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Intermittent "403 FORBIDDEN" during outbound auto dialling

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Intermittent "403 FORBIDDEN" during outbound auto dialling

Postby wjohnson133 » Wed Jul 19, 2017 4:43 pm

I am intermittently getting "403 forbidden" and "Everyone is busy/congested" during a blended autodial campaign. Sometimes it works and sometimes it doesn't. Here is the asterisk log for two separate occasions where it has happened in the hopes that somebody will be able to decode why it is happening and what I could do to troubleshoot it:

DEBUG1856- manager.c: Running action 'Command'
DEBUG5957- manager.c: Running action 'Login'
VERBOSE5957- manager.c: == Manager 'sendcron' logged on from 127.0.0.1
DEBUG5957- manager.c: Running action 'Originate'
DEBUG1864- manager.c: Examining event:
Event: Newchannel
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002bf;1
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten: 70571819351438XXXXXXX
Context: default
Uniqueid: 1500497009.1981


DEBUG1864- manager.c: Examining event:
Event: Newchannel
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002bf;2
ChannelState: 4
ChannelStateDesc: Ring
CallerIDNum:
CallerIDName:
AccountCode:
Exten: 70571819351438XXXXXXX
Context: default
Uniqueid: 1500497009.1982


DEBUG1864- manager.c: Examining event:
Event: NewAccountCode
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002bf;1
Uniqueid: 1500497009.1981
AccountCode:
OldAccountCode:


DEBUG1864- manager.c: Examining event:
Event: NewCallerid
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002bf;1
CallerIDNum: 4502391367
CallerIDName: V7191643290000000537
Uniqueid: 1500497009.1981
CID-CallingPres: 0 (Presentation Allowed, Not Screened)


DEBUG1864- manager.c: Examining event:
Event: LocalBridge
Privilege: call,all
Channel1: Local/70571819351438XXXXXXX default-000002bf;1
Channel2: Local/70571819351438XXXXXXX default-000002bf;2
Uniqueid1: 1500497009.1981
Uniqueid2: 1500497009.1982
Context: default
Exten: 70571819351438XXXXXXX
LocalOptimization: Yes


DEBUG5958-C-00000314- pbx.c: Launching 'AGI'
VERBOSE5958-C-00000314- pbx.c: -- Executing 70571819351438XXXXXXX default:1- AGI("Local/70571819351438XXXXXXX default-000002bf;2", "agi://127.0.0.1:4577/call_log") in new stack
DEBUG5958-C-00000314- netsock2.c: Splitting '127.0.0.1:4577' into...
DEBUG5958-C-00000314- netsock2.c: ...host '127.0.0.1' and port '4577'.
DEBUG5958-C-00000314- res_agi.c: Wow, connected!
VERBOSE5958-C-00000314- res_agi.c: -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=75432148))
VERBOSE5958-C-00000314- res_agi.c: -- <Local/70571819351438XXXXXXX default-000002bf;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
DEBUG5958-C-00000314- pbx.c: Result of 'EXTEN' is '70571819351438XXXXXXX'
DEBUG5958-C-00000314- pbx.c: Launching 'Dial'
VERBOSE5958-C-00000314- pbx.c: -- Executing 70571819351438XXXXXXX default:2- Dial("Local/70571819351438XXXXXXX default-000002bf;2", "SIP/1438XXXXXXX MyOwnTelco,,tTo") in new stack
DEBUG5958-C-00000314- chan_sip.c: Asked to create a SIP channel with formats: (slin)
DEBUG5958-C-00000314- chan_sip.c: Allocating new SIP dialog for 45546da72bd03cac4c884a9a62f5a5e5 127.0.0.2:5060 - INVITE (No RTP)
DEBUG5958-C-00000314- rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f01d4018da8'
DEBUG5958-C-00000314- res_rtp_asterisk.c: Allocated port 11898 for RTP instance '0x7f01d4018da8'
DEBUG5958-C-00000314- res_rtp_asterisk.c: Creating ICE session 0.0.0.0:11898 (11898) for RTP instance '0x7f01d4018da8'
DEBUG5958-C-00000314- netsock2.c: Splitting '192.168.0.107' into...
DEBUG5958-C-00000314- netsock2.c: ...host '192.168.0.107' and port ''.
DEBUG5958-C-00000314- rtp_engine.c: RTP instance '0x7f01d4018da8' is setup and ready to go
DEBUG5958-C-00000314- res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f01d4018da8'
VERBOSE5958-C-00000314- netsock2.c: == Using SIP RTP CoS mark 5
DEBUG5958-C-00000314- chan_sip.c: Setting NAT on RTP to On
DEBUG5958-C-00000314- chan_sip.c: OBPROXY: Not applying OBproxy to this call
DEBUG5958-C-00000314- acl.c: For destination '205.204.72.125', our source address is '192.168.0.107'.
DEBUG5958-C-00000314- chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.0.107:5060
DEBUG5958-C-00000314- chan_sip.c: Setting NAT on RTP to On
DEBUG5958-C-00000314- chan_sip.c: SIP call-id changed from '45546da72bd03cac4c884a9a62f5a5e5 127.0.0.2:5060' to '4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060'
DEBUG1864- manager.c: Examining event:
Event: Newchannel
Privilege: call,all
Channel: SIP/MyOwnTelco-00000220
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: default
Uniqueid: 1500497009.1983


DEBUG5958-C-00000314- chan_sip.c: *** Our native formats are (ulaw)
DEBUG5958-C-00000314- chan_sip.c: *** Joint capabilities are (nothing)
DEBUG5958-C-00000314- chan_sip.c: *** Our capabilities are (gsm|ulaw)
DEBUG5958-C-00000314- chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
DEBUG5958-C-00000314- chan_sip.c: *** Our preferred formats from the incoming channel are (slin)
DEBUG5958-C-00000314- chan_sip.c: This channel will not be able to handle video.
DEBUG1864- manager.c: Examining event:
Event: ChannelUpdate
Privilege: system,all
Channel: SIP/MyOwnTelco-00000220
Uniqueid: 1500497009.1983
Channeltype: SIP
SIPcallid: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
SIPfullcontact:


DEBUG1864- manager.c: Examining event:
Event: ChannelUpdate
Privilege: system,all
Channel: SIP/MyOwnTelco-00000220
Channeltype: SIP
SIPcallid: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
SIPfullcontact:
Peername: MyOwnTelco


DEBUG5958-C-00000314- channel_internal_api.c: Channel Call ID changing from C-00000314- to C-00000314-
DEBUG5958-C-00000314- rtp_engine.c: Can't find native functions for channel 'Local/70571819351438XXXXXXX default-000002bf;2'
DEBUG5958-C-00000314- channel.c: Inheriting variable CAMPCUST from Local/70571819351438XXXXXXX default-000002bf;2 to SIP/MyOwnTelco-00000220.
DEBUG5958-C-00000314- chan_sip.c: Outgoing Call for 1438XXXXXXX
DEBUG5958-C-00000314- chan_sip.c: Updating call counter for outgoing call
DEBUG1864- manager.c: Examining event:
Event: NewCallerid
Privilege: call,all
Channel: SIP/MyOwnTelco-00000220
CallerIDNum: 4502391367
CallerIDName: V7191643290000000537
Uniqueid: 1500497009.1983
CID-CallingPres: 0 (Presentation Allowed, Not Screened)


