarding the files, they are from the 8.10 release, and i sent to see how to make all this work with asterisk realtime in 10.04.

56
Installation issues / Re: LMCE as complete security system.
« on: October 18, 2012, 01:48:38 am »
Thx for the idea. Could you point me the contact to this KNX manufacturer? MfG aus Rio de Janeiro.

57
Users / Re: FreePbx
« on: October 18, 2012, 01:35:28 am »
For pw44, he has what sounds like three issues. One, the trunk detail to his SIP provider and two possibly a NAT issue (one way audio is normally caused by NAT issues), and three - as  far as the SPA-3102, if the parameters within it are set properly it should register with the LMCE if not then he needs to check its parameters. For the SPA-3102 he can determine what's going on by running sip debug command on a command line entry on the main server and watch what is going on when it tries to register.

Sorry to disagree, all the spa3102 configs are the same that were working on 8.10. spa3102 was not touched. And sip debug shows that it does not register in asterisk 1.8.11.
Thx!

58
Users / Re: FreePbx
« on: October 18, 2012, 01:31:50 am »
Posde,
thx for answering:

I would not say it's a feature request, because is expected that any user would be able to have at least one trunk working with the sip provider of choice :). Ok, not all sip providers are supported by the devel group, so, some documentation should be provided, and making the trunk work would add new providers. I did it with sipgate and voipcheap, but i had something to research and digg. spa3102 was done by Seth.
 My 8.10 working config.
Trunks: SPA-3102 - the spa config is the same as found in: http://wiki.linuxmce.org/index.php/Linksys_SPA3102
            Voipcheap wiki: http://wiki.linuxmce.org/index.php/VoIP_with_voipscheap.com
            sipgate.de wiki: i remember that i created it, but it's not there :(

Well, to the asterisk confi files. If the freepbx version is needed, please let me know.

The sip.conf from the working config:
Code: [Select]
[general]
#include sip_general_additional.conf

bindport = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
alwaysauthreject=yes ; required by fail2ban
disallow=all
allow=ulaw
allow=alaw
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; Reported as required for Asterisk 1.4
notifyringing=yes
notifyhold=yes
limitonpeers=yes

; enable and force the sip jitterbuffer. If these settings are desired
; they should be set in the sip_general_custom.conf file as this file
; will get overwritten during reloads and upgrades.
;
; jbenable=yes
; jbforce=yes

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_general_custom.conf
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
#include sip_custom.conf
#include sip_additional.conf


sip_additional.conf
Code: [Select]
[sip.voipcheap.com]
type=friend
qualify=yes
insecure=invite,port
host=sip.voipcheap.com
dtmfmode=auto
disallow=all
context=from-pstn
allow=ulaw
allow=alaw
allow=g729

[sipgate]
username=username
type=peer
secret=xxxxxxxxxxxxxxxxxxx
qualify=yes
port=5060
nat=yes
insecure=invite,port
host=sipgate.de
fromuser=username
fromdomain=sipgate.de
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=yes
authuser=username
allow=alaw
allow=ulaw
allow=g726
allow=gsm
allow=g729
call-limit=50

[sipgate_de]
username=username
type=friend
secret=xxxxxxxxxxxxxxxxxx
qualify=yes
port=5060
insecure=invite,port
host=sipgate.de
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=yes
allow=alaw
allow=ulaw
allow=g726
allow=gsm
allow=g729

[spa3102]
username=spa3102
type=friend
secret=lmce
qualify=yes
port=5061
nat=never
incominglimit=1
host=dynamic
dtmfmode=auto
context=from-trunk
canreinvite=no
allow=ulaw
call-limit=50

[voipcheap]
username=username
type=friend
sendrpid=yes
secret=xxxxxxxxxxxxxxx
qualify=yes
port=5060
nat=yes
insecure=invite,port
host=sip.voipcheap.com
fromuser=username
fromdomain=sip.voipcheap.com
dtmfmode=auto
disallow=all
context=from-pstn
canreinvite=yes
authuser=username
allow=ulaw
allow=ulaw
allow=g729
call-limit=50

sip_registrations.conf
Code: [Select]
register=usname:xxxxxxxxxx@sip.voipcheap.com/2062036594
register=username:xxxxxxxxxx@sipgate.de/054138594676

sip_nat.conf
Code: [Select]
nat=yes
externip=myhostdyndns.homeunix.org
externrefresh=10
localnet=192.168.80.0/255.255.255.0

localprefixes.conf
Code: [Select]
[trunk-4]
rule1=00+XXXXXXX.

[trunk-2]
rule1=XXXXXXXX
rule2=08+08|00XXXXX.
rule3=005521|XXXXXXXX
rule4=031+0055|XXXXXXXXXX
rule5=031+0|XXXXXXXXXX
rule6=031+XXXXXXXXXX
rule7=031+011XXXXXXXXX

[trunk-3]
rule1=00+XXXXXXX.


If there is any additional configuration file you need, please let me know.

TIA,

Paulo


59
Installation issues / Re: LMCE as complete security system.
« on: October 17, 2012, 10:31:07 pm »
60 Hz would be a problem for motors, imho, my washing machine i.e i do set for 900 rpm, when it is able to 1200 rpm. Would 50/60hz pose a problem for knx?

60
Users / Re: FreePbx
« on: October 17, 2012, 10:27:37 pm »
On 8.10 nothing worked for me ootb, all trunks needed to be feeded, because sipgate, voipcheap and spa-3102 did not had the amp_create****, and later i did create amp_create_sipgate and amp_create_voipcheap (and created wiki for it). spa-3102 was manually created, so as the dialplans.

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