<--- SIP read from WS:x.x.x.x:65083 --->
INVITE sip:1061@y.y.y.y SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKoH1wEVekyxX1EdEK42XkixUoy3KHoXbJ;rport
From: "1060"<sip:1060@y.y.y.y>;tag=rf4ZEllaSI3suiZv9IqW
To: <sip:1061@y.y.y.y>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=1060;ha1=7f73f6e3b99397dda7d2bee39d586a32;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: a18a9df3-8396-0692-bc8b-578c2715b651
CSeq: 53755 INVITE
Content-Type: application/sdp
Content-Length: 949
Route: <sip:y.y.y.y:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.01.27
Organization: Doubango Telecom


v=0
o=Mozilla-SIPUA-29.0a1 23498 1 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:4445a85c
a=ice-pwd:673675c4dc614d992159118d92a812f6
a=fingerprint:sha-256 40:30:A2:FD:C4:C8:CF:44:FB:7F:3C:4A:34:B2:08:25:CC:27:C7:78:16:8B:3D:34:5B:31:A5:28:3A:16:A0:FF
m=audio 50152 UDP/TLS/RTP/SAVPF 109 0 8 101
c=IN IP4 x.x.x.x
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=setup:actpass
a=candidate:0 1 UDP 2128609535 z.z.z.z 50152 typ host
a=candidate:4 1 UDP 1692467199 x.x.x.x 50152 typ srflx raddr z.z.z.z rport 50152
a=candidate:5 1 UDP 2128543999 192.168.56.1 50153 typ host
a=candidate:0 2 UDP 2128609534 z.z.z.z 50154 typ host
a=candidate:4 2 UDP 1692467198 x.x.x.x 50154 typ srflx raddr z.z.z.z rport 50154
a=candidate:5 2 UDP 2128543998 192.168.56.1 50155 typ host
a=rtcp-mux
<------------->
--- (13 headers 24 lines) ---
Using INVITE request as basis request - a18a9df3-8396-0692-bc8b-578c2715b651
Found peer '1060' for '1060' from x.x.x.x:65083


<--- Reliably Transmitting (NAT) to x.x.x.x:65083 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKoH1wEVekyxX1EdEK42XkixUoy3KHoXbJ;received=x.x.x.x;rport=65083
From: "1060"<sip:1060@y.y.y.y>;tag=rf4ZEllaSI3suiZv9IqW
To: <sip:1061@y.y.y.y>;tag=as2e8929c8
Call-ID: a18a9df3-8396-0692-bc8b-578c2715b651
CSeq: 53755 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="y.y.y.y", nonce="55da2ff8"
Content-Length: 0




<------------>
Scheduling destruction of SIP dialog 'a18a9df3-8396-0692-bc8b-578c2715b651' in 32000 ms (Method: INVITE)


<--- SIP read from WS:x.x.x.x:65083 --->
ACK sip:1061@y.y.y.y SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKoH1wEVekyxX1EdEK42XkixUoy3KHoXbJ;rport
From: "1060"<sip:1060@y.y.y.y>;tag=rf4ZEllaSI3suiZv9IqW
To: <sip:1061@y.y.y.y>;tag=as2e8929c8
Call-ID: a18a9df3-8396-0692-bc8b-578c2715b651
CSeq: 53755 ACK
Content-Length: 0
Route: <sip:y.y.y.y:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70


<------------->
--- (9 headers 0 lines) ---


<--- SIP read from WS:x.x.x.x:65083 --->
INVITE sip:1061@y.y.y.y SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKT28PDG0I9qfGBCqWAwQpm1daL6b1Hqyy;rport
From: "1060"<sip:1060@y.y.y.y>;tag=rf4ZEllaSI3suiZv9IqW
To: <sip:1061@y.y.y.y>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=1060;ha1=7f73f6e3b99397dda7d2bee39d586a32;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: a18a9df3-8396-0692-bc8b-578c2715b651
CSeq: 53756 INVITE
Content-Type: application/sdp
Content-Length: 949
Route: <sip:y.y.y.y:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="y.y.y.y",nonce="55da2ff8",uri="sip:1061@y.y.y.y",response="bab983375e084747e991c92a5344032d",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.01.27
Organization: Doubango Telecom


