You need to ask your carrier what you are not sending them that they would
like. It's usually a fromdomain or authname.
----- Original Message -----
From: Aaron chen
To: Asterisk Users Mailing List - Non-Commercial Discussion ; Asterisk
Developers Mailing List
Sent: Friday, March 26, 2010 09:22
Subject: [asterisk-users] SIP/2.0 403 Forbidden
hi,all
when i send a call to other server use SIP trunk,
i got this below,
<--- SIP read from 222.46.18.52:5060 --->
SIP/2.0 403 Forbidden
what's rong with is?
Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <marks...@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': Found
Connected to Asterisk 1.4.21.2 currently running on gd-branch (pid = 3145)
Verbosity is at least 3
-- Executing [015921256...@from-internal:1] Set("SIP/75002-b7705298",
"MOHCLASS=none") in new stack
-- Executing [015921256...@from-internal:2] Macro("SIP/75002-b7705298",
"user-callerid|SKIPTTL|") in new stack
-- Executing [...@macro-user-callerid:1] Set("SIP/75002-b7705298",
"AMPUSER=75002") in new stack
-- Executing [...@macro-user-callerid:2] GotoIf("SIP/75002-b7705298",
"0?report") in new stack
-- Executing [...@macro-user-callerid:3] ExecIf("SIP/75002-b7705298",
"1|Set|REALCALLERIDNUM=75002") in new stack
-- Executing [...@macro-user-callerid:4] Set("SIP/75002-b7705298",
"AMPUSER=75002") in new stack
-- Executing [...@macro-user-callerid:5] Set("SIP/75002-b7705298",
"AMPUSERCIDNAME=75002") in new stack
-- Executing [...@macro-user-callerid:6] GotoIf("SIP/75002-b7705298",
"0?report") in new stack
-- Executing [...@macro-user-callerid:7] Set("SIP/75002-b7705298",
"AMPUSERCID=75002") in new stack
-- Executing [...@macro-user-callerid:8] Set("SIP/75002-b7705298",
"CALLERID(all)="75002" <75002>") in new stack
-- Executing [...@macro-user-callerid:9] ExecIf("SIP/75002-b7705298",
"0|Set|CHANNEL(language)=") in new stack
-- Executing [...@macro-user-callerid:10] GotoIf("SIP/75002-b7705298",
"1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [...@macro-user-callerid:19] NoOp("SIP/75002-b7705298",
"Using CallerID "75002" <75002>") in new stack
-- Executing [015921256...@from-internal:3] Set("SIP/75002-b7705298",
"_NODEST=") in new stack
-- Executing [015921256...@from-internal:4] Macro("SIP/75002-b7705298",
"record-enable|75002|OUT|") in new stack
-- Executing [...@macro-record-enable:1] GotoIf("SIP/75002-b7705298",
"1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [...@macro-record-enable:4] AGI("SIP/75002-b7705298",
"recordingcheck|20100326-141638|1269584198.62") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20100326-141638|1269584198.62: Outbound recording enabled.
recordingcheck|20100326-141638|1269584198.62:
CALLFILENAME=OUT75002-20100326-141638-1269584198.62
-- AGI Script recordingcheck completed, returning 0
-- Executing [...@macro-record-enable:999]
MixMonitor("SIP/75002-b7705298",
"/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav||")
in new stack
-- Executing [...@macro-record-enable:1000] Set("SIP/75002-b7705298",
"RecordingFileName=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav")
in new stack
-- Executing [...@macro-record-enable:1001] NoOp("SIP/75002-b7705298",
"/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav")
in new stack
-- Executing [...@macro-record-enable:1002] Set("SIP/75002-b7705298",
"CDR(userfield)=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav")
in new stack
-- Executing [015921256...@from-internal:5] Macro("SIP/75002-b7705298",
"dialout-trunk|7|015921256331||") in new stack
-- Executing [...@macro-dialout-trunk:1] Set("SIP/75002-b7705298",
"DIAL_TRUNK=7") in new stack
-- Executing [...@macro-dialout-trunk:2] GosubIf("SIP/75002-b7705298",
"0?sub-pincheck|s|1") in new stack
-- Executing [...@macro-dialout-trunk:3] GotoIf("SIP/75002-b7705298",
"0?disabletrunk|1") in new stack
-- Executing [...@macro-dialout-trunk:4] Set("SIP/75002-b7705298",
"DIAL_NUMBER=015921256331") in new stack
-- Executing [...@macro-dialout-trunk:5] Set("SIP/75002-b7705298",
"DIAL_TRUNK_OPTIONS=Ttr") in new stack
-- Executing [...@macro-dialout-trunk:6] Set("SIP/75002-b7705298",
"OUTBOUND_GROUP=OUT_7") in new stack
-- Executing [...@macro-dialout-trunk:7] GotoIf("SIP/75002-b7705298",
"1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [...@macro-dialout-trunk:9] GotoIf("SIP/75002-b7705298",
"0?skipoutcid") in new stack
-- Executing [...@macro-dialout-trunk:10] Set("SIP/75002-b7705298",
"DIAL_TRUNK_OPTIONS=Tt") in new stack
== Begin MixMonitor Recording SIP/75002-b7705298
-- Executing [...@macro-dialout-trunk:11] Macro("SIP/75002-b7705298",
"outbound-callerid|7") in new stack
-- Executing [...@macro-outbound-callerid:1]
ExecIf("SIP/75002-b7705298", "0|SetCallerPres|") in new stack
-- Executing [...@macro-outbound-callerid:2]
ExecIf("SIP/75002-b7705298", "0|Set|REALCALLERIDNUM=75002") in new stack
-- Executing [...@macro-outbound-callerid:3]
GotoIf("SIP/75002-b7705298", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [...@macro-outbound-callerid:6] Set("SIP/75002-b7705298",
"USEROUTCID=") in new stack
-- Executing [...@macro-outbound-callerid:7] Set("SIP/75002-b7705298",
