[SOLVED] Prefix, dialplan in 1004

[cfernandes]

well  if incomming is ok

you have problem  to autorize your  call ,   


 == Using SIP RTP CoS mark 5
    -- Called SIP/003212121212/1234567890
    -- SIP/003212121212-00000007 is making progress passing it to SIP/201-00000006
    -- Got SIP response 503 "Service Unavailable" back from 91.208.12.133:5060
    -- SIP/003212121212-00000007 is circuit-busy

and on debug  i can see

<--- SIP read from UDP:91.208.12.133:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK226a2160;received=178.117.103.101;rport=5060
From: "pl_172" <sip:329923232323@ssw3.brussels.weepee.org>;tag=as038cadc7
To: <sip:1234567890@ssw3.brussels.weepee.org:5060>;tag=as0c5744cc
Call-ID: 3be558c25424cc7d0e296fd27683ddfe@ssw3.brussels.weepee.org
CSeq: 102 INVITE
Server: weepee


can you test  from softphone  on a windows pc  direct to a isp  if  you can place call   ?

i think  that is is reject  address 192.168.0.184 

[brononius]

you have problem  to autorize your  call ,   
can you test  from softphone  on a windows pc  direct to a isp  if  you can place call   ?

I'll see if i can install a softphone this evening and test some things.
With version 804 everything was working fine. All same hardware and credentials, only version 1004 now...
So i don't think it's a provider problem. 

i think that is is reject address 192.168.0.184

This is my 'private' external IP of my linuxmce. So not the public ip of my internet router.
I suppose that this should be the public ip that tries to reach my sip providers?
I can image that the SIP provider simply blocks private adresses.

Can I give somewhere an option that the "linuxMCE 1004" hides or NAT the outgoing ip with my public IP?

[cfernandes]

on sip conf  have a confi for external ip address you can add this

run this  on mysql  asterisk database
replace xxx.xxx.xxx.xxx  with your external valid ip address

insert into ast_config (cat_metric,var_metric,commented,filename,category,var_name,var_val) values(0,18,0,'sip.conf','general','externip','xxx.xxx.xxx.xxx')

[brononius]

The rule is in the database. But didn't worked... :$

Code:

id cat_metric var_metric commented filename category var_name var_val
81 0 18 0 sip.conf general externip 178.117.103.101

I've got the impression that he still use the private one (i've restarted asterisk, rebooted the server)...

Log:

Code:

[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c:
<--- SIP read from UDP:192.168.111.76:5060 --->
INVITE sip:0AAAAAAAAAA@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK0e78122d
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:206@192.168.111.76:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "202" <sip:206@192.168.111.1>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 280
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 27662 0 IN IP4 192.168.111.76
s=SIP Call
t=0 0
m=audio 21270 RTP/AVP 0 8 18 101
c=IN IP4 192.168.111.76
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: --- (17 headers 13 lines) ---
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Sending to 192.168.111.76:5060 (NAT)
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Using INVITE request as basis request - 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Found peer '206' for '206' from 192.168.111.76:5060
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.111.76:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK0e78122d;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>;tag=as1eb3880c
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
CSeq: 101 INVITE
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="1da92852"
Content-Length: 0


<------------>
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Scheduling destruction of SIP dialog '001a6c7b-60740012-44a70612-0067f409@192.168.111.76' in 32000 ms (Method: INVITE)
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c:
<--- SIP read from UDP:192.168.111.76:5060 --->
ACK sip:0AAAAAAAAAA@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK0e78122d
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>;tag=as1eb3880c
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
CSeq: 101 ACK
Content-Length: 0

<------------->
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: --- (7 headers 0 lines) ---
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c:
<--- SIP read from UDP:192.168.111.76:5060 --->
INVITE sip:0AAAAAAAAAA@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK566a43ec
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:206@192.168.111.76:5060;transport=udp>
Authorization: Digest username="206",realm="asterisk",uri="sip:0AAAAAAAAAA@192.168.111.1",response="b1740665de83919b44534b837b39e159",nonce="1da92852",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "202" <sip:206@192.168.111.1>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 280
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 27662 0 IN IP4 192.168.111.76
s=SIP Call
t=0 0
m=audio 21270 RTP/AVP 0 8 18 101
c=IN IP4 192.168.111.76
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: --- (18 headers 13 lines) ---
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Sending to 192.168.111.76:5060 (NAT)
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Using INVITE request as basis request - 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Found peer '206' for '206' from 192.168.111.76:5060
[Aug  3 18:34:53] VERBOSE[26462] netsock2.c:   == Using SIP RTP CoS mark 5
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Found RTP audio format 0
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Found RTP audio format 8
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Found RTP audio format 18
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Found RTP audio format 101
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Found audio description format PCMU for ID 0
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Found audio description format PCMA for ID 8
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Found audio description format G729 for ID 18
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Found audio description format telephone-event for ID 101
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Peer audio RTP is at port 192.168.111.76:21270
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: Looking for 0AAAAAAAAAA in from-internal (domain 192.168.111.1)
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c: list_route: hop: <sip:206@192.168.111.76:5060;transport=udp>
[Aug  3 18:34:53] VERBOSE[26462] chan_sip.c:
<--- Transmitting (NAT) to 192.168.111.76:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK566a43ec;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0AAAAAAAAAA@192.168.111.1:5060>
Content-Length: 0


