PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
SIP Debugging Enabled for IP: 83.172.40.9
<--- SIP read from UDP:83.172.40.9:5060 --->
INVITE sip:712199@109.194.33.177;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 83.172.40.9:5060;rport;branch=z9hG4bKrcfkdakrhcm2456g9p8k
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 INVITE
Contact: <sip:m-200@83.172.40.9;transport=UDP>
Max-Forwards: 70
User-Agent: M-200 Motor 5.86.75 (MPSS)
Accept: application/sdp, application/dtmf, application/dtmf-relay
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, UPDATE, PRACK, REFER, SUBSCRIBE, REGISTER
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 249
v=0
o=712199 3591895880 3591895880 IN IP4 83.172.40.9
s=(769/56,8:5)session@m-200
c=IN IP4 83.172.40.9
t=0 0
m=audio 8010 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:13 CN/8000
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 83.172.40.9:5060 (NAT)
Sending to 83.172.40.9:5060 (NAT)
Using INVITE request as basis request - 9o71jwdlt85se98ixwql38lso@83.172.40.9
Found peer 'neotelecom' for 'm-200' from 83.172.40.9:5060
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found audio description format CN for ID 13
Capabilities: us - (gsm|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x3 (telephone-event|CN|), combined - 0x0 (nothing)
Peer audio RTP is at port 83.172.40.9:8010
Looking for 712199 in from-trunk-sip-neotelecom (domain 109.194.33.177)
list_route: hop: <sip:m-200@83.172.40.9;transport=UDP>
<--- Transmitting (NAT) to 83.172.40.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 83.172.40.9:5060;branch=z9hG4bKrcfkdakrhcm2456g9p8k;received=83.172.40.9;rport=5060
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 INVITE
Server: FPBX-12.0.76(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:712199@109.194.33.177:5060>
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 83.172.40.9:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 83.172.40.9:5060;branch=z9hG4bKrcfkdakrhcm2456g9p8k;received=83.172.40.9;rport=5060
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>;tag=as370fa79d
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 INVITE
Server: FPBX-12.0.76(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:712199@109.194.33.177:5060>
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 83.172.40.9:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 83.172.40.9:5060;branch=z9hG4bKrcfkdakrhcm2456g9p8k;received=83.172.40.9;rport=5060
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>;tag=as370fa79d
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 INVITE
Server: FPBX-12.0.76(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:712199@109.194.33.177:5060>
Content-Length: 0
<------------>
Audio is at 10018
Adding codec 100004 (alaw) to SDP
<--- Transmitting (NAT) to 83.172.40.9:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 83.172.40.9:5060;branch=z9hG4bKrcfkdakrhcm2456g9p8k;received=83.172.40.9;rport=5060
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>;tag=as370fa79d
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 INVITE
Server: FPBX-12.0.76(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:712199@109.194.33.177:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 182
v=0
o=root 732136328 732136328 IN IP4 109.194.33.177
s=Asterisk PBX 11.13.1
c=IN IP4 109.194.33.177
t=0 0
m=audio 10018 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
<------------>
Audio is at 10006
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Reliably Transmitting (NAT) to 83.172.40.9:5060:
INVITE sip:332682@83.172.40.9 SIP/2.0
Via: SIP/2.0/UDP 109.194.33.177:5060;branch=z9hG4bK384dbcc0;rport
Max-Forwards: 70
From: <sip:712199@109.194.33.177>;tag=as3ce5903e
To: <sip:332682@83.172.40.9>
Contact: <sip:712199@109.194.33.177:5060>
Call-ID: 18a7a241071e3bef7a11033b30180149@109.194.33.177:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.76(11.13.1)
Date: Tue, 15 Sep 2015 06:29:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 205
v=0
o=root 395890469 395890469 IN IP4 109.194.33.177
s=Asterisk PBX 11.13.1
c=IN IP4 109.194.33.177
t=0 0
m=audio 10006 RTP/AVP 8 3
