Everyone is busy/congested at this time (1:0/0/1)

Asterisk Asterisk Support
1

I cannot figure out this error. I used to be able to manual dial and it worked. Couldn’t get autodial to work but now both don’t work giving the same error as in the subject. Currently trying to dial Norwegian numbers. I am not sure about the ‘context’. When I was successful in manul dial, I didn’t change the context.

Account entry:

[Norway] disallow=all allow=gsm allow=ulaw type=friend dtmfmode=rfc2833 context=trunkinbound qualify=yes insecure=very nat=yes host=194.6.238.84 username=carrier username secret=carrier password

Dialplan Entry:

exten => _00.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _00.,2,Dial(SIP/${EXTEN:2}@Norway,,tTo) exten => _00.,3,Hangup

Unsuccessful Dial log:

 == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [004747289862@default:3] Hangup("Local/8600052@default-00000005;1", "") in new stack
  == Spawn extension (default, 004747289862, 3) exited non-zero on 'Local/8600052@default-00000005;1'
    -- Executing [h@default:1] AGI("Local/8600052@default-00000005;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CHANUNAVAIL----------") in new stack
    -- <Local/8600052@default-00000005;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CHANUNAVAIL---------- completed, returning 0
  == Spawn extension (default, 8600052, 1) exited non-zero on 'Local/8600052@default-00000005;2'
    -- Executing [h@default:1] AGI("Local/8600052@default-00000005;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- <Local/8600052@default-00000005;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0[/code]

[b]Successful dial log[/b]

[code]Connected to Asterisk 1.8.23.0-1_centos7.go RPM by demian@goautodial.com currently running on CentOS-71-64-minimal (pid = 8993)
Verbosity is at least 3
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000038;2", "8600051,F") in new stack
    -- Executing [004797718208@default:1] AGI("Local/8600051@default-00000038;1", "agi://127.0.0.1:4577/call_log") in new stack
  == Manager 'sendcron' logged off from 127.0.0.1
    -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=NORWAY))
    -- <Local/8600051@default-00000038;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing [004797718208@default:2] Dial("Local/8600051@default-00000038;1", "SIP/97718208@Norway,,tTo") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/97718208@Norway
    -- SIP/Norway-000000dc is making progress passing it to Local/8600051@default-00000038;1
[May 29 09:20:49] NOTICE[12166]: chan_sip.c:25829 handle_request_register: Registration from '"10818" <sip:10818@XXX.XXX.XXX.XXX:5060>' failed for '212.83.171.95:5090' - Wrong password
    -- SIP/Norway-000000dc answered Local/8600051@default-00000038;1
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing [h@default:1] AGI("Local/8600051@default-00000038;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----20-----4") in new stack
  == Manager 'sendcron' logged off from 127.0.0.1
    -- <Local/8600051@default-00000038;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----20-----4 completed, returning 0
    -- Executing [h@default:2] Dial("Local/8600051@default-00000038;1", "SIP/@Norway,,tTo") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/@Norway
  == Spawn extension (default, h, 2) exited non-zero on 'Local/8600051@default-00000038;1'
  == Spawn extension (default, 004797718208, 2) exited non-zero on 'Local/8600051@default-00000038;1'
  == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000038;2'
    -- Executing [h@default:1] AGI("Local/8600051@default-00000038;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- <Local/8600051@default-00000038;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
    -- Executing [h@default:2] Dial("Local/8600051@default-00000038;2", "SIP/@Norway,,tTo") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/@Norway
  == Spawn extension (default, h, 2) exited non-zero on 'Local/8600051@default-00000038;2'[/code]

[b]SIP logs[/b]

[code]<--- SIP read from UDP:194.6.238.84:5060 --->
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK1a6c1cad;rport=5060
To: <sip:194.6.238.84>;tag=07218832
From: "asterisk" <sip:asterisk@serverip>;tag=as7e1e3b98
Call-ID: 25efe57329d994294c0d80f95f854b93@serverip:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '25efe57329d994294c0d80f95f854b93@serverip:5060' Method: OPTIONS
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
Reliably Transmitting (NAT) to 10.10.10.100:54396:
OPTIONS sip:8002@10.10.10.100:54396;rinstance=fbd4445fc8206add SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK30cacb9c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as54bfb17b
To: <sip:8002@10.10.10.100:54396;rinstance=fbd4445fc8206add>
Contact: <sip:asterisk@10.10.10.1:5060>
Call-ID: 293dfa2757b18ad76ddc89e13250f601@10.10.10.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Date: Wed, 10 Jun 2015 11:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 10.10.10.100:54396:
OPTIONS sip:8002@10.10.10.100:54396;rinstance=fbd4445fc8206add SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK30cacb9c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as54bfb17b
To: <sip:8002@10.10.10.100:54396;rinstance=fbd4445fc8206add>
Contact: <sip:asterisk@10.10.10.1:5060>
Call-ID: 293dfa2757b18ad76ddc89e13250f601@10.10.10.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Date: Wed, 10 Jun 2015 11:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (NAT) to 10.10.10.100:54396:
OPTIONS sip:8002@10.10.10.100:54396;rinstance=fbd4445fc8206add SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK30cacb9c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as54bfb17b
To: <sip:8002@10.10.10.100:54396;rinstance=fbd4445fc8206add>
Contact: <sip:asterisk@10.10.10.1:5060>
Call-ID: 293dfa2757b18ad76ddc89e13250f601@10.10.10.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Date: Wed, 10 Jun 2015 11:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (NAT) to 10.10.10.100:54396:
OPTIONS sip:8002@10.10.10.100:54396;rinstance=fbd4445fc8206add SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK30cacb9c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as54bfb17b
To: <sip:8002@10.10.10.100:54396;rinstance=fbd4445fc8206add>
Contact: <sip:asterisk@10.10.10.1:5060>
Call-ID: 293dfa2757b18ad76ddc89e13250f601@10.10.10.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Date: Wed, 10 Jun 2015 11:29:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.10.10.100:54396 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK30cacb9c;rport=5060
To: <sip:8002@10.10.10.100:54396;rinstance=fbd4445fc8206add>;tag=96128726
From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as54bfb17b
Call-ID: 293dfa2757b18ad76ddc89e13250f601@10.10.10.1:5060
CSeq: 102 OPTIONS
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
[Jun 10 13:29:22] NOTICE[1925]: chan_sip.c:21647 handle_response_peerpoke: Peer '8002' is now Lagged. (3058ms / 2000ms)
Really destroying SIP dialog '293dfa2757b18ad76ddc89e13250f601@10.10.10.1:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.10.10.100:54396:
OPTIONS sip:8002@10.10.10.100:54396;rinstance=fbd4445fc8206add SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK188ce788;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as77b96784
To: <sip:8002@10.10.10.100:54396;rinstance=fbd4445fc8206add>
Contact: <sip:asterisk@10.10.10.1:5060>
Call-ID: 25100f3f7c4ed74b279291331a6cf064@10.10.10.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Date: Wed, 10 Jun 2015 11:29:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.10.10.100:54396 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK188ce788;rport=5060
To: <sip:8002@10.10.10.100:54396;rinstance=fbd4445fc8206add>;tag=f79d9941
From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as77b96784
Call-ID: 25100f3f7c4ed74b279291331a6cf064@10.10.10.1:5060
CSeq: 102 OPTIONS
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Length: 0

Thanks in advance.

2

This seems a vicidial installation rather than a plain asterisk. Your logs are incomplete :frowning: please try with complete logs to the same number.

3

Hi,
The issue is with your campaign dial out prefix.

4

[quote=“mshoaib”]Hi,
The issue is with your campaign dial out prefix.[/quote]

How should it be set ? I just include 00 because when selecting the campaign in goautodial it asks you for the country code which is 47 in this case. In the logs, the number being dialed begins with 0047 which should be correct.