DEBUG5958-C-00000314- chan_sip.c: ** Our capability: (gsm|ulaw) Video flag: False Text flag: False
DEBUG5958-C-00000314- chan_sip.c: ** Our prefcodec: (slin)
VERBOSE5958-C-00000314- chan_sip.c: Audio is at 11898
VERBOSE5958-C-00000314- chan_sip.c: Adding codec 100003 (ulaw) to SDP
VERBOSE5958-C-00000314- chan_sip.c: Adding codec 100002 (gsm) to SDP
VERBOSE5958-C-00000314- chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
DEBUG5958-C-00000314- chan_sip.c: -- Done with adding codecs to SDP
DEBUG5958-C-00000314- chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw)
DEBUG5958-C-00000314- chan_sip.c: Initializing initreq for method INVITE - callid 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
DEBUG5958-C-00000314- chan_sip.c: Header 0 49-: INVITE sip:1438XXXXXXX sip.myowntelco.net SIP/2.0
DEBUG5958-C-00000314- chan_sip.c: Header 1 64-: Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK1a7b4bf4;rport
DEBUG5958-C-00000314- chan_sip.c: Header 2 16-: Max-Forwards: 70
DEBUG5958-C-00000314- chan_sip.c: Header 3 74-: From: "V7191643290000000537" <sip:4502391367 192.168.0.107>;tag=as680e1bb5
DEBUG5958-C-00000314- chan_sip.c: Header 4 40-: To: <sip:1438XXXXXXX sip.myowntelco.net>
DEBUG5958-C-00000314- chan_sip.c: Header 5 44-: Contact: <sip:4502391367 192.168.0.107:5060>
DEBUG5958-C-00000314- chan_sip.c: Header 6 60-: Call-ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
DEBUG5958-C-00000314- chan_sip.c: Header 7 16-: CSeq: 102 INVITE
DEBUG5958-C-00000314- chan_sip.c: Header 8 37-: User-Agent: Asterisk PBX 11.25.1-vici
DEBUG5958-C-00000314- chan_sip.c: Header 9 35-: Date: Wed, 19 Jul 2017 20:43:29 GMT
DEBUG5958-C-00000314- chan_sip.c: Header 10 90-: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
DEBUG5958-C-00000314- chan_sip.c: Header 11 26-: Supported: replaces, timer
DEBUG5958-C-00000314- chan_sip.c: Header 12 106-: Remote-Party-ID: "V7191643290000000537" <sip:4502391367 192.168.0.107>;party=calling;privacy=off;screen=no
DEBUG5958-C-00000314- chan_sip.c: Header 13 29-: Content-Type: application/sdp
VERBOSE5958-C-00000314- chan_sip.c: Reliably Transmitting (NAT) to 205.204.72.125:5060:
INVITE sip:1438XXXXXXX sip.myowntelco.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK1a7b4bf4;rport
Max-Forwards: 70
From: "V7191643290000000537" <sip:4502391367 192.168.0.107>;tag=as680e1bb5
To: <sip:1438XXXXXXX sip.myowntelco.net>
Contact: <sip:4502391367 192.168.0.107:5060>
Call-ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.25.1-vici
Date: Wed, 19 Jul 2017 20:43:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "V7191643290000000537" <sip:4502391367 192.168.0.107>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 2106041833 2106041833 IN IP4 192.168.0.107
s=Asterisk PBX 11.25.1-vici
c=IN IP4 192.168.0.107
t=0 0
m=audio 11898 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
DEBUG5958-C-00000314- chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7
DEBUG5958-C-00000314- chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 205.204.72.125:5060
VERBOSE5958-C-00000314- app_dial.c: -- Called SIP/1438XXXXXXX MyOwnTelco
DEBUG5958-C-00000314- channel.c: Set channel SIP/MyOwnTelco-00000220 to read format slin
DEBUG1864- manager.c: Examining event:
Event: Dial
Privilege: call,all
SubEvent: Begin
Channel: Local/70571819351438XXXXXXX default-000002bf;2
Destination: SIP/MyOwnTelco-00000220
CallerIDNum: 4502391367
CallerIDName: V7191643290000000537
ConnectedLineNum: 4502391367
ConnectedLineName: V7191643290000000537
UniqueID: 1500497009.1982
DestUniqueID: 1500497009.1983
Dialstring: 1438XXXXXXX MyOwnTelco


DEBUG5958-C-00000314- channel.c: Set channel SIP/MyOwnTelco-00000220 to write format slin
VERBOSE1760- chan_sip.c:
<--- SIP read from UDP:205.204.72.125:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK1a7b4bf4;received=192.168.0.107;rport=5060
From: "V7191643290000000537" <sip:4502391367 192.168.0.107:5060>;tag=as680e1bb5
To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as173d771a
Call-ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
CSeq: 102 INVITE
Server: MyOwnAccount
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c57c454"
Content-Length: 0

<------------->
DEBUG1760- chan_sip.c: Header 0 24-: SIP/2.0 401 Unauthorized
DEBUG1760- chan_sip.c: Header 1 92-: Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK1a7b4bf4;received=192.168.0.107;rport=5060
DEBUG1760- chan_sip.c: Header 2 79-: From: "V7191643290000000537" <sip:4502391367 192.168.0.107:5060>;tag=as680e1bb5
DEBUG1760- chan_sip.c: Header 3 55-: To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as173d771a
DEBUG1760- chan_sip.c: Header 4 60-: Call-ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
DEBUG1760- chan_sip.c: Header 5 16-: CSeq: 102 INVITE
DEBUG1760- chan_sip.c: Header 6 20-: Server: MyOwnAccount
DEBUG1760- chan_sip.c: Header 7 81-: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
DEBUG1760- chan_sip.c: Header 8 26-: Supported: replaces, timer
DEBUG1760- chan_sip.c: Header 9 74-: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c57c454"
DEBUG1760- chan_sip.c: Header 10 17-: Content-Length: 0
VERBOSE1760- chan_sip.c: --- (11 headers 0 lines) ---
DEBUG1760- chan_sip.c: = Looking for Call ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060 (Checking To) --From tag as680e1bb5 --To-tag as173d771a
DEBUG1760-C-00000314- chan_sip.c: Acked pending invite 102
DEBUG1760-C-00000314- chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7
DEBUG1760-C-00000314- chan_sip.c: Stopping retransmission on '4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060' of Request 102: Match Found
DEBUG1760-C-00000314- chan_sip.c: SIP response 401 to standard invite
VERBOSE1760-C-00000314- chan_sip.c: Transmitting (NAT) to 205.204.72.125:5060:
ACK sip:1438XXXXXXX sip.myowntelco.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK1a7b4bf4;rport
Max-Forwards: 70
From: "V7191643290000000537" <sip:4502391367 192.168.0.107>;tag=as680e1bb5
To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as173d771a
Contact: <sip:4502391367 192.168.0.107:5060>
Call-ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.25.1-vici
Content-Length: 0