v=0
o=Mozilla-SIPUA-29.0a1 23498 1 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:4445a85c
a=ice-pwd:673675c4dc614d992159118d92a812f6
a=fingerprint:sha-256 40:30:A2:FD:C4:C8:CF:44:FB:7F:3C:4A:34:B2:08:25:CC:27:C7:78:16:8B:3D:34:5B:31:A5:28:3A:16:A0:FF
m=audio 50152 UDP/TLS/RTP/SAVPF 109 0 8 101
c=IN IP4 x.x.x.x
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=setup:actpass
a=candidate:0 1 UDP 2128609535 z.z.z.z 50152 typ host
a=candidate:4 1 UDP 1692467199 x.x.x.x 50152 typ srflx raddr z.z.z.z rport 50152
a=candidate:5 1 UDP 2128543999 192.168.56.1 50153 typ host
a=candidate:0 2 UDP 2128609534 z.z.z.z 50154 typ host
a=candidate:4 2 UDP 1692467198 x.x.x.x 50154 typ srflx raddr z.z.z.z rport 50154
a=candidate:5 2 UDP 2128543998 192.168.56.1 50155 typ host
a=rtcp-mux


<------------->
--- (14 headers 24 lines) ---
Using INVITE request as basis request - a18a9df3-8396-0692-bc8b-578c2715b651
Found peer '1060' for '1060' from x.x.x.x:65083
  == Using SIP RTP CoS mark 5
Found RTP audio format 109
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found unknown media description format opus for ID 109
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
[Feb  3 09:13:22] WARNING[13308][C-00000206]: chan_sip.c:10496 process_sdp: Processed DTLS [TRUE]
Capabilities: us - (gsm|ulaw|alaw|g729|ilbc|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port x.x.x.x:50152
Looking for 1061 in default (domain y.y.y.y)
list_route: hop: <sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>


<--- Transmitting (NAT) to x.x.x.x:65083 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKT28PDG0I9qfGBCqWAwQpm1daL6b1Hqyy;received=x.x.x.x;rport=65083
From: "1060"<sip:1060@y.y.y.y>;tag=rf4ZEllaSI3suiZv9IqW
To: <sip:1061@y.y.y.y>
Call-ID: a18a9df3-8396-0692-bc8b-578c2715b651
CSeq: 53756 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1061@y.y.y.y:5060;transport=WS>
Content-Length: 0




<------------>
    -- Executing [1061@default:1] Gosub("SIP/1060-00000064", "1061,stdexten(SIP/1061)") in new stack
       > [INSERT INTO asterisk_db_cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('CHAN_START',{ts '2014-02-03 09:13:22'},'New User','1060','','','','1061','default','SIP/1060-00000064','','',3,'','1391436802.100','1391436802.100','','','')]
    -- Executing [1061@default:50000] NoOp("SIP/1060-00000064", "Start stdexten") in new stack
    -- Executing [1061@default:50001] Set("SIP/1060-00000064", "LOCAL(ext)=1061") in new stack
    -- Executing [1061@default:50002] Set("SIP/1060-00000064", "LOCAL(dev)=SIP/1061") in new stack
    -- Executing [1061@default:50003] Set("SIP/1060-00000064", "LOCAL(cntx)=") in new stack
    -- Executing [1061@default:50004] Set("SIP/1060-00000064", "LOCAL(mbx)=1061") in new stack
    -- Executing [1061@default:50005] Dial("SIP/1060-00000064", "SIP/1061,20") in new stack
  == Using SIP RTP CoS mark 5
       > [INSERT INTO asterisk_db_cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('CHAN_START',{ts '2014-02-03 09:13:22'},'New User','','','','','s','default','SIP/1061-00000065','','',3,'','1391436802.101','1391436802.100','','','')]
Audio is at 19798
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to x.x.x.x:65057:
INVITE sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS y.y.y.y:5060;branch=z9hG4bK362dda34;rport
Max-Forwards: 70
From: "New User" <sip:1060@y.y.y.y>;tag=as27be227e
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Contact: <sip:1060@y.y.y.y:5060;transport=WS>
Call-ID: 16bc4c8e069a2c7f5d83a71a3b2e641f@y.y.y.y:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Mon, 03 Feb 2014 14:13:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1001


