<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
Sorry I do not understand it, here is result after:<br>
<br>
Audio is at 172.16.9.15 port 15022<br>
Adding codec 0x4 (ulaw) to SDP<br>
Adding codec 0x8 (alaw) to SDP<br>
Adding non-codec 0x1 (telephone-event) to SDP<br>
Reliably Transmitting (no NAT) to 64.211.94.211:5060:<br>
INVITE <a class="moz-txt-link-freetext" href="sip:639285010430@lasip1.cordiaip.net">sip:639285010430@lasip1.cordiaip.net</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport<br>
Max-Forwards: 70<br>
From: "Cordia" <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a>;tag=as2267fdcc<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:639285010430@lasip1.cordiaip.net"><sip:639285010430@lasip1.cordiaip.net></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:12d2279238e5851572c30cad11bb9492@172.16.9.15">12d2279238e5851572c30cad11bb9492@172.16.9.15</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 1.6.2.7<br>
Date: Wed, 13 Jul 2011 04:07:34 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 281<br>
<br>
v=0<br>
o=root 45158429 45158429 IN IP4 172.16.9.15<br>
s=Asterisk PBX 1.6.2.7<br>
c=IN IP4 172.16.9.15<br>
t=0 0<br>
m=audio 15022 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/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<br>
<br>
---<br>
Retransmitting #1 (no NAT) to 64.211.94.211:5060:<br>
INVITE <a class="moz-txt-link-freetext" href="sip:639285010430@lasip1.cordiaip.net">sip:639285010430@lasip1.cordiaip.net</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport<br>
Max-Forwards: 70<br>
From: "Cordia" <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a>;tag=as2267fdcc<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:639285010430@lasip1.cordiaip.net"><sip:639285010430@lasip1.cordiaip.net></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:12d2279238e5851572c30cad11bb9492@172.16.9.15">12d2279238e5851572c30cad11bb9492@172.16.9.15</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 1.6.2.7<br>
Date: Wed, 13 Jul 2011 04:07:34 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 281<br>
<br>
v=0<br>
o=root 45158429 45158429 IN IP4 172.16.9.15<br>
s=Asterisk PBX 1.6.2.7<br>
c=IN IP4 172.16.9.15<br>
t=0 0<br>
m=audio 15022 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/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<br>
<br>
---<br>
Retransmitting #2 (no NAT) to 64.211.94.211:5060:<br>
INVITE <a class="moz-txt-link-freetext" href="sip:639285010430@lasip1.cordiaip.net">sip:639285010430@lasip1.cordiaip.net</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport<br>
Max-Forwards: 70<br>
From: "Cordia" <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a>;tag=as2267fdcc<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:639285010430@lasip1.cordiaip.net"><sip:639285010430@lasip1.cordiaip.net></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:12d2279238e5851572c30cad11bb9492@172.16.9.15">12d2279238e5851572c30cad11bb9492@172.16.9.15</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 1.6.2.7<br>
Date: Wed, 13 Jul 2011 04:07:34 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 281<br>
<br>
v=0<br>
o=root 45158429 45158429 IN IP4 172.16.9.15<br>
s=Asterisk PBX 1.6.2.7<br>
c=IN IP4 172.16.9.15<br>
t=0 0<br>
m=audio 15022 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/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<br>
<br>
---<br>
Retransmitting #3 (no NAT) to 64.211.94.211:5060:<br>
INVITE <a class="moz-txt-link-freetext" href="sip:639285010430@lasip1.cordiaip.net">sip:639285010430@lasip1.cordiaip.net</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport<br>
Max-Forwards: 70<br>
From: "Cordia" <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a>;tag=as2267fdcc<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:639285010430@lasip1.cordiaip.net"><sip:639285010430@lasip1.cordiaip.net></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:12d2279238e5851572c30cad11bb9492@172.16.9.15">12d2279238e5851572c30cad11bb9492@172.16.9.15</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 1.6.2.7<br>
Date: Wed, 13 Jul 2011 04:07:34 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 281<br>
<br>
v=0<br>
o=root 45158429 45158429 IN IP4 172.16.9.15<br>
s=Asterisk PBX 1.6.2.7<br>
c=IN IP4 172.16.9.15<br>
t=0 0<br>
m=audio 15022 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/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<br>
<br>
---<br>
Retransmitting #4 (no NAT) to 64.211.94.211:5060:<br>
INVITE <a class="moz-txt-link-freetext" href="sip:639285010430@lasip1.cordiaip.net">sip:639285010430@lasip1.cordiaip.net</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport<br>