<------------>
[Aug  3 18:34:53] VERBOSE[30038] pbx.c:     -- Executing [0AAAAAAAAAA@from-internal:1] ResetCDR("SIP/206-0000000e", "") in new stack
[Aug  3 18:34:53] VERBOSE[30038] pbx.c:     -- Executing [0AAAAAAAAAA@from-internal:2] NoCDR("SIP/206-0000000e", "") in new stack
[Aug  3 18:34:53] VERBOSE[30038] pbx.c:     -- Executing [0AAAAAAAAAA@from-internal:3] Wait("SIP/206-0000000e", "1") in new stack
[Aug  3 18:34:54] VERBOSE[30038] pbx.c:     -- Executing [0AAAAAAAAAA@from-internal:4] Playback("SIP/206-0000000e", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[Aug  3 18:34:54] VERBOSE[30038] file.c:     -- <SIP/206-0000000e> Playing 'silence/1.gsm' (language 'en')
[Aug  3 18:34:55] VERBOSE[30038] file.c:     -- <SIP/206-0000000e> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
[Aug  3 18:34:56] VERBOSE[26462] chan_sip.c:
<--- SIP read from UDP:91.208.12.133:5060 --->
OPTIONS sip:329AAABBCC@192.168.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 91.208.12.133:5060;branch=z9hG4bK2696ae66;rport
Max-Forwards: 70
From: "weepee" <sip:weepee@91.208.12.133>;tag=as04f89b28
To: <sip:329AAABBCC@192.168.0.184:5060>
Contact: <sip:weepee@91.208.12.133:5060>
Call-ID: 6b54c61552efb6df452517341355151e@91.208.12.133:5060
CSeq: 102 OPTIONS
User-Agent: weepee
Date: Fri, 03 Aug 2012 16:34:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
[Aug  3 18:34:56] VERBOSE[26462] chan_sip.c: --- (13 headers 0 lines) ---
[Aug  3 18:34:56] VERBOSE[26462] chan_sip.c: Looking for 329AAABBCC in from-sip-external (domain 192.168.0.184)
[Aug  3 18:34:56] VERBOSE[26462] chan_sip.c:
<--- Transmitting (NAT) to 91.208.12.133:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.208.12.133:5060;branch=z9hG4bK2696ae66;received=91.208.12.133;rport=5060
From: "weepee" <sip:weepee@91.208.12.133>;tag=as04f89b28
To: <sip:329AAABBCC@192.168.0.184:5060>;tag=as516e1007
Call-ID: 6b54c61552efb6df452517341355151e@91.208.12.133:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.0.184:5060>
Accept: application/sdp
Content-Length: 0


<------------>
[Aug  3 18:34:56] VERBOSE[26462] chan_sip.c: Scheduling destruction of SIP dialog '6b54c61552efb6df452517341355151e@91.208.12.133:5060' in 32000 ms (Method: OPTIONS)
[Aug  3 18:34:57] VERBOSE[26462] chan_sip.c:
<--- SIP read from UDP:192.168.111.76:5060 --->
CANCEL sip:0AAAAAAAAAA@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK566a43ec
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
Max-Forwards: 70
CSeq: 102 CANCEL
User-Agent: Cisco-CP7940G/8.0
Content-Length: 0
Authorization: Digest username="206",realm="asterisk",uri="sip:0AAAAAAAAAA@192.168.111.1",response="3224724765443c2a55e7f72584cb3dbe",nonce="1da92852",algorithm=MD5

<------------->
[Aug  3 18:34:57] VERBOSE[26462] chan_sip.c: --- (10 headers 0 lines) ---
[Aug  3 18:34:57] VERBOSE[26462] chan_sip.c: Sending to 192.168.111.76:5060 (NAT)
[Aug  3 18:34:57] VERBOSE[26462] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.111.76:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK566a43ec;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>;tag=as5ac59eb5
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
CSeq: 102 INVITE
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


<------------>
[Aug  3 18:34:57] VERBOSE[26462] chan_sip.c:
<--- Transmitting (NAT) to 192.168.111.76:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK566a43ec;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>;tag=as5ac59eb5
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
CSeq: 102 CANCEL
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