These are all the logs generated with sip debug on when dialing the number:

[code]Connected to Asterisk 1.8.23.0-1_centos7.go RPM by demian@goautodial.com currently running on CentOS-71-64-minimal (pid = 8993)
Verbosity is at least 3
== Manager ‘sendcron’ logged on from 127.0.0.1
== Manager ‘sendcron’ logged off from 127.0.0.1
CentOS-71-64-minimal*CLI> sip set debug on
SIP Debugging enabled
Reliably Transmitting (NAT) to 10.10.10.101:49919:
OPTIONS sip:8002@10.10.10.101:49919;rinstance=5092f0bc9849dbf0 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK18664d04;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.10.10.1;tag=as7ecfb012
To: sip:8002@10.10.10.101:49919;rinstance=5092f0bc9849dbf0
Contact: sip:asterisk@10.10.10.1:5060
Call-ID: 36fc13e454bebad421af8863241fd831@10.10.10.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Date: Thu, 11 Jun 2015 06:35:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.10.10.101:49919 —>
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK18664d04;rport=5060
To: sip:8002@10.10.10.101:49919;rinstance=5092f0bc9849dbf0;tag=0c68e44e
From: “asterisk” sip:asterisk@10.10.10.1;tag=as7ecfb012
Call-ID: 36fc13e454bebad421af8863241fd831@10.10.10.1:5060
CSeq: 102 OPTIONS
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘36fc13e454bebad421af8863241fd831@10.10.10.1:5060’ Method: OPTIONS
== Manager ‘sendcron’ logged on from 127.0.0.1
– Executing [8600051@default:1] MeetMe(“Local/8600051@default-0000001a;2”, “8600051,F”) in new stack
– Executing [004747289862@default:1] AGI(“Local/8600051@default-0000001a;1”, “agi://127.0.0.1:4577/call_log”) in new stack
== Manager ‘sendcron’ logged off from 127.0.0.1
– AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=NORWAY))
– <Local/8600051@default-0000001a;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
– Executing [004747289862@default:2] Dial(“Local/8600051@default-0000001a;1”, “SIP/4747289862@Norway,tTo”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 14760
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 194.6.238.84:5060:
INVITE sip:4747289862@194.6.238.84 SIP/2.0
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK54cbaa08
Max-Forwards: 70
From: “M6110835400000012076” sip:Startfirma@serverip;tag=as2d41fced
To: sip:4747289862@194.6.238.84
Contact: sip:Startfirma@serverip:5060
Call-ID: 194fe2de702aa0da239417dc65b9f358@serverip:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Date: Thu, 11 Jun 2015 06:35:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “M6110835400000012076” sip:Startfirma@serverip;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 304

v=0
o=root 1031877536 1031877536 IN IP4 serverip
s=Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
c=IN IP4 serverip
t=0 0
m=audio 14760 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

-- Called SIP/4747289862@Norway

<— SIP read from UDP:194.6.238.84:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK54cbaa08
To: sip:4747289862@194.6.238.84
From: "M6110835400000012076"sip:Startfirma@serverip;tag=as2d41fced
Call-ID: 194fe2de702aa0da239417dc65b9f358@serverip:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:194.6.238.84:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK54cbaa08
Record-Route: sip:194.6.238.84;lr
To: sip:4747289862@194.6.238.84
From: "M6110835400000012076"sip:Startfirma@serverip;tag=as2d41fced
Call-ID: 194fe2de702aa0da239417dc65b9f358@serverip:5060
CSeq: 102 INVITE
Server: Sippy
WWW-Authenticate: Digest realm=“194.6.238.84”,nonce=“f6c9426dff7cb36b982145fb8828a113461e”
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Transmitting (no NAT) to 194.6.238.84:5060:
ACK sip:4747289862@194.6.238.84 SIP/2.0
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK54cbaa08
Max-Forwards: 70
From: “M6110835400000012076” sip:Startfirma@serverip;tag=as2d41fced
To: sip:4747289862@194.6.238.84
Contact: sip:Startfirma@serverip:5060
Call-ID: 194fe2de702aa0da239417dc65b9f358@serverip:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Content-Length: 0

