[asterisk-users] Problem on Dialling-out
Malvin Rito
mrito at mail.altcladding.com.ph
Tue Jul 12 23:36:51 CDT 2011
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:639285010430 at 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 at 172.16.9.15>;tag=as2267fdcc
To: <sip:639285010430 at lasip1.cordiaip.net>
Contact: <sip:Unknown at 172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492 at 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:639285010430 at 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 at 172.16.9.15>;tag=as2267fdcc
To: <sip:639285010430 at lasip1.cordiaip.net>
Contact: <sip:Unknown at 172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492 at 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:639285010430 at 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 at 172.16.9.15>;tag=as2267fdcc
To: <sip:639285010430 at lasip1.cordiaip.net>
Contact: <sip:Unknown at 172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492 at 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:639285010430 at 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 at 172.16.9.15>;tag=as2267fdcc
To: <sip:639285010430 at lasip1.cordiaip.net>
Contact: <sip:Unknown at 172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492 at 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:639285010430 at 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 at 172.16.9.15>;tag=as2267fdcc
To: <sip:639285010430 at lasip1.cordiaip.net>
Contact: <sip:Unknown at 172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492 at 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:639285010430 at 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 at 172.16.9.15>;tag=as2267fdcc
To: <sip:639285010430 at lasip1.cordiaip.net>
Contact: <sip:Unknown at 172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492 at 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:639285010430 at 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 at 172.16.9.15>;tag=as2267fdcc
To: <sip:639285010430 at lasip1.cordiaip.net>
Contact: <sip:Unknown at 172.16.9.15>
Call-ID: 12d2279238e5851572c30cad11bb9492 at 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 at 172.16.9.15' in 32000 ms (Method: INVITE)
Scheduling destruction of SIP dialog
'12d2279238e5851572c30cad11bb9492 at 172.16.9.15' in 32000 ms (Method: INVITE)
Really destroying SIP dialog
'12d2279238e5851572c30cad11bb9492 at 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
> <mrito at mail.altcladding.com.ph <mailto:mrito at 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 at macro-dialout-trunk:20]
> NoOp("SIP/1001-00000014", "Dial failed for some reason with
> DIALSTATUS = CONGESTION and HANGUPCAUSE = 0") in new stack
> -- Executing [s at macro-dialout-trunk:21]
> Goto("SIP/1001-00000014", "s-CONGESTION,1") in new stack
> -- Goto (macro-dialout-trunk,s-CONGESTION,1)
> -- Executing [s-CONGESTION at macro-dialout-trunk:1]
> Set("SIP/1001-00000014", "RC=0") in new stack
> -- Executing [s-CONGESTION at macro-dialout-trunk:2]
> Goto("SIP/1001-00000014", "0,1") in new stack
> -- Goto (macro-dialout-trunk,0,1)
> -- Executing [0 at macro-dialout-trunk:1]
> Goto("SIP/1001-00000014", "continue,1") in new stack
> -- Goto (macro-dialout-trunk,continue,1)
> -- Executing [continue at macro-dialout-trunk:1]
> GotoIf("SIP/1001-00000014", "1?noreport") in new stack
> -- Goto (macro-dialout-trunk,continue,3)
> -- Executing [continue at 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 at macro-dialout-trunk:4]
> Set("SIP/1001-00000014", "CALLERID(number)=1001") in new stack
> -- Executing [639285010430 at from-internal:8]
> Macro("SIP/1001-00000014", "outisbusy,") in new stack
> -- Executing [s at macro-outisbusy:1]
> Progress("SIP/1001-00000014", "") in new stack
> -- Executing [s at 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 at 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 at macro-outisbusy:4] Macro("SIP/1001-00000014",
> "hangupcall") in new stack
> -- Executing [s at macro-hangupcall:1] GotoIf("SIP/1001-00000014",
> "1?skiprg") in new stack
> -- Goto (macro-hangupcall,s,4)
> -- Executing [s at macro-hangupcall:4] GotoIf("SIP/1001-00000014",
> "1?skipblkvm") in new stack
> -- Goto (macro-hangupcall,s,7)
> -- Executing [s at macro-hangupcall:7] GotoIf("SIP/1001-00000014",
> "1?theend") in new stack
> -- Goto (macro-hangupcall,s,9)
> -- Executing [s at 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 at from-internal:1] Macro("SIP/1001-00000014",
> "hangupcall") in new stack
> -- Executing [s at macro-hangupcall:1] GotoIf("SIP/1001-00000014",
> "1?skiprg") in new stack
> -- Goto (macro-hangupcall,s,4)
> -- Executing [s at macro-hangupcall:4] GotoIf("SIP/1001-00000014",
> "1?skipblkvm") in new stack
> -- Goto (macro-hangupcall,s,7)
> -- Executing [s at macro-hangupcall:7] GotoIf("SIP/1001-00000014",
> "1?theend") in new stack
> -- Goto (macro-hangupcall,s,9)
> -- Executing [s at 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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