Sorry I do not understand it, here is result after:
Audio is at 172.16.9.15 port 15022
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.211.94.211:5060:
INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport
Max-Forwards: 70
From: "Cordia" <sip:Unknown@172.16.9.15>;tag=as2267fdcc
To: <sip:639285010...@lasip1.cordiaip.net>
Contact: <sip:Unknown@172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Date: Wed, 13 Jul 2011 04:07:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 45158429 45158429 IN IP4 172.16.9.15
s=Asterisk PBX 1.6.2.7
c=IN IP4 172.16.9.15
t=0 0
m=audio 15022 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #1 (no NAT) to 64.211.94.211:5060:
INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport
Max-Forwards: 70
From: "Cordia" <sip:Unknown@172.16.9.15>;tag=as2267fdcc
To: <sip:639285010...@lasip1.cordiaip.net>
Contact: <sip:Unknown@172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Date: Wed, 13 Jul 2011 04:07:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 45158429 45158429 IN IP4 172.16.9.15
s=Asterisk PBX 1.6.2.7
c=IN IP4 172.16.9.15
t=0 0
m=audio 15022 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #2 (no NAT) to 64.211.94.211:5060:
INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport
Max-Forwards: 70
From: "Cordia" <sip:Unknown@172.16.9.15>;tag=as2267fdcc
To: <sip:639285010...@lasip1.cordiaip.net>
Contact: <sip:Unknown@172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Date: Wed, 13 Jul 2011 04:07:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 45158429 45158429 IN IP4 172.16.9.15
s=Asterisk PBX 1.6.2.7
c=IN IP4 172.16.9.15
t=0 0
m=audio 15022 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #3 (no NAT) to 64.211.94.211:5060:
INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport
Max-Forwards: 70
From: "Cordia" <sip:Unknown@172.16.9.15>;tag=as2267fdcc
To: <sip:639285010...@lasip1.cordiaip.net>
Contact: <sip:Unknown@172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Date: Wed, 13 Jul 2011 04:07:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 45158429 45158429 IN IP4 172.16.9.15
s=Asterisk PBX 1.6.2.7
c=IN IP4 172.16.9.15
t=0 0
m=audio 15022 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #4 (no NAT) to 64.211.94.211:5060:
INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport
Max-Forwards: 70
From: "Cordia" <sip:Unknown@172.16.9.15>;tag=as2267fdcc
To: <sip:639285010...@lasip1.cordiaip.net>
Contact: <sip:Unknown@172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Date: Wed, 13 Jul 2011 04:07:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 45158429 45158429 IN IP4 172.16.9.15
s=Asterisk PBX 1.6.2.7
c=IN IP4 172.16.9.15
t=0 0
m=audio 15022 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #5 (no NAT) to 64.211.94.211:5060:
INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport
Max-Forwards: 70
From: "Cordia" <sip:Unknown@172.16.9.15>;tag=as2267fdcc
To: <sip:639285010...@lasip1.cordiaip.net>
Contact: <sip:Unknown@172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Date: Wed, 13 Jul 2011 04:07:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 45158429 45158429 IN IP4 172.16.9.15
s=Asterisk PBX 1.6.2.7
c=IN IP4 172.16.9.15
t=0 0
m=audio 15022 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #6 (no NAT) to 64.211.94.211:5060:
INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0
Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport
Max-Forwards: 70
From: "Cordia" <sip:Unknown@172.16.9.15>;tag=as2267fdcc
To: <sip:639285010...@lasip1.cordiaip.net>
Contact: <sip:Unknown@172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7
Date: Wed, 13 Jul 2011 04:07:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 45158429 45158429 IN IP4 172.16.9.15
s=Asterisk PBX 1.6.2.7
c=IN IP4 172.16.9.15
t=0 0
m=audio 15022 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Scheduling destruction of SIP dialog
'12d2279238e5851572c30cad11bb9492@172.16.9.15' in 32000 ms (Method: INVITE)
Scheduling destruction of SIP dialog
'12d2279238e5851572c30cad11bb9492@172.16.9.15' in 32000 ms (Method: INVITE)
Really destroying SIP dialog
'12d2279238e5851572c30cad11bb9492@172.16.9.15' Method: INVITE
localhost*CLI>
On 7/13/2011 12:30 PM, Bruce B wrote:
Your trunk shows busy:
*/ -- Called CordiaVoIP/639285010430
-- SIP/CordiaVoIP-00000015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)/*
Try this in the CLI (asterisk -rvvvvvvvvvvvv):
*core set verbose 0*
*sip set debug peer CordiaVoIP*
And then make a call and read why the SIP trunk is failing.
-Bruce
On Wed, Jul 13, 2011 at 12:23 AM, Malvin Rito
<mr...@mail.altcladding.com.ph <mailto:mr...@mail.altcladding.com.ph>>
wrote:
Hi List,
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:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called CordiaVoIP/639285010430
-- SIP/CordiaVoIP-00000015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20]
NoOp("SIP/1001-00000014", "Dial failed for some reason with
DIALSTATUS = CONGESTION and HANGUPCAUSE = 0") in new stack
-- Executing [s@macro-dialout-trunk:21]
Goto("SIP/1001-00000014", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1]
Set("SIP/1001-00000014", "RC=0") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2]
Goto("SIP/1001-00000014", "0,1") in new stack
-- Goto (macro-dialout-trunk,0,1)
-- Executing [0@macro-dialout-trunk:1]
Goto("SIP/1001-00000014", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1]
GotoIf("SIP/1001-00000014", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- 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
-- Executing [continue@macro-dialout-trunk:4]
Set("SIP/1001-00000014", "CALLERID(number)=1001") in new stack
-- Executing [639285010430@from-internal:8]
Macro("SIP/1001-00000014", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1]
Progress("SIP/1001-00000014", "") in new stack
-- Executing [s@macro-outisbusy:2]
Playback("SIP/1001-00000014", "all-circuits-busy-now,noanswer") in
new stack
-- <SIP/1001-00000014> Playing 'all-circuits-busy-now.gsm'
(language 'en')
-- Executing [s@macro-outisbusy:3]
Playback("SIP/1001-00000014", "pls-try-call-later,noanswer") in
new stack
-- <SIP/1001-00000014> Playing 'pls-try-call-later.gsm'
(language 'en')
-- Executing [s@macro-outisbusy:4] Macro("SIP/1001-00000014",
"hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/1001-00000014",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/1001-00000014",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/1001-00000014",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/1001-00000014",
"") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/1001-00000014' in macro 'hangupcall'
== Spawn extension (macro-outisbusy, s, 4) exited non-zero on
'SIP/1001-00000014' in macro 'outisbusy'
== Spawn extension (from-internal, 639285010430, 8) exited
non-zero on 'SIP/1001-00000014'
-- Executing [h@from-internal:1] Macro("SIP/1001-00000014",
"hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/1001-00000014",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/1001-00000014",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/1001-00000014",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/1001-00000014",
"") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/1001-00000014' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/1001-00000014'
localhost*CLI>
Can someone assist me please. Thanks in advance.
Regards,
Malvin
--
_____________________________________________________________________
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--
_____________________________________________________________________
-- 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