Max-Forwards: 70<br>
From: "Cordia" <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a>;tag=as2267fdcc<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:639285010430@lasip1.cordiaip.net"><sip:639285010430@lasip1.cordiaip.net></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:12d2279238e5851572c30cad11bb9492@172.16.9.15">12d2279238e5851572c30cad11bb9492@172.16.9.15</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 1.6.2.7<br>
Date: Wed, 13 Jul 2011 04:07:34 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 281<br>
<br>
v=0<br>
o=root 45158429 45158429 IN IP4 172.16.9.15<br>
s=Asterisk PBX 1.6.2.7<br>
c=IN IP4 172.16.9.15<br>
t=0 0<br>
m=audio 15022 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/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<br>
<br>
---<br>
Retransmitting #5 (no NAT) to 64.211.94.211:5060:<br>
INVITE <a class="moz-txt-link-freetext" href="sip:639285010430@lasip1.cordiaip.net">sip:639285010430@lasip1.cordiaip.net</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport<br>
Max-Forwards: 70<br>
From: "Cordia" <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a>;tag=as2267fdcc<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:639285010430@lasip1.cordiaip.net"><sip:639285010430@lasip1.cordiaip.net></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:12d2279238e5851572c30cad11bb9492@172.16.9.15">12d2279238e5851572c30cad11bb9492@172.16.9.15</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 1.6.2.7<br>
Date: Wed, 13 Jul 2011 04:07:34 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 281<br>
<br>
v=0<br>
o=root 45158429 45158429 IN IP4 172.16.9.15<br>
s=Asterisk PBX 1.6.2.7<br>
c=IN IP4 172.16.9.15<br>
t=0 0<br>
m=audio 15022 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/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<br>
<br>
---<br>
Retransmitting #6 (no NAT) to 64.211.94.211:5060:<br>
INVITE <a class="moz-txt-link-freetext" href="sip:639285010430@lasip1.cordiaip.net">sip:639285010430@lasip1.cordiaip.net</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport<br>
Max-Forwards: 70<br>
From: "Cordia" <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a>;tag=as2267fdcc<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:639285010430@lasip1.cordiaip.net"><sip:639285010430@lasip1.cordiaip.net></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:Unknown@172.16.9.15"><sip:Unknown@172.16.9.15></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:12d2279238e5851572c30cad11bb9492@172.16.9.15">12d2279238e5851572c30cad11bb9492@172.16.9.15</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 1.6.2.7<br>
Date: Wed, 13 Jul 2011 04:07:34 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 281<br>
<br>
v=0<br>
o=root 45158429 45158429 IN IP4 172.16.9.15<br>
s=Asterisk PBX 1.6.2.7<br>
c=IN IP4 172.16.9.15<br>
t=0 0<br>
m=audio 15022 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/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<br>
<br>
---<br>
Scheduling destruction of SIP dialog
'<a class="moz-txt-link-abbreviated" href="mailto:12d2279238e5851572c30cad11bb9492@172.16.9.15">12d2279238e5851572c30cad11bb9492@172.16.9.15</a>' in 32000 ms (Method:
INVITE)<br>
Scheduling destruction of SIP dialog
'<a class="moz-txt-link-abbreviated" href="mailto:12d2279238e5851572c30cad11bb9492@172.16.9.15">12d2279238e5851572c30cad11bb9492@172.16.9.15</a>' in 32000 ms (Method:
INVITE)<br>
Really destroying SIP dialog
'<a class="moz-txt-link-abbreviated" href="mailto:12d2279238e5851572c30cad11bb9492@172.16.9.15">12d2279238e5851572c30cad11bb9492@172.16.9.15</a>' Method: INVITE<br>
localhost*CLI><br>
<br>
<br>
<br>
On 7/13/2011 12:30 PM, Bruce B wrote:
<blockquote
cite="mid:CAJyE_uVO2fcHdZ18hF4zjE4EjoNpXy_6M0qRDMwamSwBv-__Tw@mail.gmail.com"
type="cite">Your trunk shows busy:
<div><br>
</div>
<div><span class="Apple-style-span" style="font-family:
arial,sans-serif; font-size: 13px; background-color: rgb(255,
255, 255);"><b><i> -- Called CordiaVoIP/639285010430<br>
-- SIP/CordiaVoIP-00000015 <font
class="Apple-style-span" color="#ff0000">is circuit-busy</font><br>
<font class="Apple-style-span" color="#ff0000"> ==
Everyone is busy/congested at this time (1:0/1/0)</font></i></b></span><br>
<br>
</div>
<div>Try this in the CLI (asterisk -rvvvvvvvvvvvv):</div>
<div><b>core set verbose 0</b></div>
<div><b>sip set debug peer CordiaVoIP</b></div>
<div><br>
</div>
<div>And then make a call and read why the SIP trunk is failing. </div>
<div><br>
</div>
<div>-Bruce</div>
<div><br>
<div class="gmail_quote"><br>
