Call goes fine, then just hangs up. I’ve captured the SIP log and also included my super simple dialplan. I have my ports 5060 and 10000-20000 open as well…
Dialplan:
[incoming]
exten = _.,1,Answer()
same => n,Set(CALLERID(num)=prohib)
same = n,Wait(1)
same = n,Dial(SIP/callcentric1/xxxxxxxxxx)SIP log:
SIP Debugging enabled
<--- SIP read from UDP:210.3.88.146:1218 --->
<------------->
Reliably Transmitting (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK35b9f64f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as01ae64cf
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 754750a330fd96887e7cb1b321975cb5@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Retransmitting #1 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK35b9f64f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as01ae64cf
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 754750a330fd96887e7cb1b321975cb5@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Retransmitting #2 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK35b9f64f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as01ae64cf
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 754750a330fd96887e7cb1b321975cb5@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Retransmitting #3 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK35b9f64f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as01ae64cf
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 754750a330fd96887e7cb1b321975cb5@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Retransmitting #4 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK35b9f64f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as01ae64cf
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 754750a330fd96887e7cb1b321975cb5@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Really destroying SIP dialog '754750a330fd96887e7cb1b321975cb5@192.168.100.178:5060' Method: OPTIONS
<--- SIP read from UDP:204.11.192.132:5068 --->
INVITE sip:6001@192.168.100.178 SIP/2.0
Via: SIP/2.0/UDP 204.11.192.132:5068;branch=z9hG4bK4967c4e0;rport
Max-Forwards: 70
From: "36981900" <sip:36981900@204.11.192.132:5068>;tag=as539ced8a
To: <sip:6001@192.168.100.178>
Contact: <sip:36981900@204.11.192.132:5068>
Call-ID: 079db49e71db7a616aad012616de9544@204.11.192.132
CSeq: 102 INVITE
User-Agent: Callcentric-b2b
Date: Wed, 09 Apr 2014 08:20:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 323
v=0
o=root 455728958 455728958 IN IP4 204.11.192.132
s=session
c=IN IP4 204.11.192.132
t=0 0
m=audio 13348 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 15 lines) ---
Sending to 204.11.192.132:5068 (no NAT)
Sending to 204.11.192.132:5068 (no NAT)
Using INVITE request as basis request - 079db49e71db7a616aad012616de9544@204.11.192.132
No matching peer for '36981900' from '204.11.192.132:5068'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 204.11.192.132:13348
Looking for 6001 in incoming (domain 192.168.100.178)
list_route: route/path hop: <sip:36981900@204.11.192.132:5068>
<--- Transmitting (no NAT) to 204.11.192.132:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.132:5068;branch=z9hG4bK4967c4e0;received=204.11.192.132;rport=5068
From: "36981900" <sip:36981900@204.11.192.132:5068>;tag=as539ced8a
To: <sip:6001@192.168.100.178>
Call-ID: 079db49e71db7a616aad012616de9544@204.11.192.132
CSeq: 102 INVITE
Server: Asterisk PBX 12.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:6001@192.168.100.178:5060>
Content-Length: 0
<------------>
Audio is at 13810
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 204.11.192.132:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.132:5068;branch=z9hG4bK4967c4e0;received=204.11.192.132;rport=5068
From: "36981900" <sip:36981900@204.11.192.132:5068>;tag=as539ced8a
To: <sip:6001@192.168.100.178>;tag=as3e20d47d
Call-ID: 079db49e71db7a616aad012616de9544@204.11.192.132
CSeq: 102 INVITE
Server: Asterisk PBX 12.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:6001@192.168.100.178:5060>
Content-Type: application/sdp
Content-Length: 305
v=0
o=root 126666816 126666816 IN IP4 180.150.153.178
s=Asterisk PBX 12.1.1
c=IN IP4 180.150.153.178
t=0 0
m=audio 13810 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:230
a=sendrecv
<------------>
<--- SIP read from UDP:204.11.192.132:5068 --->
ACK sip:6001@192.168.100.178:5060 SIP/2.0
Via: SIP/2.0/UDP 204.11.192.132:5068;branch=z9hG4bK59ea0d93;rport
Max-Forwards: 70
From: "36981900" <sip:36981900@204.11.192.132:5068>;tag=as539ced8a
To: <sip:6001@192.168.100.178>;tag=as3e20d47d
Contact: <sip:36981900@204.11.192.132:5068>
Call-ID: 079db49e71db7a616aad012616de9544@204.11.192.132
CSeq: 102 ACK
User-Agent: Callcentric-b2b
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Audio is at 14156
Adding codec 100008 (g729) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 204.11.192.170:5080:
INVITE sip:01185264664514@callcentric.com SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK6b4ad457
Max-Forwards: 70
From: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
To: <sip:01185264664514@callcentric.com>
Contact: <sip:17772189338@192.168.100.178:5060>
Call-ID: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 420
v=0
o=root 353748462 353748462 IN IP4 180.150.153.178
s=Asterisk PBX 12.1.1
c=IN IP4 180.150.153.178
t=0 0
m=audio 14156 RTP/AVP 18 97 0 8 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:30
a=sendrecv
---
