For some reason I’m getting the “Spawn extension (dialout, 7865551111, 4) exited non-zero” error after the dial command. The error doesn’t occur when the ringtimeout is set to 20 seconds. When it’s set to 30 seconds it never jumps to the 5th priority. I’m using ver. 1.4
extensions.conf:
[dialout]
exten => _X.,1,Set(CALLERID(number)=${callerid})
exten => _X.,2,Set(CALLERID(name)=${calleridname})
exten => _X.,3,Set(${phone}=${EXTEN})
exten => _X.,4,Dial(SIP/1${EXTEN}@vitel-outbound,30,r) ;Where the call drops off
exten => _X.,5,Set(CALLSTATUS=${dialstatus})
exten => _X.,6,Goto(s-${DIALSTATUS},1)
sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
nat=yes
externip= 111.111.111.111
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
register => username:********@inbound23.vitelity.net:5060
[vitel-inbound]
type=friend
dtmfmode=auto
host=inbound23.vitelity.net
context=inbound
username=username
secret=********
allow=all
insecure=very
canreinvite=no
[vitel-outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
username=username
fromuser=username
trustrpid=yes
sendrpid=yes
secret=********
allow=all
canreinvite=no
CLI (with SIP Debug):
[Jan 14 20:19:02] VERBOSE[10467] logger.c: Reliably Transmitting (NAT) to 222.222.222.222:5060:
OPTIONS sip:222.222.222.222 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK288fd1ff;rport
From: “asterisk” sip:asterisk@111.111.111.111;tag=as478b93f6
To: sip:222.222.222.222
Contact: sip:asterisk@111.111.111.111
Call-ID: 46a117ec32782a030100cbfe3180e13f@111.111.111.111
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 14 Jan 2010 20:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[Jan 14 20:19:03] VERBOSE[10467] logger.c:
<— SIP read from 222.222.222.222:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK288fd1ff;rport=5060
From: “asterisk” sip:asterisk@111.111.111.111;tag=as478b93f6
To: sip:222.222.222.222;tag=9c6a9fdfd4d16ebaa52f34c4c528cbe5.dda4
Call-ID: 46a117ec32782a030100cbfe3180e13f@111.111.111.111
CSeq: 102 OPTIONS
Accept: /
Accept-Encoding:
Accept-Language: en
Support:
Server: Sip EXpress router (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 192.168.1.50:5060 “Noisy feedback tells: pid=27175 req_src_ip=111.111.111.111 req_src_port=5060 in_uri=sip:222.222.222.222 out_uri=sip:222.222.222.222 via_cnt==1”
<------------->
[Jan 14 20:19:03] VERBOSE[10467] logger.c: — (13 headers 0 lines) —
[Jan 14 20:19:03] VERBOSE[10467] logger.c: Really destroying SIP dialog ‘46a117ec32782a030100cbfe3180e13f@111.111.111.111’ Method: OPTIONS
[Jan 14 20:19:06] VERBOSE[22102] logger.c: == Manager ‘devtel’ logged off from 64.150.188.69
[Jan 14 20:19:06] VERBOSE[22229] logger.c: == Parsing ‘/etc/asterisk/manager.conf’: [Jan 14 20:19:06] VERBOSE[22229] logger.c: Found
[Jan 14 20:19:06] VERBOSE[22229] logger.c: == Manager ‘devtel’ logged on from 64.150.188.69
[Jan 14 20:19:06] VERBOSE[22230] logger.c: – Executing [7865551111@dialout:1] Set(“Local/7865551111@dialout-b4d7,2”, “CALLERID(number)=7865332227”) in new stack
[Jan 14 20:19:06] VERBOSE[22230] logger.c: – Executing [7865551111@dialout:2] Set(“Local/7865551111@dialout-b4d7,2”, “CALLERID(name)=Approval Notifications”) in new stack