---
DEBUG1760-C-00000314- chan_sip.c: Trying to put 'ACK sip:143' onto UDP socket destined for 205.204.72.125:5060
DEBUG1760-C-00000314- chan_sip.c: Auth attempt 1 on INVITE
DEBUG1760-C-00000314- chan_sip.c: ** Our capability: (gsm|ulaw) Video flag: False Text flag: False
DEBUG1760-C-00000314- chan_sip.c: ** Our prefcodec: (slin)
VERBOSE1760-C-00000314- chan_sip.c: Audio is at 11898
VERBOSE1760-C-00000314- chan_sip.c: Adding codec 100003 (ulaw) to SDP
VERBOSE1760-C-00000314- chan_sip.c: Adding codec 100002 (gsm) to SDP
VERBOSE1760-C-00000314- chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
DEBUG1760-C-00000314- chan_sip.c: -- Done with adding codecs to SDP
DEBUG1760-C-00000314- chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw)
VERBOSE1760-C-00000314- chan_sip.c: Reliably Transmitting (NAT) to 205.204.72.125:5060:
INVITE sip:1438XXXXXXX sip.myowntelco.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK5505d802;rport
Max-Forwards: 70
From: "V7191643290000000537" <sip:4502391367 192.168.0.107>;tag=as680e1bb5
To: <sip:1438XXXXXXX sip.myowntelco.net>
Contact: <sip:4502391367 192.168.0.107:5060>
Call-ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.25.1-vici
Authorization: Digest username="217164", realm="asterisk", algorithm=MD5, uri="sip:1438XXXXXXX sip.myowntelco.net", nonce="3c57c454", response="d2ca00a742139ec88a8158f965c0f171"
Date: Wed, 19 Jul 2017 20:43:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "V7191643290000000537" <sip:4502391367 192.168.0.107>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 2106041833 2106041834 IN IP4 192.168.0.107
s=Asterisk PBX 11.25.1-vici
c=IN IP4 192.168.0.107
t=0 0
m=audio 11898 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
DEBUG1760-C-00000314- chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #27
DEBUG1760-C-00000314- chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 205.204.72.125:5060
DEBUG1864- manager.c: Examining event:
Event: SIP-Hangup-Cause
Privilege: system,all
ChannelDriver: SIP
Channel: SIP/MyOwnTelco-00000220
CallerIDName: V7191643290000000537
Uniqueid: 1500497009.1983
Result: 401|Unauthorized


VERBOSE1760- chan_sip.c:
<--- SIP read from UDP:205.204.72.125:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK5505d802;received=192.168.0.107;rport=5060
From: "V7191643290000000537" <sip:4502391367 192.168.0.107:5060>;tag=as680e1bb5
To: <sip:1438XXXXXXX sip.myowntelco.net>
Call-ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
CSeq: 103 INVITE
Server: MyOwnAccount
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1438XXXXXXX 205.204.72.125:5060>
Content-Length: 0

<------------->
DEBUG1760- chan_sip.c: Header 0 18-: SIP/2.0 100 Trying
DEBUG1760- chan_sip.c: Header 1 92-: Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK5505d802;received=192.168.0.107;rport=5060
DEBUG1760- chan_sip.c: Header 2 79-: From: "V7191643290000000537" <sip:4502391367 192.168.0.107:5060>;tag=as680e1bb5
DEBUG1760- chan_sip.c: Header 3 40-: To: <sip:1438XXXXXXX sip.myowntelco.net>
DEBUG1760- chan_sip.c: Header 4 60-: Call-ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
DEBUG1760- chan_sip.c: Header 5 16-: CSeq: 103 INVITE
DEBUG1760- chan_sip.c: Header 6 20-: Server: MyOwnAccount
DEBUG1760- chan_sip.c: Header 7 81-: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
DEBUG1760- chan_sip.c: Header 8 26-: Supported: replaces, timer
DEBUG1760- chan_sip.c: Header 9 35-: Session-Expires: 1800;refresher=uas
DEBUG1760- chan_sip.c: Header 10 46-: Contact: <sip:1438XXXXXXX 205.204.72.125:5060>
DEBUG1760- chan_sip.c: Header 11 17-: Content-Length: 0
VERBOSE1760- chan_sip.c: --- (12 headers 0 lines) ---
DEBUG1760- chan_sip.c: = Looking for Call ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060 (Checking To) --From tag as680e1bb5 --To-tag
DEBUG1760-C-00000314- chan_sip.c: *** SIP TIMER: Cancelling retransmission #27 - INVITE (got response)
DEBUG1760-C-00000314- chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060' Request 103: Found
DEBUG1760-C-00000314- chan_sip.c: SIP response 100 to standard invite
VERBOSE1760- chan_sip.c:
<--- SIP read from UDP:205.204.72.125:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK5505d802;received=192.168.0.107;rport=5060
From: "V7191643290000000537" <sip:4502391367 192.168.0.107:5060>;tag=as680e1bb5
To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as5d291fc1
Call-ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
CSeq: 103 INVITE
Server: MyOwnAccount
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1438XXXXXXX 205.204.72.125:5060>
Content-Length: 0

<------------->
DEBUG1760- chan_sip.c: Header 0 19-: SIP/2.0 180 Ringing
DEBUG1760- chan_sip.c: Header 1 92-: Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK5505d802;received=192.168.0.107;rport=5060
DEBUG1760- chan_sip.c: Header 2 79-: From: "V7191643290000000537" <sip:4502391367 192.168.0.107:5060>;tag=as680e1bb5
DEBUG1760- chan_sip.c: Header 3 55-: To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as5d291fc1
DEBUG1760- chan_sip.c: Header 4 60-: Call-ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060
DEBUG1760- chan_sip.c: Header 5 16-: CSeq: 103 INVITE
DEBUG1760- chan_sip.c: Header 6 20-: Server: MyOwnAccount
DEBUG1760- chan_sip.c: Header 7 81-: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
DEBUG1760- chan_sip.c: Header 8 26-: Supported: replaces, timer
DEBUG1760- chan_sip.c: Header 9 35-: Session-Expires: 1800;refresher=uas
DEBUG1760- chan_sip.c: Header 10 46-: Contact: <sip:1438XXXXXXX 205.204.72.125:5060>
DEBUG1760- chan_sip.c: Header 11 17-: Content-Length: 0
VERBOSE1760- chan_sip.c: --- (12 headers 0 lines) ---
DEBUG1760- chan_sip.c: = Looking for Call ID: 4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060 (Checking To) --From tag as680e1bb5 --To-tag as5d291fc1
DEBUG1760-C-00000314- chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4c3821a46334638d0d589f8b23dc41fc 192.168.0.107:5060' Request 103: Found
DEBUG1760-C-00000314- chan_sip.c: SIP response 180 to standard invite
DEBUG1760-C-00000314- chan_sip.c: build_route: Contact hop: <sip:1438XXXXXXX 205.204.72.125:5060>
VERBOSE1760-C-00000314- chan_sip.c: list_route: hop: <sip:1438XXXXXXX 205.204.72.125:5060>
DEBUG1698- devicestate.c: No provider found, checking channel drivers for SIP - MyOwnTelco
DEBUG1698- chan_sip.c: Checking device state for peer MyOwnTelco
DEBUG1698- devicestate.c: Changing state for SIP/MyOwnTelco - state 1 (Not in use)
DEBUG1698- devicestate.c: device 'SIP/MyOwnTelco' state '1'
DEBUG1864- manager.c: Examining event:
Event: Newstate
Privilege: call,all
Channel: SIP/MyOwnTelco-00000220
ChannelState: 5
ChannelStateDesc: Ringing
CallerIDNum: 4502391367
CallerIDName: V7191643290000000537
ConnectedLineNum: 4502391367
ConnectedLineName: V7191643290000000537
Uniqueid: 1500497009.1983


VERBOSE5958-C-00000314- app_dial.c: -- SIP/MyOwnTelco-00000220 is ringing
DEBUG1698- devicestate.c: No provider found, checking channel drivers for Local - 70571819351438XXXXXXX default
DEBUG1698- chan_local.c: Checking if extension 70571819351438XXXXXXX default exists (devicestate)
DEBUG1698- devicestate.c: Changing state for Local/70571819351438XXXXXXX default - state 2 (In use)
DEBUG1698- devicestate.c: device 'Local/70571819351438XXXXXXX default' state '2'
DEBUG1864- manager.c: Examining event:
Event: Newstate
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002bf;1
ChannelState: 5
ChannelStateDesc: Ringing
CallerIDNum: 4502391367
CallerIDName: V7191643290000000537
ConnectedLineNum: 4502391367
ConnectedLineName: V7191643290000000537
Uniqueid: 1500497009.1981


DEBUG5962- manager.c: Running action 'Login'
VERBOSE5962- manager.c: == Manager 'sendcron' logged on from 127.0.0.1
DEBUG5962- manager.c: Running action 'Originate'
DEBUG1864- manager.c: Examining event:
Event: Newchannel
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002c0;1
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten: 70571819351438XXXXXXX
Context: default
Uniqueid: 1500497009.1984


DEBUG1864- manager.c: Examining event:
Event: Newchannel
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002c0;2
ChannelState: 4
ChannelStateDesc: Ring
CallerIDNum:
CallerIDName:
AccountCode:
Exten: 70571819351438XXXXXXX
Context: default
Uniqueid: 1500497009.1985


DEBUG1864- manager.c: Examining event:
Event: NewAccountCode
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002c0;1
Uniqueid: 1500497009.1984
AccountCode:
OldAccountCode:


DEBUG1864- manager.c: Examining event:
Event: NewCallerid
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002c0;1
CallerIDNum: 4502391367
CallerIDName: V7191643290000000536
Uniqueid: 1500497009.1984
CID-CallingPres: 0 (Presentation Allowed, Not Screened)


DEBUG1864- manager.c: Examining event:
Event: LocalBridge
Privilege: call,all
Channel1: Local/70571819351438XXXXXXX default-000002c0;1
Channel2: Local/70571819351438XXXXXXX default-000002c0;2
Uniqueid1: 1500497009.1984
Uniqueid2: 1500497009.1985
Context: default
Exten: 70571819351438XXXXXXX
LocalOptimization: Yes


DEBUG5963-C-00000315- pbx.c: Launching 'AGI'
VERBOSE5963-C-00000315- pbx.c: -- Executing 70571819351438XXXXXXX default:1- AGI("Local/70571819351438XXXXXXX default-000002c0;2", "agi://127.0.0.1:4577/call_log") in new stack
DEBUG5963-C-00000315- netsock2.c: Splitting '127.0.0.1:4577' into...
DEBUG5963-C-00000315- netsock2.c: ...host '127.0.0.1' and port '4577'.
DEBUG5963-C-00000315- res_agi.c: Wow, connected!
VERBOSE5963-C-00000315- res_agi.c: -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=75432148))
VERBOSE5963-C-00000315- res_agi.c: -- <Local/70571819351438XXXXXXX default-000002c0;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
DEBUG5963-C-00000315- pbx.c: Result of 'EXTEN' is '70571819351438XXXXXXX'
DEBUG5963-C-00000315- pbx.c: Launching 'Dial'
VERBOSE5963-C-00000315- pbx.c: -- Executing 70571819351438XXXXXXX default:2- Dial("Local/70571819351438XXXXXXX default-000002c0;2", "SIP/1438XXXXXXX MyOwnTelco,,tTo") in new stack
DEBUG5963-C-00000315- chan_sip.c: Asked to create a SIP channel with formats: (slin)
DEBUG5963-C-00000315- chan_sip.c: Allocating new SIP dialog for 6897343c61a521de41a67a341e1e0cc7 127.0.0.2:5060 - INVITE (No RTP)
DEBUG5963-C-00000315- rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f01dc01c8e8'
DEBUG5963-C-00000315- res_rtp_asterisk.c: Allocated port 16024 for RTP instance '0x7f01dc01c8e8'
DEBUG5963-C-00000315- res_rtp_asterisk.c: Creating ICE session 0.0.0.0:16024 (16024) for RTP instance '0x7f01dc01c8e8'
DEBUG5963-C-00000315- netsock2.c: Splitting '192.168.0.107' into...
DEBUG5963-C-00000315- netsock2.c: ...host '192.168.0.107' and port ''.
DEBUG5963-C-00000315- rtp_engine.c: RTP instance '0x7f01dc01c8e8' is setup and ready to go
DEBUG5963-C-00000315- res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f01dc01c8e8'
VERBOSE5963-C-00000315- netsock2.c: == Using SIP RTP CoS mark 5
DEBUG5963-C-00000315- chan_sip.c: Setting NAT on RTP to On
DEBUG5963-C-00000315- chan_sip.c: OBPROXY: Not applying OBproxy to this call
DEBUG5963-C-00000315- acl.c: For destination '205.204.72.125', our source address is '192.168.0.107'.
DEBUG5963-C-00000315- chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.0.107:5060
DEBUG5963-C-00000315- chan_sip.c: Setting NAT on RTP to On
DEBUG5963-C-00000315- chan_sip.c: SIP call-id changed from '6897343c61a521de41a67a341e1e0cc7 127.0.0.2:5060' to '59baed1d574a4b557202d2042a83e668 192.168.0.107:5060'
DEBUG1864- manager.c: Examining event:
Event: Newchannel
Privilege: call,all
Channel: SIP/MyOwnTelco-00000221
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: default
Uniqueid: 1500497009.1986


DEBUG5963-C-00000315- chan_sip.c: *** Our native formats are (ulaw)
DEBUG5963-C-00000315- chan_sip.c: *** Joint capabilities are (nothing)
DEBUG5963-C-00000315- chan_sip.c: *** Our capabilities are (gsm|ulaw)
DEBUG5963-C-00000315- chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
DEBUG5963-C-00000315- chan_sip.c: *** Our preferred formats from the incoming channel are (slin)
DEBUG5963-C-00000315- chan_sip.c: This channel will not be able to handle video.
DEBUG1864- manager.c: Examining event:
Event: ChannelUpdate
Privilege: system,all
Channel: SIP/MyOwnTelco-00000221
Uniqueid: 1500497009.1986
Channeltype: SIP
SIPcallid: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
SIPfullcontact:


DEBUG1864- manager.c: Examining event:
Event: ChannelUpdate
Privilege: system,all
Channel: SIP/MyOwnTelco-00000221
Channeltype: SIP
SIPcallid: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
SIPfullcontact:
Peername: MyOwnTelco


DEBUG5963-C-00000315- channel_internal_api.c: Channel Call ID changing from C-00000315- to C-00000315-
DEBUG5963-C-00000315- rtp_engine.c: Can't find native functions for channel 'Local/70571819351438XXXXXXX default-000002c0;2'
DEBUG5963-C-00000315- channel.c: Inheriting variable CAMPCUST from Local/70571819351438XXXXXXX default-000002c0;2 to SIP/MyOwnTelco-00000221.
DEBUG1864- manager.c: Examining event:
Event: NewCallerid
Privilege: call,all
Channel: SIP/MyOwnTelco-00000221
CallerIDNum: 4502391367
CallerIDName: V7191643290000000536
Uniqueid: 1500497009.1986
CID-CallingPres: 0 (Presentation Allowed, Not Screened)