<------------>
[Aug  3 18:34:57] VERBOSE[30038] pbx.c:   == Spawn extension (from-internal, 0AAAAAAAAAA, 4) exited non-zero on 'SIP/206-0000000e'
[Aug  3 18:34:57] VERBOSE[30038] pbx.c:     -- Executing [h@from-internal:1] Macro("SIP/206-0000000e", "hangupcall") in new stack
[Aug  3 18:34:57] VERBOSE[30038] pbx.c:     -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/206-0000000e", "w") in new stack
[Aug  3 18:34:57] VERBOSE[30038] pbx.c:     -- Executing [s@macro-hangupcall:2] NoCDR("SIP/206-0000000e", "") in new stack
[Aug  3 18:34:57] VERBOSE[30038] pbx.c:     -- Executing [s@macro-hangupcall:3] GotoIf("SIP/206-0000000e", "1?skiprg") in new stack
[Aug  3 18:34:57] VERBOSE[30038] pbx.c:     -- Goto (macro-hangupcall,s,6)
[Aug  3 18:34:57] VERBOSE[30038] pbx.c:     -- Executing [s@macro-hangupcall:6] GotoIf("SIP/206-0000000e", "1?skipblkvm") in new stack
[Aug  3 18:34:57] VERBOSE[30038] pbx.c:     -- Goto (macro-hangupcall,s,9)
[Aug  3 18:34:57] VERBOSE[30038] pbx.c:     -- Executing [s@macro-hangupcall:9] GotoIf("SIP/206-0000000e", "1?theend") in new stack
[Aug  3 18:34:57] VERBOSE[30038] pbx.c:     -- Goto (macro-hangupcall,s,11)
[Aug  3 18:34:57] VERBOSE[30038] pbx.c:     -- Executing [s@macro-hangupcall:11] Hangup("SIP/206-0000000e", "") in new stack
[Aug  3 18:34:57] VERBOSE[30038] app_macro.c:   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/206-0000000e' in macro 'hangupcall'
[Aug  3 18:34:57] VERBOSE[30038] pbx.c:   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/206-0000000e'
[Aug  3 18:34:57] VERBOSE[26462] chan_sip.c:
<--- SIP read from UDP:192.168.111.76:5060 --->
ACK sip:0AAAAAAAAAA@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK656b9666
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
Max-Forwards: 70
CSeq: 102 ACK
User-Agent: Cisco-CP7940G/8.0
Authorization: Digest username="206",realm="asterisk",uri="sip:0AAAAAAAAAA@192.168.111.1",response="3224724765443c2a55e7f72584cb3dbe",nonce="1da92852",algorithm=MD5
Remote-Party-ID: "202" <sip:206@192.168.111.1>;party=calling;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0

<------------->
[Aug  3 18:34:57] VERBOSE[26462] chan_sip.c: --- (11 headers 0 lines) ---
[Aug  3 18:34:57] VERBOSE[26462] chan_sip.c: Retransmitting #1 (NAT) to 192.168.111.76:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK566a43ec;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>;tag=as5ac59eb5
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
CSeq: 102 INVITE
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


---
[Aug  3 18:34:58] VERBOSE[26462] chan_sip.c: Retransmitting #2 (NAT) to 192.168.111.76:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK566a43ec;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>;tag=as5ac59eb5
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
CSeq: 102 INVITE
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


---
[Aug  3 18:34:59] VERBOSE[26462] chan_sip.c:
<--- SIP read from UDP:192.168.111.76:5060 --->
REGISTER sip:192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK31c92497
From: <sip:206@192.168.111.1>;tag=001a6c7b607401d56fa44ec7-40aca2be
To: <sip:206@192.168.111.1>
Call-ID: 001a6c7b-60740002-1d45f84c-7253dc00@192.168.111.76
Max-Forwards: 70
CSeq: 1004 REGISTER
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:206@192.168.111.76:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001a6c7b6074>";+u.sip!model.ccm.cisco.com="8"
Content-Length: 0
Expires: 180

<------------->
[Aug  3 18:34:59] VERBOSE[26462] chan_sip.c: --- (11 headers 0 lines) ---
[Aug  3 18:34:59] VERBOSE[26462] chan_sip.c: Sending to 192.168.111.76:5060 (NAT)
[Aug  3 18:34:59] VERBOSE[26462] chan_sip.c:
<--- Transmitting (NAT) to 192.168.111.76:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK31c92497;received=192.168.111.76;rport=5060
From: <sip:206@192.168.111.1>;tag=001a6c7b607401d56fa44ec7-40aca2be
To: <sip:206@192.168.111.1>;tag=as2f735419
Call-ID: 001a6c7b-60740002-1d45f84c-7253dc00@192.168.111.76
CSeq: 1004 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="3431614d"
Content-Length: 0


<------------>
[Aug  3 18:34:59] VERBOSE[26462] chan_sip.c: Scheduling destruction of SIP dialog '001a6c7b-60740002-1d45f84c-7253dc00@192.168.111.76' in 32000 ms (Method: REGISTER)
[Aug  3 18:34:59] VERBOSE[26462] chan_sip.c:
<--- SIP read from UDP:192.168.111.76:5060 --->
REGISTER sip:192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK249a416a
From: <sip:206@192.168.111.1>;tag=001a6c7b607401d56fa44ec7-40aca2be
To: <sip:206@192.168.111.1>
Call-ID: 001a6c7b-60740002-1d45f84c-7253dc00@192.168.111.76
Max-Forwards: 70
CSeq: 1005 REGISTER
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:206@192.168.111.76:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001a6c7b6074>";+u.sip!model.ccm.cisco.com="8"
Authorization: Digest username="206",realm="asterisk",uri="sip:192.168.111.1",response="e92fd49b1abbae30b99596ac2037cfde",nonce="3431614d",algorithm=MD5
Content-Length: 0
Expires: 180

