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<DIV><FONT face=Arial size=2>You need to ask your carrier what you are not
sending them that they would like. It's usually a fromdomain or
authname.</FONT></DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=evane1890@gmail.com href="mailto:evane1890@gmail.com">Aaron chen</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">Asterisk Users Mailing List -
Non-Commercial Discussion</A> ; <A title=asterisk-dev@lists.digium.com
href="mailto:asterisk-dev@lists.digium.com">Asterisk Developers Mailing
List</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, March 26, 2010 09:22</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [asterisk-users] SIP/2.0 403
Forbidden</DIV>
<DIV><BR></DIV>
<DIV>hi,all</DIV>
<DIV> </DIV>
<DIV>when i send a call to other server use SIP trunk,</DIV>
<DIV> </DIV>
<DIV>i got this below,</DIV>
<DIV> </DIV>
<DIV><--- SIP read from <A
href="http://222.46.18.52:5060">222.46.18.52:5060</A> ---><BR>SIP/2.0 403
Forbidden</DIV>
<DIV> </DIV>
<DIV>what's rong with is?</DIV>
<DIV> </DIV>
<DIV> </DIV>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<DIV>Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and
others.<BR>Created by Mark Spencer <<A
href="mailto:markster@digium.com">markster@digium.com</A>><BR>Asterisk
comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.<BR>This is free software, with components licensed under the GNU
General Public<BR>License version 2 and other licenses; you are welcome to
redistribute it under<BR>certain conditions. Type 'core show license' for
details.<BR>=========================================================================<BR>
== Parsing '/etc/asterisk/asterisk.conf': Found<BR>Connected to Asterisk
1.4.21.2 currently running on gd-branch (pid = 3145)<BR>Verbosity is at
least 3<BR> -- Executing [015921256331@from-internal:1]
Set("SIP/75002-b7705298", "MOHCLASS=none") in new
stack<BR> -- Executing [015921256331@from-internal:2]
Macro("SIP/75002-b7705298", "user-callerid|SKIPTTL|") in new
stack<BR> -- Executing [s@macro-user-callerid:1]
Set("SIP/75002-b7705298", "AMPUSER=75002") in new
stack<BR> -- Executing [s@macro-user-callerid:2]
GotoIf("SIP/75002-b7705298", "0?report") in new stack<BR>
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/75002-b7705298",
"1|Set|REALCALLERIDNUM=75002") in new stack<BR> --
Executing [s@macro-user-callerid:4] Set("SIP/75002-b7705298",
"AMPUSER=75002") in new stack<BR> -- Executing
[s@macro-user-callerid:5] Set("SIP/75002-b7705298", "AMPUSERCIDNAME=75002")
in new stack<BR> -- Executing [s@macro-user-callerid:6]
GotoIf("SIP/75002-b7705298", "0?report") in new stack<BR>
-- Executing [s@macro-user-callerid:7] Set("SIP/75002-b7705298",
"AMPUSERCID=75002") in new stack<BR> -- Executing
[s@macro-user-callerid:8] Set("SIP/75002-b7705298", "CALLERID(all)="75002"
<75002>") in new stack<BR> -- Executing
[s@macro-user-callerid:9] ExecIf("SIP/75002-b7705298",
"0|Set|CHANNEL(language)=") in new stack<BR> -- Executing
[s@macro-user-callerid:10] GotoIf("SIP/75002-b7705298", "1?continue") in new
stack<BR> -- Goto
(macro-user-callerid,s,19)<BR> -- Executing
[s@macro-user-callerid:19] NoOp("SIP/75002-b7705298", "Using CallerID
"75002" <75002>") in new stack<BR> -- Executing
[015921256331@from-internal:3] Set("SIP/75002-b7705298", "_NODEST=") in new