v=0
o=root 474782337 474782337 IN IP4 y.y.y.y
s=Asterisk PBX 11.7.0
c=IN IP4 y.y.y.y
t=0 0
m=audio 19798 RTP/SAVPF 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:683ac0870b5c607e3675f87e140eed8e
a=ice-pwd:737b09545e7f64a11f56065339bcd19f
a=candidate:Ha2f2df83 1 UDP 2130706431 y.y.y.y 19798 typ host
a=candidate:Hab0a297 1 UDP 2130706431 10.176.162.151 19798 typ host
a=candidate:Hc0a8a804 1 UDP 2130706431 192.168.168.4 19798 typ host
a=candidate:Sa2f2df83 1 UDP 1694498815 y.y.y.y 19798 typ srflx
a=candidate:Ha2f2df83 2 UDP 2130706430 y.y.y.y 19799 typ host
a=candidate:Hab0a297 2 UDP 2130706430 10.176.162.151 19799 typ host
a=candidate:Hc0a8a804 2 UDP 2130706430 192.168.168.4 19799 typ host
a=candidate:Sa2f2df83 2 UDP 1694498814 y.y.y.y 19799 typ srflx
a=connection:new
a=setup:active
a=sendrecv


---
    -- Called SIP/1061


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '309484d103312ff825e78f6476793bbd@y.y.y.y:5060' Method: OPTIONS


<--- SIP read from WS:x.x.x.x:65057 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS y.y.y.y:5060;rport=5060;branch=z9hG4bK362dda34
From: "New User"<sip:1060@y.y.y.y>;tag=as27be227e
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>
Call-ID: 16bc4c8e069a2c7f5d83a71a3b2e641f@y.y.y.y:5060
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---


<--- SIP read from WS:x.x.x.x:65057 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS y.y.y.y:5060;rport=5060;branch=z9hG4bK362dda34
From: "New User"<sip:1060@y.y.y.y>;tag=as27be227e
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=1MBCSw86q3I4tNl6T5rf
Contact: <sip:1061@df7jal23ls0d.invalid;transport=ws>
Call-ID: 16bc4c8e069a2c7f5d83a71a3b2e641f@y.y.y.y:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE


<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:1061@df7jal23ls0d.invalid;transport=ws>
    -- SIP/1061-00000065 is ringing


<--- Transmitting (NAT) to x.x.x.x:65083 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKT28PDG0I9qfGBCqWAwQpm1daL6b1Hqyy;received=x.x.x.x;rport=65083
From: "1060"<sip:1060@y.y.y.y>;tag=rf4ZEllaSI3suiZv9IqW
To: <sip:1061@y.y.y.y>;tag=as2641dea4
Call-ID: a18a9df3-8396-0692-bc8b-578c2715b651
CSeq: 53756 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1061@y.y.y.y:5060;transport=WS>
Content-Length: 0




<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '51b4c92b1dc9961b0c382c365e082e63@y.y.y.y:5060' Method: OPTIONS
    -- ICE is now complete


<--- SIP read from WS:x.x.x.x:65057 --->
SIP/2.0 603 Failed to get local SDP
Via: SIP/2.0/WS y.y.y.y:5060;rport=5060;branch=z9hG4bK362dda34
From: "New User"<sip:1060@y.y.y.y>;tag=as27be227e
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=1MBCSw86q3I4tNl6T5rf
Call-ID: 16bc4c8e069a2c7f5d83a71a3b2e641f@y.y.y.y:5060
CSeq: 102 INVITE
Content-Length: 0
Reason: SIP; cause=603; text="Failed to get local SDP"


<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 603 "Failed to get local SDP" back from x.x.x.x:65057
set_destination: Parsing <sip:1061@df7jal23ls0d.invalid;transport=ws> for address/port to send to
set_destination: URI is for WebSocket, we can't set destination
Transmitting (NAT) to x.x.x.x:65057:
ACK sip:1061@df7jal23ls0d.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS y.y.y.y:5060;branch=z9hG4bK362dda34;rport
Max-Forwards: 70
From: "New User" <sip:1060@y.y.y.y>;tag=as27be227e
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=1MBCSw86q3I4tNl6T5rf
Contact: <sip:1060@y.y.y.y:5060;transport=WS>
Call-ID: 16bc4c8e069a2c7f5d83a71a3b2e641f@y.y.y.y:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0
Content-Length: 0




