
<--- SIP read from 89.35.79.69:61286 --->
INVITE sip:0748881658@voip.telefonet.ro SIP/2.0
Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-9b3e837afc10592c-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:23003@89.35.79.69:61286>
To: "0748881658"<sip:0748881658@voip.telefonet.ro>
From: "Znuff"<sip:23003@voip.telefonet.ro>;tag=ee2e3811
Call-ID: YTkzNzAwNDI3MDQ1OTIzNWI5OGU1MGMxOTNmNWVkMzA.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 232

v=0
o=- 6 2 IN IP4 89.35.79.69
s=CounterPath X-Lite 3.0
c=IN IP4 89.35.79.69
t=0 0
m=audio 56108 RTP/AVP 0 8 101
a=alt:1 1 : bMXyrCNL s3Pt6TaB 89.35.79.69 56108
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (12 headers 10 lines) ---
Sending to 89.35.79.69 : 61286 (NAT)
Using INVITE request as basis request - YTkzNzAwNDI3MDQ1OTIzNWI5OGU1MGMxOTNmNWVkMzA.
Found peer '23003'
voip*CLI>
<--- Reliably Transmitting (no NAT) to 89.35.79.69:61286 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-9b3e837afc10592c-1--d87543-;received=89.35.79.69;rport=61286
From: "Znuff"<sip:23003@voip.telefonet.ro>;tag=ee2e3811
To: "0748881658"<sip:0748881658@voip.telefonet.ro>;tag=as1b165a69
Call-ID: YTkzNzAwNDI3MDQ1OTIzNWI5OGU1MGMxOTNmNWVkMzA.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="voip.telefonet.ro", nonce="23df6c75"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'YTkzNzAwNDI3MDQ1OTIzNWI5OGU1MGMxOTNmNWVkMzA.' in 32000 ms (Method: INVITE)
voip*CLI>
<--- SIP read from 89.35.79.69:61286 --->
ACK sip:0748881658@voip.telefonet.ro SIP/2.0
Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-9b3e837afc10592c-1--d87543-;rport
To: "0748881658"<sip:0748881658@voip.telefonet.ro>;tag=as1b165a69
From: "Znuff"<sip:23003@voip.telefonet.ro>;tag=ee2e3811
Call-ID: YTkzNzAwNDI3MDQ1OTIzNWI5OGU1MGMxOTNmNWVkMzA.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
voip*CLI>
<--- SIP read from 89.35.79.69:61286 --->
INVITE sip:0748881658@voip.telefonet.ro SIP/2.0
Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-d66b0150cb2a892f-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:23003@89.35.79.69:61286>
To: "0748881658"<sip:0748881658@voip.telefonet.ro>
From: "Znuff"<sip:23003@voip.telefonet.ro>;tag=ee2e3811
Call-ID: YTkzNzAwNDI3MDQ1OTIzNWI5OGU1MGMxOTNmNWVkMzA.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="23003",realm="voip.telefonet.ro",nonce="23df6c75",uri="sip:0748881658@voip.telefonet.ro",response="73e473919666a3e7978983bdee51594f",algorithm=MD5
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 232

v=0
o=- 6 2 IN IP4 89.35.79.69
s=CounterPath X-Lite 3.0
c=IN IP4 89.35.79.69
t=0 0
m=audio 56108 RTP/AVP 0 8 101
a=alt:1 1 : bMXyrCNL s3Pt6TaB 89.35.79.69 56108
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (13 headers 10 lines) ---
Sending to 89.35.79.69 : 61286 (NAT)
Using INVITE request as basis request - YTkzNzAwNDI3MDQ1OTIzNWI5OGU1MGMxOTNmNWVkMzA.
Found peer '23003'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 89.35.79.69:56108
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 89.35.79.69:56108
Looking for 0748881658 in numberplan-custom-1 (domain voip.telefonet.ro)
list_route: hop: <sip:23003@89.35.79.69:61286>
voip*CLI>
<--- Transmitting (no NAT) to 89.35.79.69:61286 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-d66b0150cb2a892f-1--d87543-;received=89.35.79.69;rport=61286
From: "Znuff"<sip:23003@voip.telefonet.ro>;tag=ee2e3811
To: "0748881658"<sip:0748881658@voip.telefonet.ro>
Call-ID: YTkzNzAwNDI3MDQ1OTIzNWI5OGU1MGMxOTNmNWVkMzA.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0748881658@89.35.79.5>
Content-Length: 0


<------------>
    -- Executing [0748881658@numberplan-custom-1:1] Dial("SIP/23003-b5a05d60", "SIP/0748881658") in new stack
[Mar  5 06:15:08] WARNING[9949]: chan_sip.c:2719 create_addr: No such host: 0748881658
Really destroying SIP dialog '514631ec706b176554ed1bbd20f70b6a@127.0.0.1' Method: INVITE
[Mar  5 06:15:08] WARNING[9949]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [0748881658@numberplan-custom-1:2] Dial("SIP/23003-b5a05d60", "SIP/0748881658@10.0.0.1") in new stack
Audio is at 10.0.0.2 port 14480
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.1:5060:
INVITE sip:0748881658@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK187a7d57;rport
From: "Znuff" <sip:23003@10.0.0.2>;tag=as701fbc4c
To: <sip:0748881658@10.0.0.1>
Contact: <sip:23003@10.0.0.2>
Call-ID: 2e09babf4fa8a8da0b6255a5020e788d@10.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 05 Mar 2007 04:15:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 8856 8856 IN IP4 89.35.79.69
s=session
c=IN IP4 89.35.79.69
t=0 0
m=audio 56108 RTP/AVP 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=sendrecv

---
    -- Called 0748881658@10.0.0.1
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
SIP/2.0 100 Trying
From: "Znuff"<sip:23003@10.0.0.2>;tag=as701fbc4c
To: <sip:0748881658@10.0.0.1>;tag=a000001-13c4-22a4-876f5c-229a
Call-ID: 2e09babf4fa8a8da0b6255a5020e788d@10.0.0.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK187a7d57;rport
Contact: <sip:0748881658@10.0.0.1>
Content-Length:0


<------------->
--- (8 headers 0 lines) ---
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
SIP/2.0 415 Unsupported Media Type
From: "Znuff"<sip:23003@10.0.0.2>;tag=as701fbc4c
To: <sip:0748881658@10.0.0.1>;tag=a000001-13c4-22a4-876f5c-229a
Call-ID: 2e09babf4fa8a8da0b6255a5020e788d@10.0.0.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK187a7d57;rport
Content-Length:0


<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 415 "Unsupported Media Type" back from 10.0.0.1
Transmitting (no NAT) to 10.0.0.1:5060:
ACK sip:0748881658@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK187a7d57;rport
From: "Znuff" <sip:23003@10.0.0.2>;tag=as701fbc4c
To: <sip:0748881658@10.0.0.1>;tag=a000001-13c4-22a4-876f5c-229a
Contact: <sip:23003@10.0.0.2>
Call-ID: 2e09babf4fa8a8da0b6255a5020e788d@10.0.0.2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [0748881658@numberplan-custom-1:3] Congestion("SIP/23003-b5a05d60", "") in new stack
voip*CLI>
<--- Transmitting (no NAT) to 89.35.79.69:61286 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-d66b0150cb2a892f-1--d87543-;received=89.35.79.69;rport=61286
From: "Znuff"<sip:23003@voip.telefonet.ro>;tag=ee2e3811
To: "0748881658"<sip:0748881658@voip.telefonet.ro>;tag=as54394cd9
Call-ID: YTkzNzAwNDI3MDQ1OTIzNWI5OGU1MGMxOTNmNWVkMzA.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0748881658@89.35.79.5>
Content-Length: 0
X-Asterisk-HangupCause: Interworking, unspecified
X-Asterisk-HangupCauseCode: 127


<------------>
  == Spawn extension (numberplan-custom-1, 0748881658, 3) exited non-zero on 'SIP/23003-b5a05d60'
voip*CLI>
<--- SIP read from 89.35.79.69:61286 --->
ACK sip:0748881658@voip.telefonet.ro SIP/2.0
Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-d66b0150cb2a892f-1--d87543-;rport
To: "0748881658"<sip:0748881658@voip.telefonet.ro>;tag=as54394cd9
From: "Znuff"<sip:23003@voip.telefonet.ro>;tag=ee2e3811
Call-ID: YTkzNzAwNDI3MDQ1OTIzNWI5OGU1MGMxOTNmNWVkMzA.
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '2e09babf4fa8a8da0b6255a5020e788d@10.0.0.2' Method: INVITE
Really destroying SIP dialog 'YTkzNzAwNDI3MDQ1OTIzNWI5OGU1MGMxOTNmNWVkMzA.' Method: ACK
voip*CLI>