"EMERGENCYCID=") in new stack
-- Executing [...@macro-outbound-callerid:8] Set("SIP/75002-b7705298",
"TRUNKOUTCID=s2") in new stack
-- Executing [...@macro-outbound-callerid:9]
GotoIf("SIP/75002-b7705298", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [...@macro-outbound-callerid:12]
ExecIf("SIP/75002-b7705298", "1|Set|CALLERID(all)=s2") in new stack
-- Executing [...@macro-outbound-callerid:13]
ExecIf("SIP/75002-b7705298", "0|Set|CALLERID(all)=") in new stack
-- Executing [...@macro-outbound-callerid:14]
ExecIf("SIP/75002-b7705298", "0|SetCallerPres|prohib_passed_screen") in new
stack
-- Executing [...@macro-dialout-trunk:12] ExecIf("SIP/75002-b7705298",
"0|AGI|fixlocalprefix") in new stack
-- Executing [...@macro-dialout-trunk:13] Set("SIP/75002-b7705298",
"OUTNUM=015921256331") in new stack
-- Executing [...@macro-dialout-trunk:14] Set("SIP/75002-b7705298",
"custom=SIP/s2") in new stack
-- Executing [...@macro-dialout-trunk:15] ExecIf("SIP/75002-b7705298",
"1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack
-- Executing [...@macro-dialout-trunk:16] Macro("SIP/75002-b7705298",
"dialout-trunk-predial-hook|") in new stack
-- Executing [...@macro-dialout-trunk-predial-hook:1]
MacroExit("SIP/75002-b7705298", "") in new stack
-- Executing [...@macro-dialout-trunk:17] GotoIf("SIP/75002-b7705298",
"0?bypass|1") in new stack
-- Executing [...@macro-dialout-trunk:18] GotoIf("SIP/75002-b7705298",
"0?customtrunk") in new stack
-- Executing [...@macro-dialout-trunk:19] Dial("SIP/75002-b7705298",
"SIP/s2/015921256331|300|M(setmusic^none)Tt") in new stack
Audio is at 219.235.234.238 port 17136
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 222.46.18.52:5060:
INVITE sip:015921256...@222.46.18.52 SIP/2.0
Via: SIP/2.0/UDP 219.235.234.238:5060;branch=z9hG4bK368b5ad8;rport
From: "s2" <sip:unkn...@222.46.18.52>;tag=as75543a2d
To: <sip:015921256...@222.46.18.52>
Contact: <sip:unkn...@219.235.234.238>
Call-ID: 5cf71e106209cf65344e24031354f...@222.46.18.52
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 26 Mar 2010 06:16:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 3145 3145 IN IP4 219.235.234.238
s=session
c=IN IP4 219.235.234.238
t=0 0
m=audio 17136 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called s2/015921256331
gd-branch*CLI>
<--- SIP read from 222.46.18.52:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
219.235.234.238:5060;branch=z9hG4bK368b5ad8;received=58.247.12.18;rport=11028
From: "s2" <sip:unkn...@222.46.18.52>;tag=as75543a2d
To: <sip:015921256...@222.46.18.52>
Contact: <sip:015921256...@222.46.18.52:5060>
Call-ID: 5cf71e106209cf65344e24031354f...@222.46.18.52
CSeq: 102 INVITE
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: timer
Server: VOS2009 V2.1.1.8
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 222.46.18.52:5060:
ACK sip:015921256...@222.46.18.52 SIP/2.0
Via: SIP/2.0/UDP 219.235.234.238:5060;branch=z9hG4bK368b5ad8;rport
From: "s2" <sip:unkn...@222.46.18.52>;tag=as75543a2d
To: <sip:015921256...@222.46.18.52>
Contact: <sip:unkn...@219.235.234.238>
Call-ID: 5cf71e106209cf65344e24031354f...@222.46.18.52
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/s2-088f72e8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [...@macro-dialout-trunk:20] Goto("SIP/75002-b7705298",
"s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-congest...@macro-dialout-trunk:1]
GotoIf("SIP/75002-b7705298", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [s-congest...@macro-dialout-trunk:3]
NoOp("SIP/75002-b7705298", "TRUNK Dial failed due to CONGESTION - failing
through to other trunks") in new stack
-- Executing [015921256...@from-internal:6] Macro("SIP/75002-b7705298",
"outisbusy|") in new stack
-- Executing [...@macro-outisbusy:1] Playback("SIP/75002-b7705298",
"all-circuits-busy-now|noanswer") in new stack
-- <SIP/75002-b7705298> Playing 'all-circuits-busy-now' (language 'en')
Really destroying SIP dialog
'5cf71e106209cf65344e24031354f...@222.46.18.52' Method: INVITE
-- Executing [...@macro-outisbusy:2] Playback("SIP/75002-b7705298",
"pls-try-call-later|noanswer") in new stack
-- <SIP/75002-b7705298> Playing 'pls-try-call-later' (language 'en')
-- Executing [...@macro-outisbusy:3] Macro("SIP/75002-b7705298",
"hangupcall") in new stack
-- Executing [...@macro-hangupcall:1] GotoIf("SIP/75002-b7705298",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [...@macro-hangupcall:4] GotoIf("SIP/75002-b7705298",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [...@macro-hangupcall:7] GotoIf("SIP/75002-b7705298",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [...@macro-hangupcall:9] Hangup("SIP/75002-b7705298", "")
in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/75002-b7705298' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/75002-b7705298' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/75002-b7705298'
== End MixMonitor Recording SIP/75002-b7705298
--
Best Regards!
Aaron Chen
------------------------------------------------------------------------------
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