<------------->
[Aug  3 18:34:59] VERBOSE[26462] chan_sip.c: --- (12 headers 0 lines) ---
[Aug  3 18:34:59] VERBOSE[26462] chan_sip.c: Sending to 192.168.111.76:5060 (NAT)
[Aug  3 18:34:59] VERBOSE[26462] chan_sip.c:
<--- Transmitting (NAT) to 192.168.111.76:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK249a416a;received=192.168.111.76;rport=5060
From: <sip:206@192.168.111.1>;tag=001a6c7b607401d56fa44ec7-40aca2be
To: <sip:206@192.168.111.1>;tag=as2f735419
Call-ID: 001a6c7b-60740002-1d45f84c-7253dc00@192.168.111.76
CSeq: 1005 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: 180
Contact: <sip:206@192.168.111.76:5060;transport=udp>;expires=180
Date: Fri, 03 Aug 2012 16:34:59 GMT
Content-Length: 0


<------------>
[Aug  3 18:34:59] VERBOSE[26462] chan_sip.c: Scheduling destruction of SIP dialog '001a6c7b-60740002-1d45f84c-7253dc00@192.168.111.76' in 32000 ms (Method: REGISTER)
[Aug  3 18:35:00] VERBOSE[26462] chan_sip.c: Retransmitting #3 (NAT) to 192.168.111.76:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK566a43ec;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>;tag=as5ac59eb5
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
CSeq: 102 INVITE
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


---
[Aug  3 18:35:03] VERBOSE[26462] chan_sip.c: Really destroying SIP dialog '30214d55133e611b70144c413c18f5ff@91.208.12.133:5060' Method: OPTIONS
[Aug  3 18:35:04] VERBOSE[26462] chan_sip.c: Retransmitting #4 (NAT) to 192.168.111.76:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK566a43ec;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607401d45d25714c-4e16c462
To: <sip:0AAAAAAAAAA@192.168.111.1>;tag=as5ac59eb5
Call-ID: 001a6c7b-60740012-44a70612-0067f409@192.168.111.76
CSeq: 102 INVITE
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


---

[cfernandes]

can you add on

insert into ast_config (cat_metric,var_metric,commented,filename,category,var_name,var_val) values(0,19,0,'sip.conf','general','nat','yes')
 and do test again
after insert only need to 
asterisk -r 
reload

[cfernandes]

more elegant  config
 you can add  remember to change  xxx.xxxx to your internal network


insert into ast_config (cat_metric,var_metric,commented,filename,category,var_name,var_val) values(0,0,0,'sip_nat.conf','general','nat','yes')
insert into ast_config (cat_metric,var_metric,commented,filename,category,var_name,var_val) values(0,1,0,'sip_nat.conf','general','externhost','host.dyndns.org')
insert into ast_config (cat_metric,var_metric,commented,filename,category,var_name,var_val) values(0,2,0,'sip_nat.conf','general','externrefresh','60')
insert into ast_config (cat_metric,var_metric,commented,filename,category,var_name,var_val) values(0,3,0,'sip_nat.conf','general','localnet','xxx.xxx.xxx.xxx/xxx.xxx.xxx.xxx')

[brononius]

insert into ast_config (cat_metric,var_metric,commented,filename,category,var_name,var_val) values(0,19,0,'sip.conf','general','nat','yes')

Done it, but still can't reach an external phone.

I'm not able to dial a complete number. But i'm now sure that the prefix number is in order. 

• When i call AAAABBCCDD, i've got a voice telling me that the number can't be dialed as formed.
• When i call 7AAAABBCCDD, i've got a message that all lines are busy.

Maybe a weird fact:
My provider has some short number (fe to know the status of your bills). And this can be reached?

• So when i call 71950, i'm hearing the status of my bill?
• When i call 71970, i'm hearing my account number.

Calling a short number (7 1950):

Code:

<--- SIP read from UDP:192.168.111.76:5060 --->
INVITE sip:71950@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK089a5afe
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b6074001c6fb2cadb-3fcf306a
To: <sip:71950@192.168.111.1>
Call-ID: 001a6c7b-60740018-7ed68b3f-5aa08f84@192.168.111.76
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:206@192.168.111.76:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "202" <sip:206@192.168.111.1>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 280
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 17232 0 IN IP4 192.168.111.76
s=SIP Call
t=0 0
m=audio 17716 RTP/AVP 0 8 18 101
c=IN IP4 192.168.111.76
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (17 headers 13 lines) ---
Sending to 192.168.111.76:5060 (NAT)
Using INVITE request as basis request - 001a6c7b-60740018-7ed68b3f-5aa08f84@192.168.111.76
Found peer '206' for '206' from 192.168.111.76:5060

<--- Reliably Transmitting (NAT) to 192.168.111.76:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK089a5afe;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b6074001c6fb2cadb-3fcf306a
To: <sip:71950@192.168.111.1>;tag=as1e8ab51e
Call-ID: 001a6c7b-60740018-7ed68b3f-5aa08f84@192.168.111.76
CSeq: 101 INVITE
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="5988f94a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '001a6c7b-60740018-7ed68b3f-5aa08f84@192.168.111.76' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.111.76:5060 --->
ACK sip:71950@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK089a5afe
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b6074001c6fb2cadb-3fcf306a
To: <sip:71950@192.168.111.1>;tag=as1e8ab51e
Call-ID: 001a6c7b-60740018-7ed68b3f-5aa08f84@192.168.111.76
CSeq: 101 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.111.76:5060 --->
INVITE sip:71950@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK197884c3
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b6074001c6fb2cadb-3fcf306a
To: <sip:71950@192.168.111.1>
Call-ID: 001a6c7b-60740018-7ed68b3f-5aa08f84@192.168.111.76
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:206@192.168.111.76:5060;transport=udp>
Authorization: Digest username="206",realm="asterisk",uri="sip:71950@192.168.111.1",response="6805027617319fa8fe8d4bad736165de",nonce="5988f94a",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "202" <sip:206@192.168.111.1>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 280
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 17232 0 IN IP4 192.168.111.76
s=SIP Call
t=0 0
m=audio 17716 RTP/AVP 0 8 18 101
c=IN IP4 192.168.111.76
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (18 headers 13 lines) ---
Sending to 192.168.111.76:5060 (NAT)
Using INVITE request as basis request - 001a6c7b-60740018-7ed68b3f-5aa08f84@192.168.111.76
Found peer '206' for '206' from 192.168.111.76:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.111.76:17716
Looking for 71950 in from-internal (domain 192.168.111.1)
list_route: hop: <sip:206@192.168.111.76:5060;transport=udp>

<--- Transmitting (NAT) to 192.168.111.76:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK197884c3;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b6074001c6fb2cadb-3fcf306a
To: <sip:71950@192.168.111.1>
Call-ID: 001a6c7b-60740018-7ed68b3f-5aa08f84@192.168.111.76
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:71950@192.168.111.1:5060>
Content-Length: 0


<------------>
Audio is at 11778
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 91.208.12.133:5060:
INVITE sip:1950@ssw3.brussels.weepee.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK367f4ede;rport
Max-Forwards: 70
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as2d3db311
To: <sip:1950@ssw3.brussels.weepee.org:5060>
Contact: <sip:329909011639@192.168.0.184:5060>
Call-ID: 0c9e3ff00d550ad40ac39e921b2a9cca@ssw3.brussels.weepee.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Date: Fri, 03 Aug 2012 17:52:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 656911415 656911415 IN IP4 192.168.0.184
s=Asterisk PBX 1.8.11.1-1digium1~lucid
c=IN IP4 192.168.0.184
t=0 0
m=audio 11778 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:91.208.12.133:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK367f4ede;received=178.117.103.101;rport=1719
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as2d3db311
To: <sip:1950@ssw3.brussels.weepee.org:5060>;tag=as612b27ea
Call-ID: 0c9e3ff00d550ad40ac39e921b2a9cca@ssw3.brussels.weepee.org
CSeq: 102 INVITE
Server: weepee
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="weepee", nonce="1dc6e8bc"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:1950@ssw3.brussels.weepee.org:5060> for address/port to send to
set_destination: set destination to 91.208.12.133:5060
Transmitting (NAT) to 91.208.12.133:5060:
ACK sip:1950@ssw3.brussels.weepee.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK367f4ede;rport
Max-Forwards: 70
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as2d3db311
To: <sip:1950@ssw3.brussels.weepee.org:5060>;tag=as612b27ea
Contact: <sip:329909011639@192.168.0.184:5060>
Call-ID: 0c9e3ff00d550ad40ac39e921b2a9cca@ssw3.brussels.weepee.org
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Content-Length: 0