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK6b4ad457;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="5811bce3b8af6b556d7b795e42931796", opaque="", stale=TRUE, algorithm=MD5
l: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 204.11.192.170:5080:
ACK sip:01185264664514@callcentric.com SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK6b4ad457
Max-Forwards: 70
From: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
To: <sip:01185264664514@callcentric.com>
Contact: <sip:17772189338@192.168.100.178:5060>
Call-ID: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.1.1
Content-Length: 0
---
Audio is at 14156
Adding codec 100008 (g729) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 204.11.192.170:5080:
INVITE sip:01185264664514@callcentric.com SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK1ee4b2f8
Max-Forwards: 70
From: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
To: <sip:01185264664514@callcentric.com>
Contact: <sip:17772189338@192.168.100.178:5060>
Call-ID: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX 12.1.1
Proxy-Authorization: Digest username="17772189338", realm="callcentric.com", algorithm=MD5, uri="sip:sip:callcentric.com", nonce="5811bce3b8af6b556d7b795e42931796", response="7050148dd4e3d4dea9c6489b95327a31"
Date: Wed, 09 Apr 2014 08:20:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 420
v=0
o=root 353748462 353748463 IN IP4 180.150.153.178
s=Asterisk PBX 12.1.1
c=IN IP4 180.150.153.178
t=0 0
m=audio 14156 RTP/AVP 18 97 0 8 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:30
a=sendrecv
---
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK1ee4b2f8;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 103 INVITE
l: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK1ee4b2f8;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 103 INVITE
l: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK1ee4b2f8;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 103 INVITE
m: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
c: application/sdp
l: 289
v=0
o=NexTone-MSW 10666202 10666202 IN IP4 204.11.192.170
s=sip call
c=IN IP4 204.11.192.170
t=0 0
m=audio 63466 RTP/AVP 18 101
a=maxptime:30
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
a=setup:actpass
<------------->
--- (9 headers 13 lines) ---
list_route: route/path hop: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
Found RTP audio format 18
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found audio description format G729 for ID 18
Capabilities: us - (gsm|ulaw|alaw|g729|ilbc), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 204.11.192.170:63466
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK1ee4b2f8;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 103 INVITE
m: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
c: application/sdp
l: 289
v=0
o=NexTone-MSW 10666202 10666202 IN IP4 204.11.192.170
s=sip call
c=IN IP4 204.11.192.170
t=0 0
m=audio 63466 RTP/AVP 18 101
a=maxptime:30
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
a=setup:actpass
<------------->
--- (9 headers 13 lines) ---
list_route: route/path hop: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK1ee4b2f8;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 103 INVITE
m: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
c: application/sdp
l: 289
v=0
o=NexTone-MSW 10666202 10666202 IN IP4 204.11.192.170
s=sip call
c=IN IP4 204.11.192.170
t=0 0
m=audio 63466 RTP/AVP 18 101
a=maxptime:30
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
a=setup:actpass
<------------->
--- (9 headers 13 lines) ---
list_route: route/path hop: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK1ee4b2f8;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 103 INVITE
m: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
c: application/sdp
l: 289
v=0
o=NexTone-MSW 10666202 10666202 IN IP4 204.11.192.170
s=sip call
c=IN IP4 204.11.192.170
t=0 0
m=audio 63466 RTP/AVP 18 101
a=maxptime:30
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
a=setup:actpass
<------------->
--- (9 headers 13 lines) ---
list_route: route/path hop: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 180 Ringing
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK1ee4b2f8;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 103 INVITE
m: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
l: 0
<------------->
--- (8 headers 0 lines) ---
list_route: route/path hop: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
[Apr 9 16:20:32] WARNING[23910][C-00000000]: channel.c:5332 set_format: Unable to find a codec translation path from (g729) to (slin)
[Apr 9 16:20:32] WARNING[23910][C-00000000]: indications.c:157 playtones_alloc: Unable to set 'SIP/204.11.192.132-00000000' to signed linear format (write)
[Apr 9 16:20:32] WARNING[23910][C-00000000]: channel.c:4651 ast_indicate_data: Unable to handle indication 3 for 'SIP/204.11.192.132-00000000'
Reliably Transmitting (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK2575f8fc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as4b334f9d
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 605a3f52259c8c917e4693aa2dfdfca2@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 180 Ringing
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK1ee4b2f8;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 103 INVITE
m: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
l: 0
<------------->
--- (8 headers 0 lines) ---
list_route: route/path hop: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