[Jan 14 20:19:06] VERBOSE[22230] logger.c: – Executing [7865551111@dialout:3] Set(“Local/7865551111@dialout-b4d7,2”, “7865551111=7865551111”) in new stack
[Jan 14 20:19:06] VERBOSE[22230] logger.c: – Executing [7865551111@dialout:4] Dial(“Local/7865551111@dialout-b4d7,2”, “SIP/17865551111@vitel-outbound|30|r”) in new stack
[Jan 14 20:19:06] VERBOSE[22230] logger.c: Audio is at 111.111.111.111 port 20988
[Jan 14 20:19:06] VERBOSE[22230] logger.c: Adding codec 0x40 (slin) to SDP
[Jan 14 20:19:06] VERBOSE[22230] logger.c: Adding codec 0x4 (ulaw) to SDP
[Jan 14 20:19:06] VERBOSE[22230] logger.c: Adding codec 0x8 (alaw) to SDP
[Jan 14 20:19:06] VERBOSE[22230] logger.c: Adding codec 0x2 (gsm) to SDP
[Jan 14 20:19:06] VERBOSE[22230] logger.c: Adding codec 0x10 (g726aal2) to SDP
[Jan 14 20:19:06] VERBOSE[22230] logger.c: Adding codec 0x20 (adpcm) to SDP
[Jan 14 20:19:06] VERBOSE[22230] logger.c: Adding codec 0x80 (lpc10) to SDP
[Jan 14 20:19:06] VERBOSE[22230] logger.c: Adding codec 0x800 (g726) to SDP
[Jan 14 20:19:06] VERBOSE[22230] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jan 14 20:19:06] VERBOSE[22230] logger.c: Reliably Transmitting (NAT) to 64.2.142.215:5060:
INVITE sip:17865551111@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK50b91de1;rport
From: “Approval Notifications” sip:username@111.111.111.111;tag=as5f575c9a
To: sip:17865551111@outbound.vitelity.net
Contact: sip:username@111.111.111.111
Call-ID: 489085fb7595dd0758785bc90f4caa4b@111.111.111.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “Approval Notifications” sip:7865332227@111.111.111.111;privacy=off;screen=no
Date: Thu, 14 Jan 2010 20:19:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 430
v=0
o=root 10434 10434 IN IP4 111.111.111.111
s=session
c=IN IP4 111.111.111.111
t=0 0
m=audio 20988 RTP/AVP 10 0 8 3 112 5 7 111 101
a=rtpmap:10 L16/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[Jan 14 20:19:06] VERBOSE[22230] logger.c: – Called 17865551111@vitel-outbound
[Jan 14 20:19:06] VERBOSE[10467] logger.c:
<— SIP read from 64.2.142.215:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK50b91de1;received=111.111.111.111;rport=5060
From: “Approval Notifications” sip:username@111.111.111.111;tag=as5f575c9a
To: sip:17865551111@outbound.vitelity.net;tag=as1b183ece
Call-ID: 489085fb7595dd0758785bc90f4caa4b@111.111.111.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0ceb0e5e"
Content-Length: 0
<------------->
[Jan 14 20:19:06] VERBOSE[10467] logger.c: — (11 headers 0 lines) —
[Jan 14 20:19:06] VERBOSE[10467] logger.c: Transmitting (NAT) to 64.2.142.215:5060:
ACK sip:17865551111@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK50b91de1;rport
From: “Approval Notifications” sip:username@111.111.111.111;tag=as5f575c9a
To: sip:17865551111@outbound.vitelity.net;tag=as1b183ece
Contact: sip:username@111.111.111.111
Call-ID: 489085fb7595dd0758785bc90f4caa4b@111.111.111.111
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “Approval Notifications” sip:7865332227@111.111.111.111;privacy=off;screen=no
Content-Length: 0
[Jan 14 20:19:06] VERBOSE[10467] logger.c: Audio is at 111.111.111.111 port 20988
[Jan 14 20:19:06] VERBOSE[10467] logger.c: Adding codec 0x40 (slin) to SDP
[Jan 14 20:19:06] VERBOSE[10467] logger.c: Adding codec 0x4 (ulaw) to SDP