---
Audio is at 11778
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 91.208.12.133:5060:
INVITE sip:1950@ssw3.brussels.weepee.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK7b3a7fd0;rport
Max-Forwards: 70
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as2d3db311
To: <sip:1950@ssw3.brussels.weepee.org:5060>
Contact: <sip:329909011639@192.168.0.184:5060>
Call-ID: 0c9e3ff00d550ad40ac39e921b2a9cca@ssw3.brussels.weepee.org
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="329909011639", realm="weepee", algorithm=MD5, uri="sip:1950@ssw3.brussels.weepee.org:5060", nonce="1dc6e8bc", response="99f608863404a7db12b7fd4bf6fad572"
Date: Fri, 03 Aug 2012 17:52:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 656911415 656911416 IN IP4 192.168.0.184
s=Asterisk PBX 1.8.11.1-1digium1~lucid
c=IN IP4 192.168.0.184
t=0 0
m=audio 11778 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:91.208.12.133:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK7b3a7fd0;received=178.117.103.101;rport=1719
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as2d3db311
To: <sip:1950@ssw3.brussels.weepee.org:5060>
Call-ID: 0c9e3ff00d550ad40ac39e921b2a9cca@ssw3.brussels.weepee.org
CSeq: 103 INVITE
Server: weepee
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1950@91.208.12.133:5060>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:91.208.12.133:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK7b3a7fd0;received=178.117.103.101;rport=1719
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as2d3db311
To: <sip:1950@ssw3.brussels.weepee.org:5060>;tag=as01168731
Call-ID: 0c9e3ff00d550ad40ac39e921b2a9cca@ssw3.brussels.weepee.org
CSeq: 103 INVITE
Server: weepee
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1950@91.208.12.133:5060>
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 49593661 49593661 IN IP4 91.208.12.133
s=weepee
c=IN IP4 91.208.12.133
t=0 0
m=audio 30848 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 91.208.12.133:30848
list_route: hop: <sip:1950@91.208.12.133:5060>
set_destination: Parsing <sip:1950@91.208.12.133:5060> for address/port to send to
set_destination: set destination to 91.208.12.133:5060
Transmitting (NAT) to 91.208.12.133:5060:
ACK sip:1950@91.208.12.133:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK191d123a;rport
Max-Forwards: 70
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as2d3db311
To: <sip:1950@ssw3.brussels.weepee.org:5060>;tag=as01168731
Contact: <sip:329909011639@192.168.0.184:5060>
Call-ID: 0c9e3ff00d550ad40ac39e921b2a9cca@ssw3.brussels.weepee.org
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Content-Length: 0


---
Audio is at 19124
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.111.76:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK197884c3;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b6074001c6fb2cadb-3fcf306a
To: <sip:71950@192.168.111.1>;tag=as0d009068
Call-ID: 001a6c7b-60740018-7ed68b3f-5aa08f84@192.168.111.76
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:71950@192.168.111.1:5060>
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 1828900027 1828900027 IN IP4 192.168.111.1
s=Asterisk PBX 1.8.11.1-1digium1~lucid
c=IN IP4 192.168.111.1
t=0 0
m=audio 19124 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:192.168.111.76:5060 --->
ACK sip:71950@192.168.111.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK28807c3f
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b6074001c6fb2cadb-3fcf306a
To: <sip:71950@192.168.111.1>;tag=as0d009068
Call-ID: 001a6c7b-60740018-7ed68b3f-5aa08f84@192.168.111.76
Max-Forwards: 70
CSeq: 102 ACK
User-Agent: Cisco-CP7940G/8.0
Authorization: Digest username="206",realm="asterisk",uri="sip:71950@192.168.111.1",response="6805027617319fa8fe8d4bad736165de",nonce="5988f94a",algorithm=MD5
Remote-Party-ID: "202" <sip:206@192.168.111.1>;party=calling;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '62dd493b76f116cf035775e87d17249a@ssw3.brussels.weepee.org' Method: BYE
dcerouter*CLI>
Disconnected from Asterisk server

Calling a 'normal' number (7 0479123456):

Code:

<--- SIP read from UDP:91.208.12.133:5060 --->
OPTIONS sip:3293959892@192.168.0.184:5060 SIP/2.0
Via: SIP/2.0/UDP 91.208.12.133:5060;branch=z9hG4bK6fba0df8;rport
Max-Forwards: 70
From: "weepee" <sip:weepee@91.208.12.133>;tag=as12c0518a
To: <sip:3293959892@192.168.0.184:5060>
Contact: <sip:weepee@91.208.12.133:5060>
Call-ID: 1390c06f3f039ce5556f9a075577484b@91.208.12.133:5060
CSeq: 102 OPTIONS
User-Agent: weepee
Date: Fri, 03 Aug 2012 17:56:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Looking for 3293959892 in from-sip-external (domain 192.168.0.184)

<--- Transmitting (NAT) to 91.208.12.133:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.208.12.133:5060;branch=z9hG4bK6fba0df8;received=91.208.12.133;rport=5060
From: "weepee" <sip:weepee@91.208.12.133>;tag=as12c0518a
To: <sip:3293959892@192.168.0.184:5060>;tag=as6563f7d3
Call-ID: 1390c06f3f039ce5556f9a075577484b@91.208.12.133:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.0.184:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1390c06f3f039ce5556f9a075577484b@91.208.12.133:5060' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:192.168.111.76:5060 --->
INVITE sip:70479123456@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK537dd197
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607400212d8f9701-2aedb5a7
To: <sip:70479123456@192.168.111.1>
Call-ID: 001a6c7b-6074001b-575a3247-02dce10b@192.168.111.76
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:206@192.168.111.76:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "202" <sip:206@192.168.111.1>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 280
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 27749 0 IN IP4 192.168.111.76
s=SIP Call
t=0 0
m=audio 20244 RTP/AVP 0 8 18 101
c=IN IP4 192.168.111.76
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (17 headers 13 lines) ---
Sending to 192.168.111.76:5060 (NAT)
Using INVITE request as basis request - 001a6c7b-6074001b-575a3247-02dce10b@192.168.111.76
Found peer '206' for '206' from 192.168.111.76:5060