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:83.172.40.9:5060 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 109.194.33.177:5060;branch=z9hG4bK384dbcc0;received=109.194.33.177;rport=5060
From: <sip:712199@109.194.33.177>;tag=as3ce5903e
To: <sip:332682@83.172.40.9>;tag=w9hjmv4hpub74ly
Call-ID: 18a7a241071e3bef7a11033b30180149@109.194.33.177:5060
CSeq: 102 INVITE
Accept: application/sdp, application/dtmf, application/dtmf-relay
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, UPDATE, PRACK, REFER, SUBSCRIBE, REGISTER
Supported: timer, 100rel, replaces
Server: M-200 Motor 5.86.75 (MPSS)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 83.172.40.9:5060:
ACK sip:332682@83.172.40.9 SIP/2.0
Via: SIP/2.0/UDP 109.194.33.177:5060;branch=z9hG4bK384dbcc0;rport
Max-Forwards: 70
From: <sip:712199@109.194.33.177>;tag=as3ce5903e
To: <sip:332682@83.172.40.9>;tag=w9hjmv4hpub74ly
Contact: <sip:712199@109.194.33.177:5060>
Call-ID: 18a7a241071e3bef7a11033b30180149@109.194.33.177:5060
CSeq: 102 ACK
User-Agent: FPBX-12.0.76(11.13.1)
Content-Length: 0
---
Really destroying SIP dialog '18a7a241071e3bef7a11033b30180149@109.194.33.177:5060' Method: INVITE
Scheduling destruction of SIP dialog '9o71jwdlt85se98ixwql38lso@83.172.40.9' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 83.172.40.9:5060 --->
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 83.172.40.9:5060;branch=z9hG4bKrcfkdakrhcm2456g9p8k;received=83.172.40.9;rport=5060
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>;tag=as370fa79d
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 INVITE
Server: FPBX-12.0.76(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0
<------------>
<--- SIP read from UDP:83.172.40.9:5060 --->
ACK sip:712199@109.194.33.177;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 83.172.40.9:5060;rport;branch=z9hG4bKrcfkdakrhcm2456g9p8k
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>;tag=as370fa79d
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 ACK
Contact: <sip:m-200@83.172.40.9;transport=UDP>
Max-Forwards: 70
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:83.172.40.9:5060 --->
INVITE sip:712199@109.194.33.177;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 83.172.40.9:5060;rport;branch=z9hG4bKrcfkdakrhcm2456g9p8k
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 INVITE
Contact: <sip:m-200@83.172.40.9;transport=UDP>
Max-Forwards: 70
User-Agent: M-200 Motor 5.86.75 (MPSS)
Accept: application/sdp, application/dtmf, application/dtmf-relay
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, UPDATE, PRACK, REFER, SUBSCRIBE, REGISTER
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 249
v=0
o=712199 3591895880 3591895880 IN IP4 83.172.40.9
s=(769/56,8:5)session@m-200
c=IN IP4 83.172.40.9
t=0 0
m=audio 8010 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:13 CN/8000
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 83.172.40.9:5060 (NAT)
Sending to 83.172.40.9:5060 (NAT)
Using INVITE request as basis request - 9o71jwdlt85se98ixwql38lso@83.172.40.9
Found peer 'neotelecom' for 'm-200' from 83.172.40.9:5060
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found audio description format CN for ID 13
Capabilities: us - (gsm|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x3 (telephone-event|CN|), combined - 0x0 (nothing)
Peer audio RTP is at port 83.172.40.9:8010
Looking for 712199 in from-trunk-sip-neotelecom (domain 109.194.33.177)
list_route: hop: <sip:m-200@83.172.40.9;transport=UDP>
<--- Transmitting (NAT) to 83.172.40.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 83.172.40.9:5060;branch=z9hG4bKrcfkdakrhcm2456g9p8k;received=83.172.40.9;rport=5060
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 INVITE
Server: FPBX-12.0.76(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:712199@109.194.33.177:5060>
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 83.172.40.9:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 83.172.40.9:5060;branch=z9hG4bKrcfkdakrhcm2456g9p8k;received=83.172.40.9;rport=5060
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>;tag=as370fa79d
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 INVITE