[Jan 14 20:19:06] VERBOSE[10467] logger.c: Adding codec 0x8 (alaw) to SDP
[Jan 14 20:19:06] VERBOSE[10467] logger.c: Adding codec 0x2 (gsm) to SDP
[Jan 14 20:19:06] VERBOSE[10467] logger.c: Adding codec 0x10 (g726aal2) to SDP
[Jan 14 20:19:06] VERBOSE[10467] logger.c: Adding codec 0x20 (adpcm) to SDP
[Jan 14 20:19:06] VERBOSE[10467] logger.c: Adding codec 0x80 (lpc10) to SDP
[Jan 14 20:19:06] VERBOSE[10467] logger.c: Adding codec 0x800 (g726) to SDP
[Jan 14 20:19:06] VERBOSE[10467] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jan 14 20:19:06] VERBOSE[10467] logger.c: Reliably Transmitting (NAT) to 64.2.142.215:5060:
INVITE sip:17865551111@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK3be471a1;rport
From: “Approval Notifications” sip:username@111.111.111.111;tag=as5f575c9a
To: sip:17865551111@outbound.vitelity.net
Contact: sip:username@111.111.111.111
Call-ID: 489085fb7595dd0758785bc90f4caa4b@111.111.111.111
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “Approval Notifications” sip:7865332227@111.111.111.111;privacy=off;screen=no
Proxy-Authorization: Digest username=“username”, realm=“asterisk”, algorithm=MD5, uri="sip:17865551111@outbound.vitelity.net", nonce=“0ceb0e5e”, response="a0426b8c322ce99d7d6790b92961c773"
Date: Thu, 14 Jan 2010 20:19:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 430
v=0
o=root 10434 10435 IN IP4 111.111.111.111
s=session
c=IN IP4 111.111.111.111
t=0 0
m=audio 20988 RTP/AVP 10 0 8 3 112 5 7 111 101
a=rtpmap:10 L16/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[Jan 14 20:19:06] VERBOSE[10467] logger.c:
<— SIP read from 64.2.142.215:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK3be471a1;received=111.111.111.111;rport=5060
From: “Approval Notifications” sip:username@111.111.111.111;tag=as5f575c9a
To: sip:17865551111@outbound.vitelity.net
Call-ID: 489085fb7595dd0758785bc90f4caa4b@111.111.111.111
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:17865551111@64.2.142.215
Content-Length: 0
<------------->
[Jan 14 20:19:06] VERBOSE[10467] logger.c: — (11 headers 0 lines) —
[Jan 14 20:19:08] VERBOSE[10467] logger.c: Really destroying SIP dialog ‘3e102b6329afd44322e4980c7fe1363c@127.0.0.1’ Method: REGISTER
[Jan 14 20:19:09] VERBOSE[10467] logger.c:
<— SIP read from 64.2.142.215:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK3be471a1;received=111.111.111.111;rport=5060
From: “Approval Notifications” sip:username@111.111.111.111;tag=as5f575c9a
To: sip:17865551111@outbound.vitelity.net;tag=as2b350db0
Call-ID: 489085fb7595dd0758785bc90f4caa4b@111.111.111.111
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:17865551111@64.2.142.215
Content-Length: 0
<------------->
[Jan 14 20:19:09] VERBOSE[10467] logger.c: — (11 headers 0 lines) —
[Jan 14 20:19:09] VERBOSE[22230] logger.c: – SIP/vitel-outbound-0834e868 is ringing
[Jan 14 20:19:21] NOTICE[10467] chan_sip.c: – Re-registration for username@inbound23.vitelity.net
[Jan 14 20:19:21] DEBUG[10467] chan_sip.c: >>> Re-using Auth data for username@inbound23.vitelity.net
[Jan 14 20:19:21] VERBOSE[10467] logger.c: REGISTER 13 headers, 0 lines
[Jan 14 20:19:21] VERBOSE[10467] logger.c: Reliably Transmitting (NAT) to 66.241.96.96:5060:
REGISTER sip:inbound23.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK38ab86bd;rport
From: sip:username@inbound23.vitelity.net;tag=as6799133e
To: sip:username@inbound23.vitelity.net
Call-ID: 3e102b6329afd44322e4980c7fe1363c@127.0.0.1
CSeq: 12619 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“username”, realm=“asterisk”, algorithm=MD5, uri=“sip:inbound23.vitelity.net”, nonce=“72029ef7”, response="eeb33b73344eb117861804b60717ebd0"
Expires: 120
Contact: sip:s@111.111.111.111
Event: registration
Content-Length: 0
[Jan 14 20:19:21] VERBOSE[10467] logger.c:
<— SIP read from 66.241.96.96:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK38ab86bd;received=111.111.111.111;rport=5060
From: sip:username@inbound23.vitelity.net;tag=as6799133e
To: sip:username@inbound23.vitelity.net
Call-ID: 3e102b6329afd44322e4980c7fe1363c@127.0.0.1
CSeq: 12619 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------->
[Jan 14 20:19:21] VERBOSE[10467] logger.c: — (10 headers 0 lines) —
[Jan 14 20:19:21] VERBOSE[10467] logger.c:
<— SIP read from 66.241.96.96:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK38ab86bd;received=111.111.111.111;rport=5060
From: sip:username@inbound23.vitelity.net;tag=as6799133e
To: sip:username@inbound23.vitelity.net;tag=as7a5e39eb
Call-ID: 3e102b6329afd44322e4980c7fe1363c@127.0.0.1
CSeq: 12619 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="56e2dc92"
Content-Length: 0
<------------->
[Jan 14 20:19:21] VERBOSE[10467] logger.c: — (11 headers 0 lines) —
[Jan 14 20:19:21] VERBOSE[10467] logger.c: Responding to challenge, registration to domain/host name inbound23.vitelity.net
[Jan 14 20:19:21] VERBOSE[10467] logger.c: REGISTER 13 headers, 0 lines
[Jan 14 20:19:21] VERBOSE[10467] logger.c: Reliably Transmitting (NAT) to 66.241.96.96:5060:
REGISTER sip:inbound23.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK7b632152;rport
From: sip:username@inbound23.vitelity.net;tag=as7ed8df9e
To: sip:username@inbound23.vitelity.net
Call-ID: 3e102b6329afd44322e4980c7fe1363c@127.0.0.1
CSeq: 12620 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“username”, realm=“asterisk”, algorithm=MD5, uri=“sip:inbound23.vitelity.net”, nonce=“56e2dc92”, response="18a53c9e1ca12d2684de402e6aac1594"
Expires: 120
Contact: sip:s@111.111.111.111
Event: registration
Content-Length: 0
[Jan 14 20:19:21] VERBOSE[10467] logger.c:
<— SIP read from 66.241.96.96:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK7b632152;received=111.111.111.111;rport=5060
From: sip:username@inbound23.vitelity.net;tag=as7ed8df9e
To: sip:username@inbound23.vitelity.net
Call-ID: 3e102b6329afd44322e4980c7fe1363c@127.0.0.1
CSeq: 12620 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------->
[Jan 14 20:19:21] VERBOSE[10467] logger.c: — (10 headers 0 lines) —
[Jan 14 20:19:21] VERBOSE[10467] logger.c:
<— SIP read from 66.241.96.96:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK7b632152;received=111.111.111.111;rport=5060
From: sip:username@inbound23.vitelity.net;tag=as7ed8df9e
To: sip:username@inbound23.vitelity.net;tag=as7a5e39eb
Call-ID: 3e102b6329afd44322e4980c7fe1363c@127.0.0.1
CSeq: 12620 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: sip:s@111.111.111.111;expires=60
Date: Thu, 14 Jan 2010 20:23:36 GMT