<--- Reliably Transmitting (NAT) to 192.168.111.76:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK537dd197;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607400212d8f9701-2aedb5a7
To: <sip:70479123456@192.168.111.1>;tag=as26d68a74
Call-ID: 001a6c7b-6074001b-575a3247-02dce10b@192.168.111.76
CSeq: 101 INVITE
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="24f8cde6"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '001a6c7b-6074001b-575a3247-02dce10b@192.168.111.76' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.111.76:5060 --->
ACK sip:70479123456@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK537dd197
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607400212d8f9701-2aedb5a7
To: <sip:70479123456@192.168.111.1>;tag=as26d68a74
Call-ID: 001a6c7b-6074001b-575a3247-02dce10b@192.168.111.76
CSeq: 101 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.111.76:5060 --->
INVITE sip:70479123456@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK2bf267c4
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607400212d8f9701-2aedb5a7
To: <sip:70479123456@192.168.111.1>
Call-ID: 001a6c7b-6074001b-575a3247-02dce10b@192.168.111.76
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:206@192.168.111.76:5060;transport=udp>
Authorization: Digest username="206",realm="asterisk",uri="sip:70479123456@192.168.111.1",response="ce73e89707077c8ef940d229e9989e11",nonce="24f8cde6",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "202" <sip:206@192.168.111.1>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 280
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 27749 0 IN IP4 192.168.111.76
s=SIP Call
t=0 0
m=audio 20244 RTP/AVP 0 8 18 101
c=IN IP4 192.168.111.76
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (18 headers 13 lines) ---
Sending to 192.168.111.76:5060 (NAT)
Using INVITE request as basis request - 001a6c7b-6074001b-575a3247-02dce10b@192.168.111.76
Found peer '206' for '206' from 192.168.111.76:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.111.76:20244
Looking for 70479123456 in from-internal (domain 192.168.111.1)
list_route: hop: <sip:206@192.168.111.76:5060;transport=udp>

<--- Transmitting (NAT) to 192.168.111.76:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK2bf267c4;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607400212d8f9701-2aedb5a7
To: <sip:70479123456@192.168.111.1>
Call-ID: 001a6c7b-6074001b-575a3247-02dce10b@192.168.111.76
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:70479123456@192.168.111.1:5060>
Content-Length: 0


<------------>
Audio is at 17872
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 91.208.12.133:5060:
INVITE sip:0479123456@ssw3.brussels.weepee.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK2ba350b0;rport
Max-Forwards: 70
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as0ae2d0e2
To: <sip:0479123456@ssw3.brussels.weepee.org:5060>
Contact: <sip:329909011639@192.168.0.184:5060>
Call-ID: 3ee5f692663c3da21c809d3725af567c@ssw3.brussels.weepee.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Date: Fri, 03 Aug 2012 17:56:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 2098828985 2098828985 IN IP4 192.168.0.184
s=Asterisk PBX 1.8.11.1-1digium1~lucid
c=IN IP4 192.168.0.184
t=0 0
m=audio 17872 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:91.208.12.133:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK2ba350b0;received=178.117.103.101;rport=1719
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as0ae2d0e2
To: <sip:0479123456@ssw3.brussels.weepee.org:5060>;tag=as46ec6919
Call-ID: 3ee5f692663c3da21c809d3725af567c@ssw3.brussels.weepee.org
CSeq: 102 INVITE
Server: weepee
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="weepee", nonce="70bbff1d"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:0479123456@ssw3.brussels.weepee.org:5060> for address/port to send to
set_destination: set destination to 91.208.12.133:5060
Transmitting (NAT) to 91.208.12.133:5060:
ACK sip:0479123456@ssw3.brussels.weepee.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK2ba350b0;rport
Max-Forwards: 70
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as0ae2d0e2
To: <sip:0479123456@ssw3.brussels.weepee.org:5060>;tag=as46ec6919
Contact: <sip:329909011639@192.168.0.184:5060>
Call-ID: 3ee5f692663c3da21c809d3725af567c@ssw3.brussels.weepee.org
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Content-Length: 0


---
Audio is at 17872
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 91.208.12.133:5060:
INVITE sip:0479123456@ssw3.brussels.weepee.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK6591f7f4;rport
Max-Forwards: 70
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as0ae2d0e2
To: <sip:0479123456@ssw3.brussels.weepee.org:5060>
Contact: <sip:329909011639@192.168.0.184:5060>
Call-ID: 3ee5f692663c3da21c809d3725af567c@ssw3.brussels.weepee.org
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="329909011639", realm="weepee", algorithm=MD5, uri="sip:0479123456@ssw3.brussels.weepee.org:5060", nonce="70bbff1d", response="0ab6f3d132b7baf75ecb635902703ab8"
Date: Fri, 03 Aug 2012 17:56:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 2098828985 2098828986 IN IP4 192.168.0.184
s=Asterisk PBX 1.8.11.1-1digium1~lucid
c=IN IP4 192.168.0.184
t=0 0
m=audio 17872 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:91.208.12.133:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK6591f7f4;received=178.117.103.101;rport=1719
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as0ae2d0e2
To: <sip:0479123456@ssw3.brussels.weepee.org:5060>
Call-ID: 3ee5f692663c3da21c809d3725af567c@ssw3.brussels.weepee.org
CSeq: 103 INVITE
Server: weepee
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0479123456@91.208.12.133:5060>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:91.208.12.133:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK6591f7f4;received=178.117.103.101;rport=1719
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as0ae2d0e2
To: <sip:0479123456@ssw3.brussels.weepee.org:5060>;tag=as3f2871d6
Call-ID: 3ee5f692663c3da21c809d3725af567c@ssw3.brussels.weepee.org
CSeq: 103 INVITE
Server: weepee
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0479123456@91.208.12.133:5060>
Content-Type: application/sdp
Content-Length: 275