Server: FPBX-12.0.76(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:712199@109.194.33.177:5060>
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 83.172.40.9:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 83.172.40.9:5060;branch=z9hG4bKrcfkdakrhcm2456g9p8k;received=83.172.40.9;rport=5060
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>;tag=as370fa79d
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 INVITE
Server: FPBX-12.0.76(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:712199@109.194.33.177:5060>
Content-Length: 0
<------------>
Audio is at 10018
Adding codec 100004 (alaw) to SDP
<--- Transmitting (NAT) to 83.172.40.9:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 83.172.40.9:5060;branch=z9hG4bKrcfkdakrhcm2456g9p8k;received=83.172.40.9;rport=5060
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>;tag=as370fa79d
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 INVITE
Server: FPBX-12.0.76(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:712199@109.194.33.177:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 182
v=0
o=root 732136328 732136328 IN IP4 109.194.33.177
s=Asterisk PBX 11.13.1
c=IN IP4 109.194.33.177
t=0 0
m=audio 10018 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
<------------>
Audio is at 10006
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Reliably Transmitting (NAT) to 83.172.40.9:5060:
INVITE sip:332682@83.172.40.9 SIP/2.0
Via: SIP/2.0/UDP 109.194.33.177:5060;branch=z9hG4bK384dbcc0;rport
Max-Forwards: 70
From: <sip:712199@109.194.33.177>;tag=as3ce5903e
To: <sip:332682@83.172.40.9>
Contact: <sip:712199@109.194.33.177:5060>
Call-ID: 18a7a241071e3bef7a11033b30180149@109.194.33.177:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.76(11.13.1)
Date: Tue, 15 Sep 2015 06:29:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 205
v=0
o=root 395890469 395890469 IN IP4 109.194.33.177
s=Asterisk PBX 11.13.1
c=IN IP4 109.194.33.177
t=0 0
m=audio 10006 RTP/AVP 8 3
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:83.172.40.9:5060 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 109.194.33.177:5060;branch=z9hG4bK384dbcc0;received=109.194.33.177;rport=5060
From: <sip:712199@109.194.33.177>;tag=as3ce5903e
To: <sip:332682@83.172.40.9>;tag=w9hjmv4hpub74ly
Call-ID: 18a7a241071e3bef7a11033b30180149@109.194.33.177:5060
CSeq: 102 INVITE
Accept: application/sdp, application/dtmf, application/dtmf-relay
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, UPDATE, PRACK, REFER, SUBSCRIBE, REGISTER
Supported: timer, 100rel, replaces
Server: M-200 Motor 5.86.75 (MPSS)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 83.172.40.9:5060:
ACK sip:332682@83.172.40.9 SIP/2.0
Via: SIP/2.0/UDP 109.194.33.177:5060;branch=z9hG4bK384dbcc0;rport
Max-Forwards: 70
From: <sip:712199@109.194.33.177>;tag=as3ce5903e
To: <sip:332682@83.172.40.9>;tag=w9hjmv4hpub74ly
Contact: <sip:712199@109.194.33.177:5060>
Call-ID: 18a7a241071e3bef7a11033b30180149@109.194.33.177:5060
CSeq: 102 ACK
User-Agent: FPBX-12.0.76(11.13.1)
Content-Length: 0
---
Really destroying SIP dialog '18a7a241071e3bef7a11033b30180149@109.194.33.177:5060' Method: INVITE
Scheduling destruction of SIP dialog '9o71jwdlt85se98ixwql38lso@83.172.40.9' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 83.172.40.9:5060 --->
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 83.172.40.9:5060;branch=z9hG4bKrcfkdakrhcm2456g9p8k;received=83.172.40.9;rport=5060
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>;tag=as370fa79d
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 INVITE
Server: FPBX-12.0.76(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0
<------------>
<--- SIP read from UDP:83.172.40.9:5060 --->
ACK sip:712199@109.194.33.177;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 83.172.40.9:5060;rport;branch=z9hG4bKrcfkdakrhcm2456g9p8k
From: <sip:m-200@83.172.40.9;cpc=ordinary>;tag=31hxfa23wup4v4h
To: <sip:712199@109.194.33.177>;tag=as370fa79d
Call-ID: 9o71jwdlt85se98ixwql38lso@83.172.40.9
CSeq: 51 ACK
Contact: <sip:m-200@83.172.40.9;transport=UDP>
Max-Forwards: 70
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