Retransmitting #1 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK2575f8fc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as4b334f9d
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 605a3f52259c8c917e4693aa2dfdfca2@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 180 Ringing
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK1ee4b2f8;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 103 INVITE
m: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
l: 0
<------------->
--- (8 headers 0 lines) ---
list_route: route/path hop: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK1ee4b2f8;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 103 INVITE
m: <sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp>
c: application/sdp
l: 289
v=0
o=NexTone-MSW 10666202 10666202 IN IP4 204.11.192.170
s=sip call
c=IN IP4 204.11.192.170
t=0 0
m=audio 63466 RTP/AVP 18 101
a=maxptime:30
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
a=setup:actpass
<------------->
--- (9 headers 13 lines) ---
list_route: route/path hop: <sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp>
set_destination: Parsing <sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp> for address/port to send to
set_destination: set destination to 204.11.192.170:5080
Transmitting (no NAT) to 204.11.192.170:5080:
ACK sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK66038e6b
Max-Forwards: 70
From: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
To: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
Contact: <sip:17772189338@192.168.100.178:5060>
Call-ID: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 103 ACK
User-Agent: Asterisk PBX 12.1.1
Content-Length: 0
---
set_destination: Parsing <sip:36981900@204.11.192.132:5068> for address/port to send to
set_destination: set destination to 204.11.192.132:5068
Audio is at 13810
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 204.11.192.132:5068:
INVITE sip:36981900@204.11.192.132:5068 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK217638ba;rport
Max-Forwards: 70
From: <sip:6001@192.168.100.178>;tag=as3e20d47d
To: "36981900" <sip:36981900@204.11.192.132:5068>;tag=as539ced8a
Contact: <sip:6001@192.168.100.178:5060>
Call-ID: 079db49e71db7a616aad012616de9544@204.11.192.132
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 303
v=0
o=root 126666816 126666817 IN IP4 204.11.192.170
s=Asterisk PBX 12.1.1
c=IN IP4 204.11.192.170
t=0 0
m=audio 63466 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:230
a=sendrecv
---
set_destination: Parsing <sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp> for address/port to send to
set_destination: set destination to 204.11.192.170:5080
Audio is at 14156
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 204.11.192.170:5080:
INVITE sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK29c638df
Max-Forwards: 70
From: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
To: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
Contact: <sip:17772189338@192.168.100.178:5060>
Call-ID: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 104 INVITE
User-Agent: Asterisk PBX 12.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 303
v=0
o=root 353748462 353748464 IN IP4 204.11.192.132
s=Asterisk PBX 12.1.1
c=IN IP4 204.11.192.132
t=0 0
m=audio 13348 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:230
a=sendrecv
---
Retransmitting #2 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK2575f8fc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as4b334f9d
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 605a3f52259c8c917e4693aa2dfdfca2@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
<--- SIP read from UDP:204.11.192.132:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK217638ba;received=192.168.100.178;rport=5060
From: <sip:6001@192.168.100.178>;tag=as3e20d47d
To: "36981900" <sip:36981900@204.11.192.132:5068>;tag=as539ced8a
Call-ID: 079db49e71db7a616aad012616de9544@204.11.192.132
CSeq: 102 INVITE
Server: Callcentric-b2b
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:36981900@204.11.192.132:5068>
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:204.11.192.132:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK217638ba;received=192.168.100.178;rport=5060
From: <sip:6001@192.168.100.178>;tag=as3e20d47d
To: "36981900" <sip:36981900@204.11.192.132:5068>;tag=as539ced8a
Call-ID: 079db49e71db7a616aad012616de9544@204.11.192.132
CSeq: 102 INVITE
Server: Callcentric-b2b
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:36981900@204.11.192.132:5068>
Content-Type: application/sdp
Content-Length: 275
v=0
o=root 455728958 455728959 IN IP4 204.11.192.132
s=session
c=IN IP4 204.11.192.132
t=0 0
m=audio 13348 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g729), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 204.11.192.132:13348
set_destination: Parsing <sip:36981900@204.11.192.132:5068> for address/port to send to
set_destination: set destination to 204.11.192.132:5068
Transmitting (no NAT) to 204.11.192.132:5068:
ACK sip:36981900@204.11.192.132:5068 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK117290fc;rport
Max-Forwards: 70
From: <sip:6001@192.168.100.178>;tag=as3e20d47d
To: "36981900" <sip:36981900@204.11.192.132:5068>;tag=as539ced8a
Contact: <sip:6001@192.168.100.178:5060>
Call-ID: 079db49e71db7a616aad012616de9544@204.11.192.132
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.1.1
Content-Length: 0