v=0
o=root 1015261259 1015261259 IN IP4 91.208.12.133
s=weepee
c=IN IP4 91.208.12.133
t=0 0
m=audio 22028 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 13 lines) ---
list_route: hop: <sip:0479123456@91.208.12.133:5060>
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 91.208.12.133:22028
Audio is at 12584
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 192.168.111.76:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.111.76:5060;branch=z9hG4bK2bf267c4;received=192.168.111.76;rport=5060
From: "202" <sip:206@192.168.111.1>;tag=001a6c7b607400212d8f9701-2aedb5a7
To: <sip:70479123456@192.168.111.1>;tag=as1b89ed5c
Call-ID: 001a6c7b-6074001b-575a3247-02dce10b@192.168.111.76
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.11.1-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:70479123456@192.168.111.1:5060>
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 1717487775 1717487775 IN IP4 192.168.111.1
s=Asterisk PBX 1.8.11.1-1digium1~lucid
c=IN IP4 192.168.111.1
t=0 0
m=audio 12584 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:192.168.111.1:5061 --->
jaK
<------------->

<--- SIP read from UDP:91.208.12.133:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK6591f7f4;received=178.117.103.101;rport=1719
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as0ae2d0e2
To: <sip:0479123456@ssw3.brussels.weepee.org:5060>;tag=as3f2871d6
Call-ID: 3ee5f692663c3da21c809d3725af567c@ssw3.brussels.weepee.org
CSeq: 103 INVITE
Server: weepee
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Circuit/channel congestion
X-Asterisk-HangupCauseCode: 34
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
set_destination: Parsing <sip:0479123456@91.208.12.133:5060> for address/port to send to
set_destination: set destination to 91.208.12.133:5060
Transmitting (NAT) to 91.208.12.133:5060:
ACK sip:0479123456@91.208.12.133:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK6591f7f4;rport
Max-Forwards: 70
From: "pl_177" <sip:329909011639@ssw3.brussels.weepee.org>;tag=as0ae2d0e2
To: <sip:0479123456@ssw3.brussels.weepee.org:5060>;tag=as3f2871d6
Contact: <sip:329909011639@192.168.0.184:5060>
Call-ID: 3ee5f692663c3da21c809d3725af567c@ssw3.brussels.weepee.org
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Content-Length: 0


---
Really destroying SIP dialog '3ee5f692663c3da21c809d3725af567c@ssw3.brussels.weepee.org' Method: INVITE

[cfernandes]

humm


try to enable  nat   for cisco phones   on table sccpdevice
 

[brononius]

try to enable  nat   for cisco phones   on table sccpdevice

in sccpdevice, i've changed nat from 'off' to 'on'.
Reloaded linuxmce, restarted the service asterisk and the phone.

But no luck... 

[cfernandes]

LinuxMCE after restart you can confirm that on sccpdevice continues 'on'

[brononius]

LinuxMCE after restart you can confirm that on sccpdevice continues 'on'

After a complete reboot of the machine, the value for NAT in sccpdevice is still 'on'.
I've also tried a reboot of the phone, but ends up with the sam result.
- an internal call towards the provider works.
- an external call over that provider fails with 'all circuits are busy'...

[cfernandes]

you can ask your support provider that is coming to him,

deias'm running out, I made a setup this weekend like his work and in my case Out The Box

[brononius]

you can ask your support provider

I've opened a support ticket by WeePee. A bit bad experience, last time it took 2 weeks before they answered with: "we don't have any issuse, the problem will be at your side".
But maybe today is a better day for their helpdesk...  

I just hope they don't start to be difficult when we use a 'non-standard' solution. With this i mean that we don't have a full asterisk admin page...

[brononius]

And it was a better day !!!   

I'm a bit ashamed to write this. But the problem appears to be solved. Apperantly when you change your asterisk installation, the SIP provider sees another 'SIP device' on my end. So there have to be a 'reset User Agent' at the SIP provider side.

Once they did this (apperantly i could do it myself on the user admin page of them), my calls are perfectly routed...


To remember:
If a call to the provider itself work (short numbers for testing/accounting...), your setup is OK. The problem is further away...


Thanks a lot for you assistance in this!!!

[cfernandes]

I'm glad the solution.
  even if it has not helped much.