stack<BR> -- Executing [015921256331@from-internal:4]
Macro("SIP/75002-b7705298", "record-enable|75002|OUT|") in new
stack<BR> -- Executing [s@macro-record-enable:1]
GotoIf("SIP/75002-b7705298", "1?check") in new stack<BR>
-- Goto (macro-record-enable,s,4)<BR> -- Executing
[s@macro-record-enable:4] AGI("SIP/75002-b7705298",
"recordingcheck|20100326-141638|1269584198.62") in new
stack<BR> -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck<BR>
recordingcheck|20100326-141638|1269584198.62: Outbound recording
enabled.<BR> recordingcheck|20100326-141638|1269584198.62:
CALLFILENAME=OUT75002-20100326-141638-1269584198.62<BR> --
AGI Script recordingcheck completed, returning 0<BR> --
Executing [s@macro-record-enable:999] MixMonitor("SIP/75002-b7705298",
"/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav||")
in new stack<BR> -- Executing [s@macro-record-enable:1000]
Set("SIP/75002-b7705298",
"RecordingFileName=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav")
in new stack<BR> -- Executing [s@macro-record-enable:1001]
NoOp("SIP/75002-b7705298",
"/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav")
in new stack<BR> -- Executing [s@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<BR> -- Executing
[015921256331@from-internal:5] Macro("SIP/75002-b7705298",
"dialout-trunk|7|015921256331||") in new stack<BR> --
Executing [s@macro-dialout-trunk:1] Set("SIP/75002-b7705298",
"DIAL_TRUNK=7") in new stack<BR> -- Executing
[s@macro-dialout-trunk:2] GosubIf("SIP/75002-b7705298",
"0?sub-pincheck|s|1") in new stack<BR> -- Executing
[s@macro-dialout-trunk:3] GotoIf("SIP/75002-b7705298", "0?disabletrunk|1")
in new stack<BR> -- Executing [s@macro-dialout-trunk:4]
Set("SIP/75002-b7705298", "DIAL_NUMBER=015921256331") in new
stack<BR> -- Executing [s@macro-dialout-trunk:5]
Set("SIP/75002-b7705298", "DIAL_TRUNK_OPTIONS=Ttr") in new
stack<BR> -- Executing [s@macro-dialout-trunk:6]
Set("SIP/75002-b7705298", "OUTBOUND_GROUP=OUT_7") in new
stack<BR> -- Executing [s@macro-dialout-trunk:7]
GotoIf("SIP/75002-b7705298", "1?nomax") in new stack<BR>
-- Goto (macro-dialout-trunk,s,9)<BR> -- Executing
[s@macro-dialout-trunk:9] GotoIf("SIP/75002-b7705298", "0?skipoutcid") in
new stack<BR> -- Executing [s@macro-dialout-trunk:10]
Set("SIP/75002-b7705298", "DIAL_TRUNK_OPTIONS=Tt") in new stack<BR> ==
Begin MixMonitor Recording SIP/75002-b7705298<BR> --
Executing [s@macro-dialout-trunk:11] Macro("SIP/75002-b7705298",
"outbound-callerid|7") in new stack<BR> -- Executing
[s@macro-outbound-callerid:1] ExecIf("SIP/75002-b7705298",
"0|SetCallerPres|") in new stack<BR> -- Executing
[s@macro-outbound-callerid:2] ExecIf("SIP/75002-b7705298",
"0|Set|REALCALLERIDNUM=75002") in new stack<BR> --
Executing [s@macro-outbound-callerid:3] GotoIf("SIP/75002-b7705298",
"1?normcid") in new stack<BR> -- Goto
(macro-outbound-callerid,s,6)<BR> -- Executing
[s@macro-outbound-callerid:6] Set("SIP/75002-b7705298", "USEROUTCID=") in
new stack<BR> -- Executing [s@macro-outbound-callerid:7]
Set("SIP/75002-b7705298", "EMERGENCYCID=") in new
stack<BR> -- Executing [s@macro-outbound-callerid:8]
Set("SIP/75002-b7705298", "TRUNKOUTCID=s2") in new
stack<BR> -- Executing [s@macro-outbound-callerid:9]
GotoIf("SIP/75002-b7705298", "1?trunkcid") in new
stack<BR> -- Goto
(macro-outbound-callerid,s,12)<BR> -- Executing
