Hey,
Is there a way to update/change the dialplan under LinuxMCE 1004?
I would like to control/limit my outgoing calls to only national ones.
So i need something in the form of:
- fixed number: 01 234 56 78
- cell phones: 0123 45 67 89
And since its a small home, an outgoing prefix is also a bit 'much'... ;)
As for the prefix, a ticket is made for it: http://svn.linuxmce.org/trac.cgi/ticket/1509
I've got my SIP provider registered in the system.
I can recieve incoming calls (perfectly on all my phones).
But i can't dial out?
When i try a number directly, with prefixes... the systems tells me: "All circuits are busy now", followed by a busy tone.
And on the cisco display, i'm getting: "Session Progress (in 183)".
I've tried following combinations (where ABCDEFGHIJ my cell number is):
- ABCDEFGHIJ
- 0 ABCDEFGHIJ
- 9 ABCDEFGHIJ
- 7 ABCDEFGHIJ
Any idea what i'm missing?
Hi Brononius
can you post asterisk log ? /var/log/asterisk/full
I'm glad to, but it's empty. :$
I've cleared yesterday the log file to have a more clear view what happens.
But when i try to call outside, nothing enter the log file for that call.
When i call something internal (over an orbiter since i'm not a home), the log file is filled. So the log file is working.
[Aug 1 12:51:37] WARNING[27152] file.c: Unable to open all-circuits-busy-now (format 0x0 (nothing)): No such file or directory
[Aug 1 12:51:37] WARNING[27152] app_playback.c: ast_streamfile failed on OutgoingSpoolFailed for all-circuits-busy-now,noanswer
[Aug 1 12:51:37] WARNING[27152] file.c: Unable to open pls-try-call-later (format 0x0 (nothing)): No such file or directory
[Aug 1 12:51:37] WARNING[27152] app_playback.c: ast_streamfile failed on OutgoingSpoolFailed for pls-try-call-later,noanswer
And in the 'Call Detail Records' of LinuxMCE, I see nicely the call being logged...
on asterisk console
asterisk -r
enable the core verbose and sip debug to show more details
core set verbose 9
sip set debug on
Offf, a lot of output with 'sip set debug on'
<--- SIP read from UDP:192.168.111.71:5060 --->
INVITE sip:01234567890@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.71:5060;branch=z9hG4bK6e69a5a9
From: "210" <sip:201@192.168.111.1>;tag=00127fc24bf300164304e48a-71ed853c
To: <sip:01234567890@192.168.111.1>
Call-ID: 00127fc2-4bf30005-41477508-4c847d7f@192.168.111.71
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:201@192.168.111.71:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "210" <sip:201@192.168.111.1>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 279
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 5418 0 IN IP4 192.168.111.71
s=SIP Call
t=0 0
m=audio 23122 RTP/AVP 0 8 18 101
c=IN IP4 192.168.111.71
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 192.168.111.71:5060 (NAT)
Using INVITE request as basis request - 00127fc2-4bf30005-41477508-4c847d7f@192.168.111.71
Found peer '201' for '201' from 192.168.111.71:5060
<--- Reliably Transmitting (NAT) to 192.168.111.71:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.111.71:5060;branch=z9hG4bK6e69a5a9;received=192.168.111.71;rport=5060
From: "210" <sip:201@192.168.111.1>;tag=00127fc24bf300164304e48a-71ed853c
To: <sip:01234567890@192.168.111.1>;tag=as44f96cdc
Call-ID: 00127fc2-4bf30005-41477508-4c847d7f@192.168.111.71
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="7e5a343e"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '00127fc2-4bf30005-41477508-4c847d7f@192.168.111.71' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.111.71:5060 --->
ACK sip:01234567890@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.71:5060;branch=z9hG4bK6e69a5a9
From: "210" <sip:201@192.168.111.1>;tag=00127fc24bf300164304e48a-71ed853c
To: <sip:01234567890@192.168.111.1>;tag=as44f96cdc
Call-ID: 00127fc2-4bf30005-41477508-4c847d7f@192.168.111.71
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:192.168.111.71:5060 --->
INVITE sip:01234567890@192.168.111.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.71:5060;branch=z9hG4bK0aaa45a8
From: "210" <sip:201@192.168.111.1>;tag=00127fc24bf300164304e48a-71ed853c
To: <sip:01234567890@192.168.111.1>
Call-ID: 00127fc2-4bf30005-41477508-4c847d7f@192.168.111.71
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:201@192.168.111.71:5060;transport=udp>
Authorization: Digest username="201",realm="asterisk",uri="sip:01234567890@192.168.111.1",response="47dc2d890aa7f250fb7260864ec5a957",nonce="7e5a343e",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "210" <sip:201@192.168.111.1>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 279