</div>
<div class="gmail_quote">
On Wed, Jul 13, 2011 at 12:23 AM, Malvin Rito <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:mrito@mail.altcladding.com.ph">mrito@mail.altcladding.com.ph</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
Hi List,<br>
<br>
I have a Asterisk + FreePbx Server setup with around 10 SIP
extensions and 1 VoIP trunk (CordiaVoIP), when we dial-out
to any number call is being dropped with the following
message on asterisk log:<br>
<br>
== Using SIP RTP TOS bits 184<br>
== Using SIP RTP CoS mark 5<br>
-- Called CordiaVoIP/639285010430<br>
-- SIP/CordiaVoIP-00000015 is circuit-busy<br>
== Everyone is busy/congested at this time (1:0/1/0)<br>
-- Executing [s@macro-dialout-trunk:20]
NoOp("SIP/1001-00000014", "Dial failed for some reason with
DIALSTATUS = CONGESTION and HANGUPCAUSE = 0") in new stack<br>
-- Executing [s@macro-dialout-trunk:21]
Goto("SIP/1001-00000014", "s-CONGESTION,1") in new stack<br>
-- Goto (macro-dialout-trunk,s-CONGESTION,1)<br>
-- Executing [s-CONGESTION@macro-dialout-trunk:1]
Set("SIP/1001-00000014", "RC=0") in new stack<br>
-- Executing [s-CONGESTION@macro-dialout-trunk:2]
Goto("SIP/1001-00000014", "0,1") in new stack<br>
-- Goto (macro-dialout-trunk,0,1)<br>
-- Executing [0@macro-dialout-trunk:1]
Goto("SIP/1001-00000014", "continue,1") in new stack<br>
-- Goto (macro-dialout-trunk,continue,1)<br>
-- Executing [continue@macro-dialout-trunk:1]
GotoIf("SIP/1001-00000014", "1?noreport") in new stack<br>
-- Goto (macro-dialout-trunk,continue,3)<br>
-- Executing [continue@macro-dialout-trunk:3]
NoOp("SIP/1001-00000014", "TRUNK Dial failed due to
CONGESTION HANGUPCAUSE: 0 - failing through to other
trunks") in new stack<br>
-- Executing [continue@macro-dialout-trunk:4]
Set("SIP/1001-00000014", "CALLERID(number)=1001") in new
stack<br>
-- Executing [639285010430@from-internal:8]
Macro("SIP/1001-00000014", "outisbusy,") in new stack<br>
-- Executing [s@macro-outisbusy:1]
Progress("SIP/1001-00000014", "") in new stack<br>
-- Executing [s@macro-outisbusy:2]
Playback("SIP/1001-00000014", "all-circuits-busy-now,noanswer")
in new stack<br>
-- <SIP/1001-00000014> Playing
'all-circuits-busy-now.gsm' (language 'en')<br>
-- Executing [s@macro-outisbusy:3]
Playback("SIP/1001-00000014", "pls-try-call-later,noanswer")
in new stack<br>
-- <SIP/1001-00000014> Playing
'pls-try-call-later.gsm' (language 'en')<br>
-- Executing [s@macro-outisbusy:4]
Macro("SIP/1001-00000014", "hangupcall") in new stack<br>
-- Executing [s@macro-hangupcall:1]
GotoIf("SIP/1001-00000014", "1?skiprg") in new stack<br>
-- Goto (macro-hangupcall,s,4)<br>
-- Executing [s@macro-hangupcall:4]
GotoIf("SIP/1001-00000014", "1?skipblkvm") in new stack<br>
-- Goto (macro-hangupcall,s,7)<br>
-- Executing [s@macro-hangupcall:7]
GotoIf("SIP/1001-00000014", "1?theend") in new stack<br>
-- Goto (macro-hangupcall,s,9)<br>
-- Executing [s@macro-hangupcall:9]
Hangup("SIP/1001-00000014", "") in new stack<br>
== Spawn extension (macro-hangupcall, s, 9) exited non-zero
on 'SIP/1001-00000014' in macro 'hangupcall'<br>
== Spawn extension (macro-outisbusy, s, 4) exited non-zero
on 'SIP/1001-00000014' in macro 'outisbusy'<br>
== Spawn extension (from-internal, 639285010430, 8) exited
non-zero on 'SIP/1001-00000014'<br>
-- Executing [h@from-internal:1]
Macro("SIP/1001-00000014", "hangupcall") in new stack<br>
-- Executing [s@macro-hangupcall:1]
GotoIf("SIP/1001-00000014", "1?skiprg") in new stack<br>
-- Goto (macro-hangupcall,s,4)<br>
-- Executing [s@macro-hangupcall:4]
GotoIf("SIP/1001-00000014", "1?skipblkvm") in new stack<br>
-- Goto (macro-hangupcall,s,7)<br>
-- Executing [s@macro-hangupcall:7]
GotoIf("SIP/1001-00000014", "1?theend") in new stack<br>
-- Goto (macro-hangupcall,s,9)<br>
-- Executing [s@macro-hangupcall:9]
Hangup("SIP/1001-00000014", "") in new stack<br>
== Spawn extension (macro-hangupcall, s, 9) exited non-zero
on 'SIP/1001-00000014' in macro 'hangupcall'<br>
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/1001-00000014'<br>
localhost*CLI><br>
<br>
<br>
Can someone assist me please. Thanks in advance.<br>
<br>
Regards,<br>
Malvin<br>
<br>
<br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a
moz-do-not-send="true" href="http://www.api-digital.com"
target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar
every Thurs:<br>
<a moz-do-not-send="true"
href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a moz-do-not-send="true"
href="http://lists.digium.com/mailman/listinfo/asterisk-users"
target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote>
</div>
<br>
</div>
</blockquote>
</body>
</html>