---
    -- SIP/1061-00000065 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [1061@default:50006] Goto("SIP/1060-00000064", "stdexten-BUSY,1") in new stack
    -- Goto (default,stdexten-BUSY,1)
    -- Executing [stdexten-BUSY@default:1] VoiceMail("SIP/1060-00000064", "1061,b") in new stack
    --   >> Doing DTLS handshake as well...
    --   >> [activate] check pending...
Audio is at 18268
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


<--- Reliably Transmitting (NAT) to x.x.x.x:65083 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKT28PDG0I9qfGBCqWAwQpm1daL6b1Hqyy;received=x.x.x.x;rport=65083
From: "1060"<sip:1060@y.y.y.y>;tag=rf4ZEllaSI3suiZv9IqW
To: <sip:1061@y.y.y.y>;tag=as2641dea4
Call-ID: a18a9df3-8396-0692-bc8b-578c2715b651
CSeq: 53756 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1061@y.y.y.y:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 1099


v=0
o=root 1383765036 1383765036 IN IP4 y.y.y.y
s=Asterisk PBX 11.7.0
c=IN IP4 y.y.y.y
t=0 0
m=audio 18268 RTP/SAVPF 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:0192884b4f19f1773e601d8746d019a9
a=ice-pwd:4ecbdf147f46f2151231e79c1d050f14
a=candidate:Ha2f2df83 1 UDP 2130706431 y.y.y.y 18268 typ host
a=candidate:Hab0a297 1 UDP 2130706431 10.176.162.151 18268 typ host
a=candidate:Hc0a8a804 1 UDP 2130706431 192.168.168.4 18268 typ host
a=candidate:Sa2f2df83 1 UDP 1694498815 y.y.y.y 18268 typ srflx
a=candidate:Ha2f2df83 2 UDP 2130706430 y.y.y.y 18269 typ host
a=candidate:Hab0a297 2 UDP 2130706430 10.176.162.151 18269 typ host
a=candidate:Hc0a8a804 2 UDP 2130706430 192.168.168.4 18269 typ host
a=candidate:Sa2f2df83 2 UDP 1694498814 y.y.y.y 18269 typ srflx
a=connection:new
a=setup:active
a=fingerprint:SHA-256 4B:D0:F1:39:36:B7:B0:91:57:87:9F:1F:6E:CF:C4:37:D2:BB:C7:72:F9:DC:A4:CB:50:64:9E:0D:FF:D0:40:39
a=sendrecv


<------------>
  


<--- SIP read from WS:x.x.x.x:65057 --->
SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS y.y.y.y:5060;rport=5060;branch=z9hG4bK362dda34
From: "New User"<sip:1060@y.y.y.y>;tag=as27be227e
To: <sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;tag=1MBCSw86q3I4tNl6T5rf
Call-ID: 16bc4c8e069a2c7f5d83a71a3b2e641f@y.y.y.y:5060
CSeq: 102 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
[Feb  3 09:13:34] WARNING[13306][C-00000206]: chan_sip.c:23953 handle_response: Remote host can't match request ACK to call '16bc4c8e069a2c7f5d83a71a3b2e641f@y.y.y.y:5060'. Giving up.


<--- SIP read from WS:x.x.x.x:65083 --->
ACK sip:1061@y.y.y.y:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKk58byld8GTlb9JpCc1Pd;rport
From: "1060"<sip:1060@y.y.y.y>;tag=rf4ZEllaSI3suiZv9IqW
To: <sip:1061@y.y.y.y>;tag=as2641dea4
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: a18a9df3-8396-0692-bc8b-578c2715b651
CSeq: 53756 ACK
Content-Length: 0
Route: <sip:y.y.y.y:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="y.y.y.y",nonce="55da2ff8",uri="sip:1061@y.y.y.y:5060;transport=WS",response="941a6a689cd127974f534454a98289a3",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.01.27
Organization: Doubango Telecom


<------------->
--- (13 headers 0 lines) ---
[Feb  3 09:13:34] WARNING[13405][C-00000206]: app_voicemail.c:6321 leave_voicemail: No entry in voicemail config file for '1061'
    -- Executing [stdexten-BUSY@default:2] Return("SIP/1060-00000064", "") in new stack
    -- Auto fallthrough, channel 'SIP/1060-00000064' status is 'BUSY'
Really destroying SIP dialog '16bc4c8e069a2c7f5d83a71a3b2e641f@y.y.y.y:5060' Method: INVITE