DEBUG5963-C-00000315- chan_sip.c: Outgoing Call for 1438XXXXXXX
DEBUG5963-C-00000315- chan_sip.c: Updating call counter for outgoing call
DEBUG5963-C-00000315- chan_sip.c: ** Our capability: (gsm|ulaw) Video flag: False Text flag: False
DEBUG5963-C-00000315- chan_sip.c: ** Our prefcodec: (slin)
VERBOSE5963-C-00000315- chan_sip.c: Audio is at 16024
VERBOSE5963-C-00000315- chan_sip.c: Adding codec 100003 (ulaw) to SDP
VERBOSE5963-C-00000315- chan_sip.c: Adding codec 100002 (gsm) to SDP
VERBOSE5963-C-00000315- chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
DEBUG5963-C-00000315- chan_sip.c: -- Done with adding codecs to SDP
DEBUG5963-C-00000315- chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw)
DEBUG5963-C-00000315- chan_sip.c: Initializing initreq for method INVITE - callid 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
DEBUG5963-C-00000315- chan_sip.c: Header 0 49-: INVITE sip:1438XXXXXXX sip.myowntelco.net SIP/2.0
DEBUG5963-C-00000315- chan_sip.c: Header 1 64-: Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK6782f23a;rport
DEBUG5963-C-00000315- chan_sip.c: Header 2 16-: Max-Forwards: 70
DEBUG5963-C-00000315- chan_sip.c: Header 3 74-: From: "V7191643290000000536" <sip:4502391367 192.168.0.107>;tag=as45caaa0c
DEBUG5963-C-00000315- chan_sip.c: Header 4 40-: To: <sip:1438XXXXXXX sip.myowntelco.net>
DEBUG5963-C-00000315- chan_sip.c: Header 5 44-: Contact: <sip:4502391367 192.168.0.107:5060>
DEBUG5963-C-00000315- chan_sip.c: Header 6 60-: Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
DEBUG5963-C-00000315- chan_sip.c: Header 7 16-: CSeq: 102 INVITE
DEBUG5963-C-00000315- chan_sip.c: Header 8 37-: User-Agent: Asterisk PBX 11.25.1-vici
DEBUG5963-C-00000315- chan_sip.c: Header 9 35-: Date: Wed, 19 Jul 2017 20:43:29 GMT
DEBUG5963-C-00000315- chan_sip.c: Header 10 90-: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
DEBUG5963-C-00000315- chan_sip.c: Header 11 26-: Supported: replaces, timer
DEBUG5963-C-00000315- chan_sip.c: Header 12 106-: Remote-Party-ID: "V7191643290000000536" <sip:4502391367 192.168.0.107>;party=calling;privacy=off;screen=no
DEBUG5963-C-00000315- chan_sip.c: Header 13 29-: Content-Type: application/sdp
VERBOSE5963-C-00000315- chan_sip.c: Reliably Transmitting (NAT) to 205.204.72.125:5060:
INVITE sip:1438XXXXXXX sip.myowntelco.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK6782f23a;rport
Max-Forwards: 70
From: "V7191643290000000536" <sip:4502391367 192.168.0.107>;tag=as45caaa0c
To: <sip:1438XXXXXXX sip.myowntelco.net>
Contact: <sip:4502391367 192.168.0.107:5060>
Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.25.1-vici
Date: Wed, 19 Jul 2017 20:43:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "V7191643290000000536" <sip:4502391367 192.168.0.107>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 1150044292 1150044292 IN IP4 192.168.0.107
s=Asterisk PBX 11.25.1-vici
c=IN IP4 192.168.0.107
t=0 0
m=audio 16024 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
DEBUG5963-C-00000315- chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #3
DEBUG5963-C-00000315- chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 205.204.72.125:5060
VERBOSE5963-C-00000315- app_dial.c: -- Called SIP/1438XXXXXXX MyOwnTelco
DEBUG1864- manager.c: Examining event:
Event: Dial
Privilege: call,all
SubEvent: Begin
Channel: Local/70571819351438XXXXXXX default-000002c0;2
Destination: SIP/MyOwnTelco-00000221
CallerIDNum: 4502391367
CallerIDName: V7191643290000000536
ConnectedLineNum: 4502391367
ConnectedLineName: V7191643290000000536
UniqueID: 1500497009.1985
DestUniqueID: 1500497009.1986
Dialstring: 1438XXXXXXX MyOwnTelco


DEBUG5963-C-00000315- channel.c: Set channel SIP/MyOwnTelco-00000221 to read format slin
DEBUG5963-C-00000315- channel.c: Set channel SIP/MyOwnTelco-00000221 to write format slin
DEBUG1856- manager.c: Running action 'Command'
VERBOSE1760- chan_sip.c:
<--- SIP read from UDP:205.204.72.125:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK6782f23a;received=192.168.0.107;rport=5060
From: "V7191643290000000536" <sip:4502391367 192.168.0.107:5060>;tag=as45caaa0c
To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as5039dd13
Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
CSeq: 102 INVITE
Server: MyOwnAccount
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0d78fa45"
Content-Length: 0

<------------->
DEBUG1760- chan_sip.c: Header 0 24-: SIP/2.0 401 Unauthorized
DEBUG1760- chan_sip.c: Header 1 92-: Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK6782f23a;received=192.168.0.107;rport=5060
DEBUG1760- chan_sip.c: Header 2 79-: From: "V7191643290000000536" <sip:4502391367 192.168.0.107:5060>;tag=as45caaa0c
DEBUG1760- chan_sip.c: Header 3 55-: To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as5039dd13
DEBUG1760- chan_sip.c: Header 4 60-: Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
DEBUG1760- chan_sip.c: Header 5 16-: CSeq: 102 INVITE
DEBUG1760- chan_sip.c: Header 6 20-: Server: MyOwnAccount
DEBUG1760- chan_sip.c: Header 7 81-: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
DEBUG1760- chan_sip.c: Header 8 26-: Supported: replaces, timer
DEBUG1760- chan_sip.c: Header 9 74-: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0d78fa45"
DEBUG1760- chan_sip.c: Header 10 17-: Content-Length: 0
VERBOSE1760- chan_sip.c: --- (11 headers 0 lines) ---
DEBUG1760- chan_sip.c: = Looking for Call ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060 (Checking To) --From tag as45caaa0c --To-tag as5039dd13
DEBUG1760-C-00000315- chan_sip.c: Acked pending invite 102
DEBUG1760-C-00000315- chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #3
DEBUG1760-C-00000315- chan_sip.c: Stopping retransmission on '59baed1d574a4b557202d2042a83e668 192.168.0.107:5060' of Request 102: Match Found
DEBUG1760-C-00000315- chan_sip.c: SIP response 401 to standard invite
VERBOSE1760-C-00000315- chan_sip.c: Transmitting (NAT) to 205.204.72.125:5060:
ACK sip:1438XXXXXXX sip.myowntelco.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK6782f23a;rport
Max-Forwards: 70
From: "V7191643290000000536" <sip:4502391367 192.168.0.107>;tag=as45caaa0c
To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as5039dd13
Contact: <sip:4502391367 192.168.0.107:5060>
Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.25.1-vici
Content-Length: 0