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 5418 0 IN IP4 192.168.111.71
s=SIP Call
t=0 0
m=audio 23122 RTP/AVP 0 8 18 101
c=IN IP4 192.168.111.71
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 192.168.111.71:5060 (NAT)
Using INVITE request as basis request - 00127fc2-4bf30005-41477508-4c847d7f@192.168.111.71
Found peer '201' for '201' from 192.168.111.71:5060
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 192.168.111.71:23122
Looking for 01234567890 in from-internal (domain 192.168.111.1)
list_route: hop: <sip:201@192.168.111.71:5060;transport=udp>
<--- Transmitting (NAT) to 192.168.111.71:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.111.71:5060;branch=z9hG4bK0aaa45a8;received=192.168.111.71;rport=5060
From: "210" <sip:201@192.168.111.1>;tag=00127fc24bf300164304e48a-71ed853c
To: <sip:01234567890@192.168.111.1>
Call-ID: 00127fc2-4bf30005-41477508-4c847d7f@192.168.111.71
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:01234567890@192.168.111.1:5060>
Content-Length: 0
<------------>
Audio is at 16210
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:1234567890@ssw3.brussels.weepee.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK226a2160;rport
Max-Forwards: 70
From: "pl_172" <sip:329923232323@ssw3.brussels.weepee.org>;tag=as038cadc7
To: <sip:1234567890@ssw3.brussels.weepee.org:5060>
Contact: <sip:329923232323@192.168.0.184:5060>
Call-ID: 3be558c25424cc7d0e296fd27683ddfe@ssw3.brussels.weepee.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Date: Wed, 01 Aug 2012 15:58:55 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 762922019 762922019 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 16210 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=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
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="weepee", nonce="3c56a66a"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:1234567890@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:1234567890@ssw3.brussels.weepee.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK226a2160;rport
Max-Forwards: 70
From: "pl_172" <sip:329923232323@ssw3.brussels.weepee.org>;tag=as038cadc7
To: <sip:1234567890@ssw3.brussels.weepee.org:5060>;tag=as0c5744cc
Contact: <sip:329923232323@192.168.0.184:5060>
Call-ID: 3be558c25424cc7d0e296fd27683ddfe@ssw3.brussels.weepee.org
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Content-Length: 0
---
Audio is at 16210
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:1234567890@ssw3.brussels.weepee.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK048474be;rport
Max-Forwards: 70
From: "pl_172" <sip:329923232323@ssw3.brussels.weepee.org>;tag=as038cadc7
To: <sip:1234567890@ssw3.brussels.weepee.org:5060>
Contact: <sip:329923232323@192.168.0.184:5060>
Call-ID: 3be558c25424cc7d0e296fd27683ddfe@ssw3.brussels.weepee.org
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Authorization: Digest username="329923232323", realm="weepee", algorithm=MD5, uri="sip:1234567890@ssw3.brussels.weepee.org:5060", nonce="3c56a66a", response="d084bef8f1dce646cc43de2d378047d5"
Date: Wed, 01 Aug 2012 15:58:55 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 762922019 762922020 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 16210 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=z9hG4bK048474be;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>
Call-ID: 3be558c25424cc7d0e296fd27683ddfe@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:1234567890@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=z9hG4bK048474be;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=as68f4b07c
Call-ID: 3be558c25424cc7d0e296fd27683ddfe@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:1234567890@91.208.12.133:5060>
Content-Type: application/sdp
Content-Length: 275
v=0
o=root 1473405314 1473405314 IN IP4 91.208.12.133
s=weepee
c=IN IP4 91.208.12.133
t=0 0
m=audio 30640 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:1234567890@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 91.208.12.133:30640
Audio is at 18566
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 192.168.111.71:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.111.71:5060;branch=z9hG4bK0aaa45a8;received=192.168.111.71;rport=5060
From: "210" <sip:201@192.168.111.1>;tag=00127fc24bf300164304e48a-71ed853c
To: <sip:01234567890@192.168.111.1>;tag=as1327a30c
Call-ID: 00127fc2-4bf30005-41477508-4c847d7f@192.168.111.71
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:01234567890@192.168.111.1:5060>
