[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
>
>
>
>     --
>     _____________________________________________________________________
>     -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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