---
DEBUG1760-C-00000315- chan_sip.c: Trying to put 'ACK sip:143' onto UDP socket destined for 205.204.72.125:5060
DEBUG1760-C-00000315- chan_sip.c: Auth attempt 1 on INVITE
DEBUG1864- manager.c: Examining event:
Event: SIP-Hangup-Cause
Privilege: system,all
ChannelDriver: SIP
Channel: SIP/MyOwnTelco-00000221
CallerIDName: V7191643290000000536
Uniqueid: 1500497009.1986
Result: 401|Unauthorized


DEBUG1760-C-00000315- chan_sip.c: ** Our capability: (gsm|ulaw) Video flag: False Text flag: False
DEBUG1760-C-00000315- chan_sip.c: ** Our prefcodec: (slin)
VERBOSE1760-C-00000315- chan_sip.c: Audio is at 16024
VERBOSE1760-C-00000315- chan_sip.c: Adding codec 100003 (ulaw) to SDP
VERBOSE1760-C-00000315- chan_sip.c: Adding codec 100002 (gsm) to SDP
VERBOSE1760-C-00000315- chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
DEBUG1760-C-00000315- chan_sip.c: -- Done with adding codecs to SDP
DEBUG1760-C-00000315- chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw)
VERBOSE1760-C-00000315- chan_sip.c: Reliably Transmitting (NAT) to 205.204.72.125:5060:
INVITE sip:1438XXXXXXX sip.myowntelco.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK66a2786d;rport
Max-Forwards: 70
From: "V7191643290000000536" <sip:4502391367 192.168.0.107>;tag=as45caaa0c
To: <sip:1438XXXXXXX sip.myowntelco.net>
Contact: <sip:4502391367 192.168.0.107:5060>
Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.25.1-vici
Authorization: Digest username="217164", realm="asterisk", algorithm=MD5, uri="sip:1438XXXXXXX sip.myowntelco.net", nonce="0d78fa45", response="e8c97c60a34012461950689772fd5552"
Date: Wed, 19 Jul 2017 20:43:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "V7191643290000000536" <sip:4502391367 192.168.0.107>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 1150044292 1150044293 IN IP4 192.168.0.107
s=Asterisk PBX 11.25.1-vici
c=IN IP4 192.168.0.107
t=0 0
m=audio 16024 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
DEBUG1760-C-00000315- chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #28
DEBUG1760-C-00000315- chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 205.204.72.125:5060
VERBOSE1760- chan_sip.c:
<--- SIP read from UDP:205.204.72.125:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK66a2786d;received=192.168.0.107;rport=5060
From: "V7191643290000000536" <sip:4502391367 192.168.0.107:5060>;tag=as45caaa0c
To: <sip:1438XXXXXXX sip.myowntelco.net>
Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
CSeq: 103 INVITE
Server: MyOwnAccount
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1438XXXXXXX 205.204.72.125:5060>
Content-Length: 0

<------------->
DEBUG1760- chan_sip.c: Header 0 18-: SIP/2.0 100 Trying
DEBUG1760- chan_sip.c: Header 1 92-: Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK66a2786d;received=192.168.0.107;rport=5060
DEBUG1760- chan_sip.c: Header 2 79-: From: "V7191643290000000536" <sip:4502391367 192.168.0.107:5060>;tag=as45caaa0c
DEBUG1760- chan_sip.c: Header 3 40-: To: <sip:1438XXXXXXX sip.myowntelco.net>
DEBUG1760- chan_sip.c: Header 4 60-: Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
DEBUG1760- chan_sip.c: Header 5 16-: CSeq: 103 INVITE
DEBUG1760- chan_sip.c: Header 6 20-: Server: MyOwnAccount
DEBUG1760- chan_sip.c: Header 7 81-: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
DEBUG1760- chan_sip.c: Header 8 26-: Supported: replaces, timer
DEBUG1760- chan_sip.c: Header 9 35-: Session-Expires: 1800;refresher=uas
DEBUG1760- chan_sip.c: Header 10 46-: Contact: <sip:1438XXXXXXX 205.204.72.125:5060>
DEBUG1760- chan_sip.c: Header 11 17-: Content-Length: 0
VERBOSE1760- chan_sip.c: --- (12 headers 0 lines) ---
DEBUG1760- chan_sip.c: = Looking for Call ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060 (Checking To) --From tag as45caaa0c --To-tag
DEBUG1760-C-00000315- chan_sip.c: *** SIP TIMER: Cancelling retransmission #28 - INVITE (got response)
DEBUG1760-C-00000315- chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '59baed1d574a4b557202d2042a83e668 192.168.0.107:5060' Request 103: Found
DEBUG1760-C-00000315- chan_sip.c: SIP response 100 to standard invite
VERBOSE1760- chan_sip.c:
<--- SIP read from UDP:205.204.72.125:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK66a2786d;received=192.168.0.107;rport=5060
From: "V7191643290000000536" <sip:4502391367 192.168.0.107:5060>;tag=as45caaa0c
To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as6cc0a381
Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
CSeq: 103 INVITE
Server: MyOwnAccount
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1438XXXXXXX 205.204.72.125:5060>
Content-Length: 0

<------------->
DEBUG1760- chan_sip.c: Header 0 19-: SIP/2.0 180 Ringing
DEBUG1760- chan_sip.c: Header 1 92-: Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK66a2786d;received=192.168.0.107;rport=5060
DEBUG1760- chan_sip.c: Header 2 79-: From: "V7191643290000000536" <sip:4502391367 192.168.0.107:5060>;tag=as45caaa0c
DEBUG1760- chan_sip.c: Header 3 55-: To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as6cc0a381
DEBUG1760- chan_sip.c: Header 4 60-: Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
DEBUG1760- chan_sip.c: Header 5 16-: CSeq: 103 INVITE
DEBUG1760- chan_sip.c: Header 6 20-: Server: MyOwnAccount
DEBUG1760- chan_sip.c: Header 7 81-: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
DEBUG1760- chan_sip.c: Header 8 26-: Supported: replaces, timer
DEBUG1760- chan_sip.c: Header 9 35-: Session-Expires: 1800;refresher=uas
DEBUG1760- chan_sip.c: Header 10 46-: Contact: <sip:1438XXXXXXX 205.204.72.125:5060>
DEBUG1760- chan_sip.c: Header 11 17-: Content-Length: 0
VERBOSE1760- chan_sip.c: --- (12 headers 0 lines) ---
DEBUG1760- chan_sip.c: = Looking for Call ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060 (Checking To) --From tag as45caaa0c --To-tag as6cc0a381
DEBUG1760-C-00000315- chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '59baed1d574a4b557202d2042a83e668 192.168.0.107:5060' Request 103: Found
DEBUG1760-C-00000315- chan_sip.c: SIP response 180 to standard invite
DEBUG1760-C-00000315- chan_sip.c: build_route: Contact hop: <sip:1438XXXXXXX 205.204.72.125:5060>
VERBOSE1760-C-00000315- chan_sip.c: list_route: hop: <sip:1438XXXXXXX 205.204.72.125:5060>
DEBUG1698- devicestate.c: No provider found, checking channel drivers for SIP - MyOwnTelco
DEBUG1698- chan_sip.c: Checking device state for peer MyOwnTelco
DEBUG1698- devicestate.c: Changing state for SIP/MyOwnTelco - state 1 (Not in use)
DEBUG1698- devicestate.c: device 'SIP/MyOwnTelco' state '1'
DEBUG1864- manager.c: Examining event:
Event: Newstate
Privilege: call,all
Channel: SIP/MyOwnTelco-00000221
ChannelState: 5
ChannelStateDesc: Ringing
CallerIDNum: 4502391367
CallerIDName: V7191643290000000536
ConnectedLineNum: 4502391367
ConnectedLineName: V7191643290000000536
Uniqueid: 1500497009.1986