[s@macro-outbound-callerid:12] ExecIf("SIP/75002-b7705298",
"1|Set|CALLERID(all)=s2") in new stack<BR> -- Executing
[s@macro-outbound-callerid:13] ExecIf("SIP/75002-b7705298",
"0|Set|CALLERID(all)=") in new stack<BR> -- Executing
[s@macro-outbound-callerid:14] ExecIf("SIP/75002-b7705298",
"0|SetCallerPres|prohib_passed_screen") in new stack<BR>
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/75002-b7705298",
"0|AGI|fixlocalprefix") in new stack<BR> -- Executing
[s@macro-dialout-trunk:13] Set("SIP/75002-b7705298", "OUTNUM=015921256331")
in new stack<BR> -- Executing [s@macro-dialout-trunk:14]
Set("SIP/75002-b7705298", "custom=SIP/s2") in new
stack<BR> -- Executing [s@macro-dialout-trunk:15]
ExecIf("SIP/75002-b7705298", "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt")
in new stack<BR> -- Executing [s@macro-dialout-trunk:16]
Macro("SIP/75002-b7705298", "dialout-trunk-predial-hook|") in new
stack<BR> -- Executing
[s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/75002-b7705298", "")
in new stack<BR> -- Executing [s@macro-dialout-trunk:17]
GotoIf("SIP/75002-b7705298", "0?bypass|1") in new
stack<BR> -- Executing [s@macro-dialout-trunk:18]
GotoIf("SIP/75002-b7705298", "0?customtrunk") in new
stack<BR> -- Executing [s@macro-dialout-trunk:19]
Dial("SIP/75002-b7705298", "SIP/s2/015921256331|300|M(setmusic^none)Tt") in
new stack<BR>Audio is at 219.235.234.238 port 17136<BR>Adding codec 0x4
(ulaw) to SDP<BR>Adding non-codec 0x1 (telephone-event) to SDP<BR>Reliably
Transmitting (NAT) to <A
href="http://222.46.18.52:5060">222.46.18.52:5060</A>:<BR>INVITE <A
href="mailto:sip%3A015921256331@222.46.18.52">sip:015921256331@222.46.18.52</A>
SIP/2.0<BR>Via: SIP/2.0/UDP
219.235.234.238:5060;branch=z9hG4bK368b5ad8;rport<BR>From: "s2" <<A
href="mailto:sip%3AUnknown@222.46.18.52">sip:Unknown@222.46.18.52</A>>;tag=as75543a2d<BR>To:
<<A
href="mailto:sip%3A015921256331@222.46.18.52">sip:015921256331@222.46.18.52</A>><BR>Contact:
<<A
href="mailto:sip%3AUnknown@219.235.234.238">sip:Unknown@219.235.234.238</A>><BR>Call-ID:
<A
href="mailto:5cf71e106209cf65344e24031354fbda@222.46.18.52">5cf71e106209cf65344e24031354fbda@222.46.18.52</A><BR>CSeq:
102 INVITE<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Date: Fri, 26
Mar 2010 06:16:38 GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY<BR>Supported: replaces<BR>Content-Type:
application/sdp<BR>Content-Length: 244</DIV>
<P>v=0<BR>o=root 3145 3145 IN IP4 219.235.234.238<BR>s=session<BR>c=IN IP4
219.235.234.238<BR>t=0 0<BR>m=audio 17136 RTP/AVP 0 101<BR>a=rtpmap:0
PCMU/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101
0-16<BR>a=silenceSupp:off - - - -<BR>a=ptime:20<BR>a=sendrecv</P>
<P>---<BR> -- Called s2/015921256331<BR>gd-branch*CLI>
<BR><--- SIP read from <A
href="http://222.46.18.52:5060">222.46.18.52:5060</A> ---><BR>SIP/2.0 403
Forbidden<BR>Via: SIP/2.0/UDP
219.235.234.238:5060;branch=z9hG4bK368b5ad8;received=58.247.12.18;rport=11028<BR>From:
"s2" <<A
href="mailto:sip%3AUnknown@222.46.18.52">sip:Unknown@222.46.18.52</A>>;tag=as75543a2d<BR>To:
<<A
href="mailto:sip%3A015921256331@222.46.18.52">sip:015921256331@222.46.18.52</A>><BR>Contact:
<<A
href="http://sip:015921256331@222.46.18.52:5060">sip:015921256331@222.46.18.52:5060</A>><BR>Call-ID:
<A
href="mailto:5cf71e106209cf65344e24031354fbda@222.46.18.52">5cf71e106209cf65344e24031354fbda@222.46.18.52</A><BR>CSeq:
102 INVITE<BR>Max-Forwards: 70<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY<BR>Supported: timer<BR>Server: VOS2009 V2.1.1.8</P>
<P><BR><-------------><BR>--- (11 headers 0 lines) ---<BR>Transmitting
(NAT) to <A href="http://222.46.18.52:5060">222.46.18.52:5060</A>:<BR>ACK <A
href="mailto:sip%3A015921256331@222.46.18.52">sip:015921256331@222.46.18.52</A>
SIP/2.0<BR>Via: SIP/2.0/UDP
219.235.234.238:5060;branch=z9hG4bK368b5ad8;rport<BR>From: "s2" <<A
href="mailto:sip%3AUnknown@222.46.18.52">sip:Unknown@222.46.18.52</A>>;tag=as75543a2d<BR>To:
<<A
href="mailto:sip%3A015921256331@222.46.18.52">sip:015921256331@222.46.18.52</A>><BR>Contact:
<<A
href="mailto:sip%3AUnknown@219.235.234.238">sip:Unknown@219.235.234.238</A>><BR>Call-ID:
<A
href="mailto:5cf71e106209cf65344e24031354fbda@222.46.18.52">5cf71e106209cf65344e24031354fbda@222.46.18.52</A><BR>CSeq:
102 ACK<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Content-Length:
0</P>
<P><BR>---<BR> -- SIP/s2-088f72e8 is
circuit-busy<BR> == Everyone is busy/congested at this time
(1:0/1/0)<BR> -- Executing [s@macro-dialout-trunk:20]
Goto("SIP/75002-b7705298", "s-CONGESTION|1") in new
stack<BR> -- Goto
(macro-dialout-trunk,s-CONGESTION,1)<BR> -- Executing
[s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/75002-b7705298",
"1?noreport") in new stack<BR> -- Goto
(macro-dialout-trunk,s-CONGESTION,3)<BR> -- Executing
[s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/75002-b7705298", "TRUNK Dial
failed due to CONGESTION - failing through to other trunks") in new
stack<BR> -- Executing [015921256331@from-internal:6]
Macro("SIP/75002-b7705298", "outisbusy|") in new stack<BR>
-- Executing [s@macro-outisbusy:1] Playback("SIP/75002-b7705298",
"all-circuits-busy-now|noanswer") in new stack<BR> --
<SIP/75002-b7705298> Playing 'all-circuits-busy-now' (language
'en')<BR>Really destroying SIP dialog <A
href="mailto:'5cf71e106209cf65344e24031354fbda@222.46.18.52'">'5cf71e106209cf65344e24031354fbda@222.46.18.52'</A>
Method: INVITE<BR> -- Executing [s@macro-outisbusy:2]
Playback("SIP/75002-b7705298", "pls-try-call-later|noanswer") in new
stack<BR> -- <SIP/75002-b7705298> Playing
'pls-try-call-later' (language 'en')<BR> -- Executing
[s@macro-outisbusy:3] Macro("SIP/75002-b7705298", "hangupcall") in new
stack<BR> -- Executing [s@macro-hangupcall:1]
GotoIf("SIP/75002-b7705298", "1?skiprg") in new stack<BR>
-- Goto (macro-hangupcall,s,4)<BR> -- Executing
[s@macro-hangupcall:4] GotoIf("SIP/75002-b7705298", "1?skipblkvm") in new
stack<BR> -- Goto
(macro-hangupcall,s,7)<BR> -- Executing
[s@macro-hangupcall:7] GotoIf("SIP/75002-b7705298", "1?theend") in new
stack<BR> -- Goto
(macro-hangupcall,s,9)<BR> -- Executing
[s@macro-hangupcall:9] Hangup("SIP/75002-b7705298", "") in new
stack<BR> == Spawn extension (macro-hangupcall, s, 9) exited non-zero
on 'SIP/75002-b7705298' in macro 'hangupcall'<BR> == Spawn extension
(macro-hangupcall, s, 9) exited non-zero on 'SIP/75002-b7705298' in macro
'outisbusy'<BR> == Spawn extension (macro-hangupcall, s, 9) exited
non-zero on 'SIP/75002-b7705298'<BR> == End MixMonitor Recording
SIP/75002-b7705298</P></BLOCKQUOTE>
<DIV><BR clear=all><BR>-- <BR>Best Regards!</DIV>
<DIV><BR>Aaron Chen <BR></DIV>
<P>
<HR>
<P></P>--
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