Author Topic: [SOLVED] Prefix, dialplan in 1004 (Read 20254 times)

brononius · « **on:** July 31, 2012, 10:23:24 am »

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'...

pointman87 · « **Reply #1 on:** July 31, 2012, 01:37:02 pm »

As for the prefix, a ticket is made for it: http://svn.linuxmce.org/trac.cgi/ticket/1509

brononius · « **Reply #2 on:** July 31, 2012, 05:33:02 pm »

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?

cfernandes · « **Reply #3 on:** August 01, 2012, 12:28:29 pm »

Hi Brononius
can you post asterisk log ? /var/log/asterisk/full

brononius · « **Reply #4 on:** August 01, 2012, 12:54:25 pm »

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.

Code: [Select]

[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...

cfernandes · « **Reply #5 on:** August 01, 2012, 01:08:52 pm »

on asterisk console

asterisk -r
enable the core verbose and sip debug to show more details

core set verbose 9
sip set debug on

brononius · « **Reply #6 on:** August 01, 2012, 06:03:25 pm »

Offf, a lot of output with 'sip set debug on'

Code: [Select]

<--- 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:

Code: [Select]

  == 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>

cfernandes · « **Reply #7 on:** August 01, 2012, 07:42:29 pm »

cal you post a result of this query on database asterisk

select * from extensions where context='outbound-allroutes'

brononius · « **Reply #8 on:** August 01, 2012, 08:53:17 pm »

Here we go:

Code: [Select]

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... :$

cfernandes · « **Reply #9 on:** August 01, 2012, 09:28:11 pm »

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

brononius · « **Reply #10 on:** August 02, 2012, 08:48:15 am »

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

Code: [Select]

# 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"

pointman87 · « **Reply #11 on:** August 03, 2012, 03:00:01 am »

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

brononius · « **Reply #12 on:** August 03, 2012, 08:02:00 am »

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?

Fe /etc/asterisk/call_SIP -phonenumber 0145525252

cfernandes · « **Reply #13 on:** August 03, 2012, 12:46:08 pm »

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

brononius · « **Reply #14 on:** August 03, 2012, 12:59:20 pm »

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...)

News:

Author Topic: [SOLVED] Prefix, dialplan in 1004 (Read 20254 times)

brononius

[SOLVED] Prefix, dialplan in 1004

pointman87

Re: Prefix, dialplan in 1004

brononius

Re: Prefix, dialplan in 1004

cfernandes

Re: Prefix, dialplan in 1004

brononius

Re: Prefix, dialplan in 1004

cfernandes

Re: Prefix, dialplan in 1004

brononius

Re: Prefix, dialplan in 1004

cfernandes

Re: Prefix, dialplan in 1004

brononius

Re: Prefix, dialplan in 1004

cfernandes

Re: Prefix, dialplan in 1004

brononius

Re: Prefix, dialplan in 1004

pointman87

Re: Prefix, dialplan in 1004

brononius

Re: Prefix, dialplan in 1004

cfernandes

Re: Prefix, dialplan in 1004

brononius

Re: Prefix, dialplan in 1004