VERBOSE5963-C-00000315- app_dial.c: -- SIP/MyOwnTelco-00000221 is ringing
DEBUG1698- devicestate.c: No provider found, checking channel drivers for Local - 70571819351438XXXXXXX default
DEBUG1698- chan_local.c: Checking if extension 70571819351438XXXXXXX default exists (devicestate)
DEBUG1698- devicestate.c: Changing state for Local/70571819351438XXXXXXX default - state 2 (In use)
DEBUG1698- devicestate.c: device 'Local/70571819351438XXXXXXX default' state '2'
DEBUG1864- manager.c: Examining event:
Event: Newstate
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002c0;1
ChannelState: 5
ChannelStateDesc: Ringing
CallerIDNum: 4502391367
CallerIDName: V7191643290000000536
ConnectedLineNum: 4502391367
ConnectedLineName: V7191643290000000536
Uniqueid: 1500497009.1984


VERBOSE1760- chan_sip.c:
<--- SIP read from UDP:205.204.72.125:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK66a2786d;received=192.168.0.107;rport=5060
From: "V7191643290000000536" <sip:4502391367 192.168.0.107:5060>;tag=as45caaa0c
To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as6cc0a381
Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
CSeq: 103 INVITE
Server: MyOwnAccount
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1438XXXXXXX 205.204.72.125:5060>
Content-Length: 0

<------------->
DEBUG1760- chan_sip.c: Header 0 19-: SIP/2.0 180 Ringing
DEBUG1760- chan_sip.c: Header 1 92-: Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK66a2786d;received=192.168.0.107;rport=5060
DEBUG1760- chan_sip.c: Header 2 79-: From: "V7191643290000000536" <sip:4502391367 192.168.0.107:5060>;tag=as45caaa0c
DEBUG1760- chan_sip.c: Header 3 55-: To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as6cc0a381
DEBUG1760- chan_sip.c: Header 4 60-: Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
DEBUG1760- chan_sip.c: Header 5 16-: CSeq: 103 INVITE
DEBUG1760- chan_sip.c: Header 6 20-: Server: MyOwnAccount
DEBUG1760- chan_sip.c: Header 7 81-: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
DEBUG1760- chan_sip.c: Header 8 26-: Supported: replaces, timer
DEBUG1760- chan_sip.c: Header 9 35-: Session-Expires: 1800;refresher=uas
DEBUG1760- chan_sip.c: Header 10 46-: Contact: <sip:1438XXXXXXX 205.204.72.125:5060>
DEBUG1760- chan_sip.c: Header 11 17-: Content-Length: 0
VERBOSE1760- chan_sip.c: --- (12 headers 0 lines) ---
DEBUG1760- chan_sip.c: = Looking for Call ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060 (Checking To) --From tag as45caaa0c --To-tag as6cc0a381
DEBUG1760-C-00000315- chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '59baed1d574a4b557202d2042a83e668 192.168.0.107:5060' Request 103: Found
DEBUG1760-C-00000315- chan_sip.c: SIP response 180 to standard invite
DEBUG1760-C-00000315- chan_sip.c: build_route: Contact hop: <sip:1438XXXXXXX 205.204.72.125:5060>
VERBOSE1760-C-00000315- chan_sip.c: list_route: hop: <sip:1438XXXXXXX 205.204.72.125:5060>
VERBOSE5963-C-00000315- app_dial.c: -- SIP/MyOwnTelco-00000221 is ringing
VERBOSE1760- chan_sip.c:
<--- SIP read from UDP:205.204.72.125:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK66a2786d;received=192.168.0.107;rport=5060
From: "V7191643290000000536" <sip:4502391367 192.168.0.107:5060>;tag=as45caaa0c
To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as6cc0a381
Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
CSeq: 103 INVITE
Server: MyOwnAccount
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
DEBUG1760- chan_sip.c: Header 0 21-: SIP/2.0 403 Forbidden
DEBUG1760- chan_sip.c: Header 1 92-: Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK66a2786d;received=192.168.0.107;rport=5060
DEBUG1760- chan_sip.c: Header 2 79-: From: "V7191643290000000536" <sip:4502391367 192.168.0.107:5060>;tag=as45caaa0c
DEBUG1760- chan_sip.c: Header 3 55-: To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as6cc0a381
DEBUG1760- chan_sip.c: Header 4 60-: Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
DEBUG1760- chan_sip.c: Header 5 16-: CSeq: 103 INVITE
DEBUG1760- chan_sip.c: Header 6 20-: Server: MyOwnAccount
DEBUG1760- chan_sip.c: Header 7 81-: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
DEBUG1760- chan_sip.c: Header 8 26-: Supported: replaces, timer
DEBUG1760- chan_sip.c: Header 9 17-: Content-Length: 0
VERBOSE1760- chan_sip.c: --- (10 headers 0 lines) ---
DEBUG1760- chan_sip.c: = Looking for Call ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060 (Checking To) --From tag as45caaa0c --To-tag as6cc0a381
DEBUG1760-C-00000315- chan_sip.c: Acked pending invite 103
DEBUG1760-C-00000315- chan_sip.c: Stopping retransmission on '59baed1d574a4b557202d2042a83e668 192.168.0.107:5060' of Request 103: Match Found
DEBUG1760-C-00000315- chan_sip.c: SIP response 403 to standard invite
DEBUG1760-C-00000315- chan_sip.c: Strict routing enforced for session 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
VERBOSE1760-C-00000315- chan_sip.c: Transmitting (NAT) to 205.204.72.125:5060:
ACK sip:1438XXXXXXX 205.204.72.125:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK66a2786d;rport
Max-Forwards: 70
From: "V7191643290000000536" <sip:4502391367 192.168.0.107>;tag=as45caaa0c
To: <sip:1438XXXXXXX sip.myowntelco.net>;tag=as6cc0a381
Contact: <sip:4502391367 192.168.0.107:5060>
Call-ID: 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.25.1-vici
Content-Length: 0