---
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK29c638df;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 104 INVITE
l: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK29c638df;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 104 INVITE
m: <sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp>
c: application/sdp
l: 289
v=0
o=NexTone-MSW 10666202 10666203 IN IP4 204.11.192.170
s=sip call
c=IN IP4 204.11.192.170
t=0 0
m=audio 63466 RTP/AVP 18 101
a=maxptime:40
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
a=setup:actpass
<------------->
--- (9 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found audio description format G729 for ID 18
Capabilities: us - (gsm|ulaw|alaw|g729|ilbc), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 204.11.192.170:63466
set_destination: Parsing <sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp> for address/port to send to
set_destination: set destination to 204.11.192.170:5080
Transmitting (no NAT) to 204.11.192.170:5080:
ACK sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK313398c2
Max-Forwards: 70
From: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
To: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
Contact: <sip:17772189338@192.168.100.178:5060>
Call-ID: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 104 ACK
User-Agent: Asterisk PBX 12.1.1
Content-Length: 0
---
set_destination: Parsing <sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp> for address/port to send to
set_destination: set destination to 204.11.192.170:5080
Audio is at 14156
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 204.11.192.170:5080:
INVITE sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK3f75b6cd
Max-Forwards: 70
From: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
To: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
Contact: <sip:17772189338@192.168.100.178:5060>
Call-ID: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 105 INVITE
User-Agent: Asterisk PBX 12.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 303
v=0
o=root 353748462 353748465 IN IP4 204.11.192.132
s=Asterisk PBX 12.1.1
c=IN IP4 204.11.192.132
t=0 0
m=audio 13348 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:230
a=sendrecv
---
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK3f75b6cd;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 105 INVITE
l: 0
<------------->
--- (7 headers 0 lines) ---
Retransmitting #3 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK2575f8fc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as4b334f9d
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 605a3f52259c8c917e4693aa2dfdfca2@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
[Apr 9 16:20:36] NOTICE[23123]: chan_sip.c:15261 sip_reregister: -- Re-registration for 17772189338@callcentric.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 204.11.192.159:5060:
REGISTER sip:callcentric.com SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK3618b6dd
Max-Forwards: 70
From: <sip:17772189338@callcentric.com>;tag=as5664e8d9
To: <sip:17772189338@callcentric.com>
Call-ID: 6a8710066aebe730083677ea41f91355@192.168.100.178
CSeq: 104 REGISTER
Supported: replaces
User-Agent: Asterisk PBX 12.1.1
Authorization: Digest username="17772189338", realm="callcentric.com", algorithm=MD5, uri="sip:sip:callcentric.com", nonce="cd4b465f3ae025393f21604d644014dd", response="165096e88c35294147b9da9a39740e52"
Expires: 120
Contact: <sip:s@192.168.100.178:5060>
Content-Length: 0
---
Retransmitting #4 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK2575f8fc
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as4b334f9d
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 605a3f52259c8c917e4693aa2dfdfca2@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Really destroying SIP dialog '605a3f52259c8c917e4693aa2dfdfca2@192.168.100.178:5060' Method: OPTIONS
<--- SIP read from UDP:204.11.192.159:5060 --->
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK3618b6dd
f: <sip:17772189338@callcentric.com>;tag=as5664e8d9
t: <sip:17772189338@callcentric.com>
i: 6a8710066aebe730083677ea41f91355@192.168.100.178
CSeq: 104 REGISTER
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="da9a1190e61cc2a03b71f6cd96972180", opaque="", stale=TRUE, algorithm=MD5
l: 0
<------------->
--- (8 headers 0 lines) ---
Responding to challenge, registration to domain/host name callcentric.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 204.11.192.159:5060:
REGISTER sip:callcentric.com SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK544d1c3b
Max-Forwards: 70
From: <sip:17772189338@callcentric.com>;tag=as5664e8d9
To: <sip:17772189338@callcentric.com>
Call-ID: 6a8710066aebe730083677ea41f91355@192.168.100.178
CSeq: 105 REGISTER
Supported: replaces
User-Agent: Asterisk PBX 12.1.1
Proxy-Authorization: Digest username="17772189338", realm="callcentric.com", algorithm=MD5, uri="sip:sip:callcentric.com", nonce="da9a1190e61cc2a03b71f6cd96972180", response="21a1ff9348aed7d6af2da453c69dc273"
Expires: 120
Contact: <sip:s@192.168.100.178:5060>
Content-Length: 0
---
<--- SIP read from UDP:204.11.192.159:5060 --->
SIP/2.0 200 Ok
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK544d1c3b
f: <sip:17772189338@callcentric.com>;tag=as5664e8d9
t: <sip:17772189338@callcentric.com>
i: 6a8710066aebe730083677ea41f91355@192.168.100.178
CSeq: 105 REGISTER
m: <sip:s@192.168.100.178:5060>;expires=60
l: 0
<------------->