Content-Type: application/sdp
Content-Length: 303
v=0
o=root 124011141 124011141 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 18566 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/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK048474be;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=as68f4b07c
Call-ID: 3be558c25424cc7d0e296fd27683ddfe@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:1234567890@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:1234567890@91.208.12.133:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.184:5060;branch=z9hG4bK048474be;rport
Max-Forwards: 70
From: "pl_172" <sip:329923232323@ssw3.brussels.weepee.org>;tag=as038cadc7
To: <sip:1234567890@ssw3.brussels.weepee.org:5060>;tag=as68f4b07c
Contact: <sip:329923232323@192.168.0.184:5060>
Call-ID: 3be558c25424cc7d0e296fd27683ddfe@ssw3.brussels.weepee.org
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.11.1-1digium1~lucid
Content-Length: 0
---
Really destroying SIP dialog '3be558c25424cc7d0e296fd27683ddfe@ssw3.brussels.weepee.org' Method: INVITE
dcerouter*CLI>
Disconnected from Asterisk server
And with the debug:
== Using SIP RTP CoS mark 5
-- Executing [01234567890@from-internal:1] Macro("SIP/201-00000006", "dialout-trunk,SIP/003212121212,1234567890,,")
-- Executing [s@macro-dialout-trunk:1] Set("SIP/201-00000006", "DIAL_TRUNK=SIP/003212121212") in new stack
-- Executing [s@macro-dialout-trunk:2] Set("SIP/201-00000006", "DIAL_NUMBER=1234567890") in new stack
-- Executing [s@macro-dialout-trunk:3] Set("SIP/201-00000006", "ROUTE_PASSWD=") in new stack
-- Executing [s@macro-dialout-trunk:4] GotoIf("SIP/201-00000006", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing [s@macro-dialout-trunk:6] GotoIf("SIP/201-00000006", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:7] Set("SIP/201-00000006", "_NODEST=") in new stack
-- Executing [s@macro-dialout-trunk:8] Set("SIP/201-00000006", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:9] Set("SIP/201-00000006", "GROUP()=DIAL_TRUNK") in new stack
-- Executing [s@macro-dialout-trunk:10] Macro("SIP/201-00000006", "user-callerid,SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/201-00000006", "user-callerid: device 201") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/201-00000006", "AMPUSER=201") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/201-00000006", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] GotoIf("SIP/201-00000006", "0?start") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/201-00000006", "REALCALLERIDNUM=201") in new stack
-- Executing [s@macro-user-callerid:6] NoOp("SIP/201-00000006", "REALCALLERIDNUM is 201") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/201-00000006", "AMPUSER=201") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/201-00000006", "AMPUSERCIDNAME=pl_172") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/201-00000006", "0?report") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/201-00000006", "AMPUSERCID=201") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/201-00000006", "CALLERID(all)="pl_172" <201>") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/201-00000006", "REALCALLERIDNUM=201") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/201-00000006", "TTL: ARG1: SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/201-00000006", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/201-00000006", "Using CallerID "pl_172" <201>") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/201-00000006", "record-enable,201,OUT") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/201-00000006", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/201-00000006", "recordingcheck,20120801-180052,1343836852.6") in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck
recordingcheck,20120801-180052,1343836852.6: Outbound recording not enabled
-- <SIP/201-00000006>AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/201-00000006", "No recording needed") in new stack
-- Executing [s@macro-dialout-trunk:12] GotoIf("SIP/201-00000006", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/201-00000006", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:14] Macro("SIP/201-00000006", "outbound-callerid,SIP/003212121212") in new stack
-- Executing [s@macro-outbound-callerid:1] GotoIf("SIP/201-00000006", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing [s@macro-outbound-callerid:3] NoOp("SIP/201-00000006", "REALCALLERIDNUM is 201") in new stack
-- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/201-00000006", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing [s@macro-outbound-callerid:9] Set("SIP/201-00000006", "USEROUTCID="pl_172" <201>") in new stack
-- Executing [s@macro-outbound-callerid:10] Set("SIP/201-00000006", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:11] Set("SIP/201-00000006", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:12] GotoIf("SIP/201-00000006", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing [s@macro-outbound-callerid:16] GotoIf("SIP/201-00000006", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing [s@macro-outbound-callerid:18] GotoIf("SIP/201-00000006", "0?report") in new stack
-- Executing [s@macro-outbound-callerid:19] Set("SIP/201-00000006", "CALLERID(all)="pl_172" <201>") in new stack
-- Executing [s@macro-outbound-callerid:20] GotoIf("SIP/201-00000006", "1?report:hidecid") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing [s@macro-outbound-callerid:22] NoOp("SIP/201-00000006", "CallerID set to "pl_172" <201>") in new stack
-- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/201-00000006", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing [s@macro-dialout-trunk:17] Set("SIP/201-00000006", "OUTNUM=1234567890") in new stack
-- Executing [s@macro-dialout-trunk:18] Set("SIP/201-00000006", "custom=SIP/003212121212") in new stack
-- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/201-00000006", "1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,22)
-- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/201-00000006", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:23] Dial("SIP/201-00000006", "SIP/003212121212/1234567890,300,") in new stack
== 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
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:24] Goto("SIP/201-00000006", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/201-00000006", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/201-00000006", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing [01234567890@from-internal:2] Macro("SIP/201-00000006", "outisbusy,")
-- Executing [s@macro-outisbusy:1] Playback("SIP/201-00000006", "all-circuits-busy-now,noanswer") in new stack
-- <SIP/201-00000006> Playing 'all-circuits-busy-now.gsm' (language 'en')
dcerouter*CLI>
cal you post a result of this query on database asterisk
select * from extensions where context='outbound-allroutes'
Here we go:
mysql> select * from extensions where context='outbound-allroutes';
+--------+--------------------+-------+----------+-------+---------------------------------------------+
| id | context | exten | priority | app | appdata |
+--------+--------------------+-------+----------+-------+---------------------------------------------+
| 222382 | outbound-allroutes | 100 | 1 | Macro | dialout-trunk,SIP/003212121212,${EXTEN},, |
| 222383 | outbound-allroutes | 100 | 2 | Macro | outisbusy, |
| 222384 | outbound-allroutes | 101 | 1 | Macro | dialout-trunk,SIP/003212121212,${EXTEN},, |
| 222385 | outbound-allroutes | 101 | 2 | Macro | outisbusy, |
| 222386 | outbound-allroutes | _. | 1 | Macro | dialout-trunk,SIP/003212121212,${EXTEN:1},, |
| 222387 | outbound-allroutes | _. | 2 | Macro | outisbusy, |
+--------+--------------------+-------+----------+-------+---------------------------------------------+
6 rows in set (0.00 sec)
ps i've change my phonenumber with 12121212 in the whole thread so i didn't put my private number in here... :$
i think that exten 100 and 101 is that problem on my
show
112 and 911
this numbers is for emergency numbers
and 100 is goto voicemail and 101 goto lmcd-phonebook
try to change to 112 and 911
Didn't help... :-[
Maybe it has something to do with the nat feature? I know that in 8.04, i needed "nat: 1" in the sipMAC file.
I just added it (since it wasn't in there), but it doensn't solve it (damned).
/tftpboot/SIP001A6C7B6074.cnf
# Global phone
# ------------
phone_label: "...Internal 206..."
phone_prompt: "Tel 206"
phone_password: "1234"
logo_url: "http://10.10.10.1/phones/206.bmp"
directory_url: "http://10.10.10.1/phones/directory.xml"
nat_enable: "1"
# Line 1
# ------
line1_name: "206"
line1_authname: "206"
line1_password: "oskv%4dwawu-02y3"
line1_shortname: "206"
Did you actually set a number for prefix in the webadmin setup? In my case i set a 0 (zero) So if im to dial number abcdef i dial 0abcdef
I've tried with (a 0, a 9...), and without.
But didn't change a lot.
What i find a bit strange in the logs, is the rule "SIP/2.0 401 Unauthorized" when i try to call.
Like i'm not allowed to call?
I have already change my phonenumber from 09XXXXXXX towards 00329XXXXXXX. Becuase i found somewhere on the web that maybe the provider blocks my source?
Is there a way i can easy troubleshoot/simulation between the linuxmce and my SIP provider? :o
Fe /etc/asterisk/call_SIP -phonenumber 0145525252
on your log i can see
Got SIP response 503 "Service Unavailable" back from 91.208.12.133:5060
maybe a codec problem between linuxmce and your ISP or NAT between linuxmce and ISP
Incoming calls are working fine.
So can it still be a codec issue?
Where should i check/change this in the new version?
(a pitty that we don't have a complete asterisk admin page for fallback...)
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
Quote from: cfernandes on August 03, 2012, 01:13:57 PM
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. :-[
Quote from: cfernandes on August 03, 2012, 01:13:57 PM
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?
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')
The rule is in the database. But didn't worked... :$
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:
[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
---
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
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')
Quote from: cfernandes on August 03, 2012, 07:09:32 PM
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):
<--- 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):
<--- 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
humm
try to enable nat for cisco phones on table sccpdevice
Quote from: cfernandes on August 03, 2012, 09:24:51 PMtry 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... :'(
LinuxMCE after restart you can confirm that on sccpdevice continues 'on'
Quote from: cfernandes on August 04, 2012, 02:28:50 PM
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'...
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
Quote from: cfernandes on August 06, 2012, 12:24:46 PM
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... :P
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...
And it was a better day !!! :P
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!!!
I'm glad the solution.
even if it has not helped much.
Hi,
my current dialplan is:
mysql> select * from extensions where context='outbound-allroutes'
-> ;
+------+--------------------+-------+----------+-------+---------------------------------------------+
| id | context | exten | priority | app | appdata |
+------+--------------------+-------+----------+-------+---------------------------------------------+
| 5936 | outbound-allroutes | 190 | 1 | Macro | dialout-trunk,/,${EXTEN},, |
| 5937 | outbound-allroutes | 190 | 2 | Macro | outisbusy, |
| 5938 | outbound-allroutes | 193 | 1 | Macro | dialout-trunk,/,${EXTEN},, |
| 5939 | outbound-allroutes | 193 | 2 | Macro | outisbusy, |
| 5976 | outbound-allroutes | _7. | 1 | Macro | dialout-trunk,SIP/054138594676,${EXTEN:1},, |
| 5977 | outbound-allroutes | _7. | 2 | Macro | outisbusy, |
| 5958 | outbound-allroutes | _8. | 1 | Macro | dialout-trunk,SIP/2122498618,${EXTEN:1},, |
| 5959 | outbound-allroutes | _8. | 2 | Macro | outisbusy, |
| 5940 | outbound-allroutes | _9. | 1 | Macro | dialout-trunk,SIP/2062036594,${EXTEN:1},, |
| 5941 | outbound-allroutes | _9. | 2 | Macro | outisbusy, |
+------+--------------------+-------+----------+-------+---------------------------------------------+
The trunks:
1 sipgate (dialout-trunk,SIP/054138594676)
2 spa3102 (dialout-trunk,SIP/2122498618)
3 voipcheap (dialout-trunk,SIP/2062036594)
The plans (by trunk)
1 - sipgate
dial rules:
00+XXXXXXX.
outbound:
900|XXXXXXX.
trunk sequence: voipceap, sipgate
2 - spa3102
dial rules:
XXXXXXXX
08+08|00XXXXX.
005521|XXXXXXXX
031+0055|XXXXXXXXXX
031+0|XXXXXXXXXX
031+XXXXXXXXXX
outbound:
121|XXXXXXXX
19X
1|XXXXXXXXXX
9|0055ZXXXXXXXXX
9|0800XXXXX.
9|0ZXNXXXXXXX
9|NXXXXXXX
9|ZXX
trunk sequence: spa3102
3 - voipcheap
dial rules:
00+XXXXXXX.
outbound:
800|XXXXXXX.
900|XXXXXXX.
trunk sequence: voipcheap, sipgate
Ok, how do i insert it in the table, defining the rules, outbound and trunk order, please?
I want to use the prefix 9 for all. For now, as i'm not finding out how, i defined 3 prefixes (uggly :().....
Best regards,
Paulo
At the moment there is no way to specify rules. If you haven't already, feel free to create a feature request.