Audio is at 14760
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 194.6.238.84:5060:
INVITE sip:4747289862@194.6.238.84 SIP/2.0
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK5bb51f8b
Max-Forwards: 70
From: “M6110835400000012076” sip:Startfirma@serverip;tag=as2d41fced
To: sip:4747289862@194.6.238.84
Contact: sip:Startfirma@serverip:5060
Call-ID: 194fe2de702aa0da239417dc65b9f358@serverip:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Authorization: Digest username=“4721052555”, realm=“194.6.238.84”, algorithm=MD5, uri="sip:4747289862@194.6.238.84", nonce=“f6c9426dff7cb36b982145fb8828a113461e”, response=“46e72df56f094261903b26026212d754”
Date: Thu, 11 Jun 2015 06:35:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “M6110835400000012076” sip:Startfirma@serverip;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 304

v=0
o=root 1031877536 1031877537 IN IP4 serverip
s=Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
c=IN IP4 serverip
t=0 0
m=audio 14760 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:194.6.238.84:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK5bb51f8b
To: sip:4747289862@194.6.238.84
From: "M6110835400000012076"sip:Startfirma@serverip;tag=as2d41fced
Call-ID: 194fe2de702aa0da239417dc65b9f358@serverip:5060
CSeq: 103 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:194.6.238.84:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK5bb51f8b
Record-Route: sip:194.6.238.84;lr
To: sip:4747289862@194.6.238.84;tag=nvxg7ih3axkoz3dg.i
From: "M6110835400000012076"sip:Startfirma@serverip;tag=as2d41fced
Call-ID: 194fe2de702aa0da239417dc65b9f358@serverip:5060
CSeq: 103 INVITE
Server: Sippy
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Transmitting (no NAT) to 194.6.238.84:5060:
ACK sip:4747289862@194.6.238.84 SIP/2.0
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK5bb51f8b
Max-Forwards: 70
From: “M6110835400000012076” sip:Startfirma@serverip;tag=as2d41fced
To: sip:4747289862@194.6.238.84;tag=nvxg7ih3axkoz3dg.i
Contact: sip:Startfirma@serverip:5060
Call-ID: 194fe2de702aa0da239417dc65b9f358@serverip:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Content-Length: 0

Scheduling destruction of SIP dialog ‘194fe2de702aa0da239417dc65b9f358@serverip:5060’ in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [004747289862@default:3] Hangup(“Local/8600051@default-0000001a;1”, “”) in new stack
== Spawn extension (default, 004747289862, 3) exited non-zero on ‘Local/8600051@default-0000001a;1’
– Executing [h@default:1] AGI(“Local/8600051@default-0000001a;1”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----1-----CHANUNAVAIL----------”) in new stack
– <Local/8600051@default-0000001a;1>AGI Script agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----1-----CHANUNAVAIL---------- completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on ‘Local/8600051@default-0000001a;2’
– Executing [h@default:1] AGI(“Local/8600051@default-0000001a;2”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----0---------------”) in new stack
– <Local/8600051@default-0000001a;2>AGI Script agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----0--------------- completed, returning 0
Really destroying SIP dialog ‘194fe2de702aa0da239417dc65b9f358@serverip:5060’ Method: INVITE
[Jun 11 08:35:49] NOTICE[1925]: chan_sip.c:13719 sip_reregister: – Re-registration for 4721052555@194.6.238.84
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 194.6.238.84:5060:
REGISTER sip:194.6.238.84 SIP/2.0
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK3327f547;rport
Max-Forwards: 70
From: sip:4721052555@194.6.238.84;tag=as028d138c
To: sip:4721052555@194.6.238.84
Call-ID: 2950841a5822a8083d246d9c2bfdf477@[2a01:4f8:202:a::2]
CSeq: 407 REGISTER
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Authorization: Digest username=“4721052555”, realm=“194.6.238.84”, algorithm=MD5, uri=“sip:194.6.238.84”, nonce=“1434004264:10bc627673f9410aeaa144afacaf7b70”, response=“28701159b823e37b98f960b9f80ae8c6”
Expires: 120
Contact: sip:4721052555@serverip:5060
Content-Length: 0

