<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://10.X.X.X:61399 --->
INVITE sip:90950XXXXXX@WORKGROUP:5060 SIP/2.0
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-7d5eff759e47110d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:202@10.X.X.X:61399;rinstance=340261802ca46cef >
To: <sip:90950XXXXXX@WORKGROUP:5060>
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.19920.0
Content-Length: 407
v=0
o=3cxVCE 158336970 98167440 IN IP4 10.X.X.X
s=3cxVCE Audio Call
c=IN IP4 10.X.X.X
t=0 0
m=audio 40036 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40034 RTP/AVP 34
c=IN IP4 10.X.X.X
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1;
a=sendrecv
<------------->
--- (13 headers 18 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 10.X.X.X : 61399 (no NAT)
Using INVITE request as basis request - Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
Found user '202' for '202'
<--- Reliably Transmitting (no NAT) to 10.X.X.X:61399 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-7d5eff759e47110d-1---d8754z-;received=10.X.X.X;rport=61399
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
To: <sip:90950XXXXXX@WORKGROUP:5060>;tag=as2c365984
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="098e1d89"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.' in 32000 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://10.X.X.X:61399 --->
ACK sip:90950XXXXXX@WORKGROUP:5060 SIP/2.0
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-7d5eff759e47110d-1---d8754z-;rport
Max-Forwards: 70
To: <sip:90950XXXXXX@WORKGROUP:5060>;tag=as2c365984
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://10.X.X.X:61399 --->
INVITE sip:90950XXXXXX@WORKGROUP:5060 SIP/2.0
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-b65fcc098d7cc872-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:202@10.X.X.X:61399;rinstance=340261802ca46cef >
To: <sip:90950XXXXXX@WORKGROUP:5060>
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.19920.0
Authorization: Digest username="202",realm="asterisk",nonce="098e1d89",u ri="sip:90950XXXXXX@WORKGROUP:5060",response="217b 28fe239c388230418edca15ba4f9",algorithm=MD5
Content-Length: 407
v=0
o=3cxVCE 158336970 98167440 IN IP4 10.X.X.X
s=3cxVCE Audio Call
c=IN IP4 10.X.X.X
t=0 0
m=audio 40036 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40034 RTP/AVP 34
c=IN IP4 10.X.X.X
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1;
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 10.X.X.X : 61399 (no NAT)
Using INVITE request as basis request - Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
Found user '202' for '202'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
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 GSM for ID 3
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found video description format H263 for ID 34
Capabilities: us - 0x4 (ulaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.X.X.X:40036
Looking for 90950XXXXXX in appel_interne_X (domain WORKGROUP)
list_route: hop: <sip:202@10.X.X.X:61399;rinstance=340261802ca46cef >
SRVVOIP1*CLI>
<--- Transmitting (no NAT) to 10.X.X.X:61399 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-b65fcc098d7cc872-1---d8754z-;received=10.X.X.X;rport=61399
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
To: <sip:90950XXXXXX@WORKGROUP:5060>
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:90950XXXXXX@10.X.X.X>
Content-Length: 0
<------------>
-- Executing [90950XXXXXX@appel_interne_X:1] ChanIsAvail("SIP/202-0000000b", "SIP/90950XXXXXX@free-0950XXXXXX") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Really destroying SIP dialog '3d93105b788f6f256b4ddcdb1e4defce@10.X.X.X' Method: INVITE
-- Executing [90950XXXXXX@appel_interne_X:2] Dial("SIP/202-0000000b", "SIP/90950XXXXXX@free-0950XXXXXX),30,r") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Really destroying SIP dialog '2870e9af28c7940d24f94dc775dcf1dd@10.X.X.X' Method: INVITE
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [90950XXXXXX@appel_interne_X:3] Congestion("SIP/202-0000000b", "") in new stack
<--- Reliably Transmitting (no NAT) to 10.X.X.X:61399 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-b65fcc098d7cc872-1---d8754z-;received=10.X.X.X;rport=61399
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
To: <sip:90950XXXXXX@WORKGROUP:5060>;tag=as7c874b55
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 20
<------------>
== Spawn extension (appel_interne_X, 90950XXXXXX, 3) exited non-zero on 'SIP/202-0000000b'
SRVVOIP1*CLI>
<--- SIP read from UDP://10.X.X.X:61399 --->
ACK sip:90950XXXXXX@WORKGROUP:5060 SIP/2.0
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-b65fcc098d7cc872-1---d8754z-;rport
Max-Forwards: 70
To: <sip:90950XXXXXX@WORKGROUP:5060>;tag=as7c874b55
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.' Method: ACK
SRVVOIP1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[SRVVOIP1.localdomain asterisk]#
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