---
DEBUG1760-C-00000315- chan_sip.c: Trying to put 'ACK sip:143' onto UDP socket destined for 205.204.72.125:5060
WARNING1760-C-00000315- chan_sip.c: Received response: "Forbidden" from '"V7191643290000000536" <sip:4502391367 192.168.0.107>;tag=as45caaa0c'
DEBUG5963-C-00000315- channel.c: Hanging up channel 'SIP/MyOwnTelco-00000221'
DEBUG5963-C-00000315- chan_sip.c: Hangup call SIP/MyOwnTelco-00000221, SIP callid 59baed1d574a4b557202d2042a83e668 192.168.0.107:5060
DEBUG5963-C-00000315- res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f01dc01c8e8'
VERBOSE5963-C-00000315- chan_sip.c: Scheduling destruction of SIP dialog '59baed1d574a4b557202d2042a83e668 192.168.0.107:5060' in 32000 ms (Method: INVITE)
DEBUG1698- devicestate.c: No provider found, checking channel drivers for SIP - MyOwnTelco
VERBOSE5963-C-00000315- app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
DEBUG1698- chan_sip.c: Checking device state for peer MyOwnTelco
DEBUG1698- devicestate.c: Changing state for SIP/MyOwnTelco - state 1 (Not in use)
DEBUG1698- devicestate.c: device 'SIP/MyOwnTelco' state '1'
DEBUG5963-C-00000315- app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.
DEBUG5963-C-00000315- pbx.c: Launching 'Hangup'
VERBOSE5963-C-00000315- pbx.c: -- Executing 70571819351438XXXXXXX default:3- Hangup("Local/70571819351438XXXXXXX default-000002c0;2", "") in new stack
DEBUG5963-C-00000315- channel.c: Soft-Hanging up channel 'Local/70571819351438XXXXXXX default-000002c0;2'
DEBUG5963-C-00000315- pbx.c: Spawn extension (default,70571819351438XXXXXXX,3) exited non-zero on 'Local/70571819351438XXXXXXX default-000002c0;2'
VERBOSE5963-C-00000315- pbx.c: == Spawn extension (default, 70571819351438XXXXXXX, 3) exited non-zero on 'Local/70571819351438XXXXXXX default-000002c0;2'
DEBUG5963-C-00000315- channel.c: Soft-Hanging up channel 'Local/70571819351438XXXXXXX default-000002c0;2'
DEBUG5963-C-00000315- channel.c: Soft-Hanging up channel 'Local/70571819351438XXXXXXX default-000002c0;2'
DEBUG5963-C-00000315- pbx.c: Result of 'HANGUPCAUSE' is '21'
DEBUG5963-C-00000315- pbx.c: Result of 'DIALSTATUS' is 'CHANUNAVAIL'
DEBUG5963-C-00000315- pbx.c: Result of 'DIALEDTIME' is ''
DEBUG5963-C-00000315- pbx.c: Result of 'ANSWEREDTIME' is ''
DEBUG5963-C-00000315- pbx.c: Launching 'AGI'
VERBOSE5963-C-00000315- pbx.c: -- Executing h default:1- AGI("Local/70571819351438XXXXXXX default-000002c0;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CHANUNAVAIL----------") in new stack
DEBUG5963-C-00000315- res_agi.c: Hungup channel detected, running agi in dead mode.
DEBUG5963-C-00000315- netsock2.c: Splitting '127.0.0.1:4577' into...
DEBUG5963-C-00000315- netsock2.c: ...host '127.0.0.1' and port '4577'.
DEBUG1864- manager.c: Examining event:
Event: SIP-Hangup-Cause
Privilege: system,all
ChannelDriver: SIP
Channel: SIP/MyOwnTelco-00000221
CallerIDName: V7191643290000000536
Uniqueid: 1500497009.1986
Result: 403|Forbidden


DEBUG1864- manager.c: Examining event:
Event: HangupRequest
Privilege: call,all
Channel: SIP/MyOwnTelco-00000221
Uniqueid: 1500497009.1986
Cause: 21


DEBUG1864- manager.c: Examining event:
Event: Hangup
Privilege: call,all
Channel: SIP/MyOwnTelco-00000221
Uniqueid: 1500497009.1986
CallerIDNum: 4502391367
CallerIDName: V7191643290000000536
ConnectedLineNum: 4502391367
ConnectedLineName: V7191643290000000536
AccountCode:
Cause: 21
Cause-txt: Call Rejected


DEBUG1864- manager.c: Examining event:
Event: Dial
Privilege: call,all
SubEvent: End
Channel: Local/70571819351438XXXXXXX default-000002c0;2
UniqueID: 1500497009.1985
DialStatus: CHANUNAVAIL


DEBUG1864- manager.c: Examining event:
Event: SoftHangupRequest
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002c0;2
Uniqueid: 1500497009.1985
Cause: 16


DEBUG5963-C-00000315- res_agi.c: Wow, connected!
Jul 19 16:43:30- DEBUG1856- manager.c: Running action 'Command'
Jul 19 16:43:30- DEBUG1856- manager.c: Running action 'Command'
Jul 19 16:43:30- VERBOSE5963-C-00000315- res_agi.c: Jul 19 16:43:30- -- <Local/70571819351438XXXXXXX default-000002c0;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Jul 19 16:43:30- DEBUG5963-C-00000315- channel.c: Hanging up channel 'Local/70571819351438XXXXXXX default-000002c0;2'
Jul 19 16:43:30- DEBUG5962-C-00000315- channel.c: Hanging up channel 'Local/70571819351438XXXXXXX default-000002c0;1'
Jul 19 16:43:30- DEBUG1698- devicestate.c: No provider found, checking channel drivers for Local - 70571819351438XXXXXXX default
Jul 19 16:43:30- DEBUG1698- chan_local.c: Checking if extension 70571819351438XXXXXXX default exists (devicestate)
Jul 19 16:43:30- DEBUG1698- devicestate.c: Changing state for Local/70571819351438XXXXXXX default - state 1 (Not in use)
Jul 19 16:43:30- DEBUG1698- devicestate.c: device 'Local/70571819351438XXXXXXX default' state '1'
Jul 19 16:43:30- DEBUG1698- devicestate.c: No provider found, checking channel drivers for Local - 70571819351438XXXXXXX default
Jul 19 16:43:30- DEBUG1698- chan_local.c: Checking if extension 70571819351438XXXXXXX default exists (devicestate)
Jul 19 16:43:30- DEBUG1698- devicestate.c: Changing state for Local/70571819351438XXXXXXX default - state 1 (Not in use)
Jul 19 16:43:30- DEBUG1698- devicestate.c: device 'Local/70571819351438XXXXXXX default' state '1'
Jul 19 16:43:30- DEBUG1864- manager.c: Examining event:
Event: Hangup
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002c0;2
Uniqueid: 1500497009.1985
CallerIDNum: 4502391367
CallerIDName: V7191643290000000536
ConnectedLineNum: 4502391367
ConnectedLineName: V7191643290000000536
AccountCode:
Cause: 21
Cause-txt: Call Rejected


Jul 19 16:43:30- DEBUG1864- manager.c: Examining event:
Event: Hangup
Privilege: call,all
Channel: Local/70571819351438XXXXXXX default-000002c0;1
Uniqueid: 1500497009.1984
CallerIDNum: 4502391367
CallerIDName: V7191643290000000536
ConnectedLineNum: 4502391367
ConnectedLineName: V7191643290000000536
AccountCode:
Cause: 21
Cause-txt: Call Rejected
Architecture: x86_64 CPU(s): 2 CPU family: 15 Model: 6
Model name: Intel(R) Pentium(R) 4 CPU 3.20GHz
Linux version 4.1.39-56-default (geeko@buildhost) (gcc version 4.8.5 (SUSE Linux) )
VICIDIAL VERSION: 2.14-831a BUILD: 211012-0934
wjohnson133
 
Posts: 45
Joined: Wed Mar 15, 2017 2:00 pm
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Re: Intermittent "403 FORBIDDEN" during outbound auto dialli

Postby mflorell » Wed Jul 19, 2017 6:38 pm

Have you asked your carrier?

Have you tried a different carrier?
mflorell
Site Admin
 
Posts: 18387
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida
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