Content-Length: 0
<------------->
[Jan 14 20:19:21] VERBOSE[10467] logger.c: — (13 headers 0 lines) —
[Jan 14 20:19:21] VERBOSE[10467] logger.c: Scheduling destruction of SIP dialog ‘3e102b6329afd44322e4980c7fe1363c@127.0.0.1’ in 32000 ms (Method: REGISTER)
[Jan 14 20:19:21] NOTICE[10467] chan_sip.c: Outbound Registration: Expiry for inbound23.vitelity.net is 60 sec (Scheduling reregistration in 45 s)
[Jan 14 20:19:36] VERBOSE[22230] logger.c: Scheduling destruction of SIP dialog ‘489085fb7595dd0758785bc90f4caa4b@111.111.111.111’ in 32000 ms (Method: INVITE)
[Jan 14 20:19:36] VERBOSE[22230] logger.c: Reliably Transmitting (NAT) to 64.2.142.215:5060:
CANCEL sip:17865551111@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK3be471a1;rport
From: “Approval Notifications” sip:username@111.111.111.111;tag=as5f575c9a
To: sip:17865551111@outbound.vitelity.net
Call-ID: 489085fb7595dd0758785bc90f4caa4b@111.111.111.111
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “Approval Notifications” sip:7865332227@111.111.111.111;privacy=off;screen=no
Content-Length: 0
[Jan 14 20:19:36] VERBOSE[22230] logger.c: Scheduling destruction of SIP dialog ‘489085fb7595dd0758785bc90f4caa4b@111.111.111.111’ in 32000 ms (Method: INVITE)
[Jan 14 20:19:36] VERBOSE[22230] logger.c: == Spawn extension (dialout, 7865551111, 4) exited non-zero on ‘Local/7865551111@dialout-b4d7,2’
[Jan 14 20:19:36] VERBOSE[10467] logger.c:
<— SIP read from 64.2.142.215:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK3be471a1;received=111.111.111.111;rport=5060
From: “Approval Notifications” sip:username@111.111.111.111;tag=as5f575c9a
To: sip:17865551111@outbound.vitelity.net;tag=as2b350db0
Call-ID: 489085fb7595dd0758785bc90f4caa4b@111.111.111.111
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------->
[Jan 14 20:19:36] VERBOSE[10467] logger.c: — (10 headers 0 lines) —
[Jan 14 20:19:36] VERBOSE[10467] logger.c: Transmitting (NAT) to 64.2.142.215:5060:
ACK sip:17865551111@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK3be471a1;rport
From: “Approval Notifications” sip:username@111.111.111.111;tag=as5f575c9a
To: sip:17865551111@outbound.vitelity.net;tag=as2b350db0
Contact: sip:username@111.111.111.111
Call-ID: 489085fb7595dd0758785bc90f4caa4b@111.111.111.111
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “Approval Notifications” sip:7865332227@111.111.111.111;privacy=off;screen=no
Content-Length: 0
[Jan 14 20:19:36] VERBOSE[10467] logger.c:
<— SIP read from 64.2.142.215:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK3be471a1;received=111.111.111.111;rport=5060
From: “Approval Notifications” sip:username@111.111.111.111;tag=as5f575c9a
To: sip:17865551111@outbound.vitelity.net;tag=as2b350db0
Call-ID: 489085fb7595dd0758785bc90f4caa4b@111.111.111.111
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:17865551111@64.2.142.215
Content-Length: 0
<------------->
[Jan 14 20:19:36] VERBOSE[10467] logger.c: — (11 headers 0 lines) —
[Jan 14 20:19:36] VERBOSE[10467] logger.c: Really destroying SIP dialog ‘489085fb7595dd0758785bc90f4caa4b@111.111.111.111’ Method: INVITE
[Jan 14 20:19:53] VERBOSE[10467] logger.c: Really destroying SIP dialog ‘3e102b6329afd44322e4980c7fe1363c@127.0.0.1’ Method: REGISTER