--- (8 headers 0 lines) ---
[Apr 9 16:20:37] NOTICE[23123]: chan_sip.c:24015 handle_response_register: Outbound Registration: Expiry for callcentric.com is 60 sec (Scheduling reregistration in 45 s)
Really destroying SIP dialog '6a8710066aebe730083677ea41f91355@192.168.100.178' Method: REGISTER
<--- SIP read from UDP:204.11.192.170:5080 --->
SIP/2.0 504 Server Timeout
v: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK3f75b6cd;rport=5060;received=180.150.153.178
f: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
t: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
i: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 105 INVITE
m: <sip:9b97f6287d3fa7e813ab56937b6c8ce3@204.11.192.170:5080;transport=udp>
l: 0
<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp> for address/port to send to
set_destination: set destination to 204.11.192.170:5080
Transmitting (no NAT) to 204.11.192.170:5080:
ACK sip:bb4dde4855395ebc0199061f8e6e734a@204.11.192.170:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK3f75b6cd
Max-Forwards: 70
From: "36981900" <sip:17772189338@callcentric.com>;tag=as348479d9
To: <sip:01185264664514@callcentric.com>;tag=3606020429-159196
Contact: <sip:17772189338@192.168.100.178:5060>
Call-ID: 243984fe4757a35f5c862dae33f18131@callcentric.com
CSeq: 105 ACK
User-Agent: Asterisk PBX 12.1.1
Content-Length: 0
---
[Apr 9 16:20:39] WARNING[23910][C-00000000]: channel.c:5332 set_format: Unable to find a codec translation path from (g729) to (slin)
[Apr 9 16:20:39] WARNING[23910][C-00000000]: indications.c:157 playtones_alloc: Unable to set 'SIP/204.11.192.132-00000000' to signed linear format (write)
[Apr 9 16:20:39] WARNING[23910][C-00000000]: channel.c:4651 ast_indicate_data: Unable to handle indication 8 for 'SIP/204.11.192.132-00000000'
Reliably Transmitting (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK2d99e462
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as12c5d2b1
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 117499340868a01155b5e3c83d883018@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
<--- SIP read from UDP:210.3.88.146:1218 --->
<------------->
Retransmitting #1 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK2d99e462
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as12c5d2b1
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 117499340868a01155b5e3c83d883018@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Retransmitting #2 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK2d99e462
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as12c5d2b1
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 117499340868a01155b5e3c83d883018@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Retransmitting #3 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK2d99e462
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as12c5d2b1
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 117499340868a01155b5e3c83d883018@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Retransmitting #4 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK2d99e462
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as12c5d2b1
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 117499340868a01155b5e3c83d883018@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:20:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Really destroying SIP dialog '117499340868a01155b5e3c83d883018@192.168.100.178:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK4920b87f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as6a7ef1a3
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 295b529a07ef6ae37ec9c85e751052d2@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:21:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Retransmitting #1 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK4920b87f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as6a7ef1a3
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 295b529a07ef6ae37ec9c85e751052d2@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:21:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Retransmitting #2 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK4920b87f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as6a7ef1a3
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 295b529a07ef6ae37ec9c85e751052d2@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:21:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Retransmitting #3 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK4920b87f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as6a7ef1a3
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 295b529a07ef6ae37ec9c85e751052d2@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:21:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Retransmitting #4 (no NAT) to 210.3.88.146:1216:
OPTIONS sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.100.178:5060;branch=z9hG4bK4920b87f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.178>;tag=as6a7ef1a3
To: <sip:6001@210.3.88.146:1216;rinstance=f4d5a5ee19b39ba9;transport=UDP>
Contact: <sip:asterisk@192.168.100.178:5060>
Call-ID: 295b529a07ef6ae37ec9c85e751052d2@192.168.100.178:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.1.1
Date: Wed, 09 Apr 2014 08:21:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
---
Really destroying SIP dialog '295b529a07ef6ae37ec9c85e751052d2@192.168.100.178:5060' Method: OPTIONS
444397-web2*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups