All pastes #826854 Raw Edit

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#826854 ·published 2007-12-21 13:45 UTC
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SIP Debugging Enabled for IP: 10.0.1.40:5061
    -- Executing [901489574204@handsets:1] Dial("SIP/3101-081e1220", "SIP/sipuraline1/901489574204|60") in new stack
Audio is at 10.0.1.21 port 11784
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.1.40:5061:
INVITE sip:901489574204@10.0.1.40:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK686e3713;rport
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as777e8f50
To: <sip:901489574204@10.0.1.40:5061>
Contact: <sip:3101@10.0.1.21>
Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 21 Dec 2007 13:44:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 23787 23787 IN IP4 10.0.1.21
s=session
c=IN IP4 10.0.1.21
t=0 0
m=audio 11784 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
=ptime:20> 
a=sendrecv

---
    -- Called sipuraline1/901489574204
akira*CLI> 
<--- SIP read from 10.0.1.40:5061 --->
SIP/2.0 100 Trying
To: <sip:901489574204@10.0.1.40:5061>
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as777e8f50
Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK686e3713
Server: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
akira*CLI> 
<--- SIP read from 10.0.1.40:5061 --->
SIP/2.0 180 Ringing
To: <sip:901489574204@10.0.1.40:5061>;tag=20bea138f1fb2968i1
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as777e8f50
Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK686e3713
Server: Linksys/SPA3102-3.3.6(GW)
Remote-Party-ID: sipuraline1 <sip:sipuraline1@10.0.1.21>;screen=yes;party=called
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/sipuraline1-081e5188 is ringing
akira*CLI> 
<--- SIP read from 10.0.1.40:5061 --->
SIP/2.0 200 OK
To: <sip:901489574204@10.0.1.40:5061>;tag=20bea138f1fb2968i1
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as777e8f50
Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK686e3713
Contact: sipuraline1 <sip:901489574204@10.0.1.40:5061>
Server: Linksys/SPA3102-3.3.6(GW)
Remote-Party-ID: sipuraline1 <sip:sipuraline1@10.0.1.21>;screen=yes;party=called
Content-Length: 247
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 190145 190145 IN IP4 10.0.1.40
s=-
c=IN IP4 10.0.1.40
t=0 0
m=audio 16424 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 10.0.1.40:16424
Found description format PCMA for ID 8
Found description format NSE for ID 100
Found description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.1.40:16424
list_route: hop: <sip:901489574204@10.0.1.40:5061>
set_destination: Parsing <sip:901489574204@10.0.1.40:5061> for address/port to send to
set_destination: set destination to 10.0.1.40, port 5061
Transmitting (no NAT) to 10.0.1.40:5061:
ACK sip:901489574204@10.0.1.40:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK7a0e5a42;rport
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as777e8f50
To: <sip:901489574204@10.0.1.40:5061>;tag=20bea138f1fb2968i1
Contact: <sip:3101@10.0.1.21>
Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/sipuraline1-081e5188 answered SIP/3101-081e1220
    -- Packet2Packet bridging SIP/3101-081e1220 and SIP/sipuraline1-081e5188
Scheduling destruction of SIP dialog '526dc357589aff4f601ca148086f39d9@10.0.1.21' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:901489574204@10.0.1.40:5061> for address/port to send to
set_destination: set destination to 10.0.1.40, port 5061
Reliably Transmitting (no NAT) to 10.0.1.40:5061:
BYE sip:901489574204@10.0.1.40:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK39ba3204;rport
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as777e8f50
To: <sip:901489574204@10.0.1.40:5061>;tag=20bea138f1fb2968i1
Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Spawn extension (handsets, 901489574204, 1) exited non-zero on 'SIP/3101-081e1220'
akira*CLI> 
<--- SIP read from 10.0.1.40:5061 --->
SIP/2.0 200 OK
To: <sip:901489574204@10.0.1.40:5061>;tag=20bea138f1fb2968i1
From: "Handset 1" <sip:3101@10.0.1.21>;tag=as777e8f50
Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK39ba3204
Server: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '526dc357589aff4f601ca148086f39d9@10.0.1.21' Method: INVITE
Reliably Transmitting (no NAT) to 10.0.1.40:5061:
OPTIONS sip:10.0.1.40:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK50290e8a;rport
From: "asterisk" <sip:asterisk@10.0.1.21>;tag=as2c1f1d05
To: <sip:10.0.1.40:5061>
Contact: <sip:asterisk@10.0.1.21>
Call-ID: 7ee923d021d82d813cbd467c78aac9a7@10.0.1.21
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 21 Dec 2007 13:45:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
akira*CLI> 
<--- SIP read from 10.0.1.40:5061 --->
SIP/2.0 200 OK
To: <sip:10.0.1.40:5061>;tag=b328fe0c274b6cc7i1
From: "asterisk" <sip:asterisk@10.0.1.21>;tag=as2c1f1d05
Call-ID: 7ee923d021d82d813cbd467c78aac9a7@10.0.1.21
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK50290e8a
Server: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7ee923d021d82d813cbd467c78aac9a7@10.0.1.21' Method: OPTIONS
akira*CLI>