<— SIP read from UDP:194.6.238.84:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK3327f547;rport=5060
Contact: sip:4721052555@78.133.113.146:5060;rinstance=0c06d57a40ecfaa5;expires=142
Contact: sip:4721052555@serverip:5060;expires=300
To: sip:4721052555@194.6.238.84;tag=27d0b644
From: sip:4721052555@194.6.238.84;tag=as028d138c
Call-ID: 2950841a5822a8083d246d9c2bfdf477@[2a01:4f8:202:a::2]
CSeq: 407 REGISTER
Date: Thu, 11 Jun 2015 06:35:49 GMT
PortaBilling: available-funds:9388.92 currency:NOK
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Scheduling destruction of SIP dialog ‘2950841a5822a8083d246d9c2bfdf477@[2a01:4f8:202:a::2]’ in 32000 ms (Method: REGISTER)
[Jun 11 08:35:49] NOTICE[1925]: chan_sip.c:21597 handle_response_register: Outbound Registration: Expiry for 194.6.238.84 is 300 sec (Scheduling reregistration in 285 s)
== Manager ‘sendcron’ logged on from 127.0.0.1
== Manager ‘sendcron’ logged on from 127.0.0.1
== Manager ‘sendcron’ logged off from 127.0.0.1
== Manager ‘sendcron’ logged off from 127.0.0.1
Reliably Transmitting (no NAT) to 194.6.238.84:5060:
OPTIONS sip:194.6.238.84 SIP/2.0
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK1c00c5f0
Max-Forwards: 70
From: “asterisk” sip:asterisk@serverip;tag=as59275440
To: sip:194.6.238.84
Contact: sip:asterisk@serverip:5060
Call-ID: 7a2771c45a6f851f4eb8f6dd28492f6b@serverip:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Date: Thu, 11 Jun 2015 06:36:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:194.6.238.84:5060 —>
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP serverip:5060;branch=z9hG4bK1c00c5f0
To: sip:194.6.238.84;tag=d1f19965
From: “asterisk” sip:asterisk@serverip;tag=as59275440
Call-ID: 7a2771c45a6f851f4eb8f6dd28492f6b@serverip:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘7a2771c45a6f851f4eb8f6dd28492f6b@serverip:5060’ Method: OPTIONS
== Manager ‘sendcron’ logged on from 127.0.0.1
== Manager ‘sendcron’ logged off from 127.0.0.1
Really destroying SIP dialog ‘2950841a5822a8083d246d9c2bfdf477@[2a01:4f8:202:a::2]’ Method: REGISTER
Reliably Transmitting (NAT) to 10.10.10.101:49919:
OPTIONS sip:8002@10.10.10.101:49919;rinstance=5092f0bc9849dbf0 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK69f5baec;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.10.10.1;tag=as391a4fee
To: sip:8002@10.10.10.101:49919;rinstance=5092f0bc9849dbf0
Contact: sip:asterisk@10.10.10.1:5060
Call-ID: 428b652c66c0086478bbf1da6ce02b8d@10.10.10.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos7.go RPM by demian@goautodial.com
Date: Thu, 11 Jun 2015 06:36:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.10.10.101:49919 —>
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK69f5baec;rport=5060
To: sip:8002@10.10.10.101:49919;rinstance=5092f0bc9849dbf0;tag=78e4402e
From: “asterisk” sip:asterisk@10.10.10.1;tag=as391a4fee
Call-ID: 428b652c66c0086478bbf1da6ce02b8d@10.10.10.1:5060
CSeq: 102 OPTIONS
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Length: 0[/code]

5

Hi,
How would you like to pass your calls to to your SIP ? 0047 or 47 ?

if 0047 your dial plan should be

exten => _900.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _900.,2,Dial(SIP/${EXTEN:1}@Norway,,tTo)
exten => _900.,3,Hangup

If 47

exten => _900.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _900.,2,Dial(SIP/${EXTEN:3}@Norway,,tTo)
exten => _900.,3,Hangup

Do set your Dialout Prefix to 9 in goautodial campaign!

6

[quote=“mshoaib”]Hi,
How would you like to pass your calls to to your SIP ? 0047 or 47 ?

if 0047 your dial plan should be

exten => _900.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _900.,2,Dial(SIP/${EXTEN:1}@Norway,,tTo)
exten => _900.,3,Hangup

If 47

exten => _900.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _900.,2,Dial(SIP/${EXTEN:3}@Norway,,tTo)
exten => _900.,3,Hangup

Do set your Dialout Prefix to 9 in goautodial campaign![/quote]

Where do I set the prefix 9 in the campaign ?

7

by going into campaign tab & editing the campaign scroll down you’ll see out prefix or dial prefix there you have to put set the prefix.

Regards,
Mohammad