Public feed #601706

163 calls 600

public text v1 · immutable
Raw Edit
·published 2007-07-03 17:40 UTC
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trixbox*CLI> sip debug
SIP Debugging re-enabled
trixbox*CLI>
<-- SIP read from 200.200.200.41:5060:
INVITE sip:600@200.200.200.11:5060 SIP/2.0
From: 163 <sip:163@200.200.200.41>;tag=b175082a51435db
To: sip:600@200.200.200.11
Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bKbe226279c
Call-ID: 76712560dda5b8d
CSeq: 2009506258 INVITE
Max-Forwards: 70
Content-Length: 378
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,REFER,INFO,PRACK,UPDATE
P-Asserted-Identity:<sip:163@200.200.200.41>
Content-Type: application/sdp
Contact: sip:163@200.200.200.41:5060
Supported: replaces
User-Agent: HiPath 3000 V7.0 M5T SIP-UA SAFE/v3.6.6.10

v=0
o=MxSIP 0 1 IN IP4 200.200.200.168
s=SIP Call
c=IN IP4 200.200.200.168
t=0 0
a=sendrecv
m=audio 5004 RTP/AVP 0 8 9 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:4 annexa=no
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20

--- (14 headers 18 lines) ---
Using INVITE request as basis request - 76712560dda5b8d
Sending to 200.200.200.41 : 5060 (non-NAT)
Found peer 'HiPath3800'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 200.200.200.168:5004
Found description format PCMU
Found description format PCMA
Found description format G722
Found description format G729
Found description format G723
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 600 in from-sip-external (domain 200.200.200.11)
list_route: hop: <sip:163@200.200.200.41:5060>
Transmitting (no NAT) to 200.200.200.41:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bKbe226279c;received=200.200.200.41
From: 163 <sip:163@200.200.200.41>;tag=b175082a51435db
To: sip:600@200.200.200.11
Call-ID: 76712560dda5b8d
CSeq: 2009506258 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@200.200.200.11>
Content-Length: 0


---
Transmitting (no NAT) to 200.200.200.41:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bKbe226279c;received=200.200.200.41
From: 163 <sip:163@200.200.200.41>;tag=b175082a51435db
To: sip:600@200.200.200.11;tag=as1bc531fd
Call-ID: 76712560dda5b8d
CSeq: 2009506258 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@200.200.200.11>
Content-Length: 0


---
We're at 200.200.200.11 port 11088
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 200.200.200.41:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bKbe226279c;received=200.200.200.41
From: 163 <sip:163@200.200.200.41>;tag=b175082a51435db
To: sip:600@200.200.200.11;tag=as1bc531fd
Call-ID: 76712560dda5b8d
CSeq: 2009506258 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@200.200.200.11>
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 2802 2802 IN IP4 200.200.200.11
s=session
c=IN IP4 200.200.200.11
t=0 0
m=audio 11088 RTP/AVP 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 - - - -

---
trixbox*CLI>
<-- SIP read from 200.200.200.41:5060:
ACK sip:600@200.200.200.11 SIP/2.0
From: 163 <sip:163@200.200.200.41>;tag=b175082a51435db
To: sip:600@200.200.200.11;tag=as1bc531fd
Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bKdf087f008
Call-ID: 76712560dda5b8d
CSeq: 2009506258 ACK
Max-Forwards: 70
Content-Length: 0
User-Agent: HiPath 3000 V7.0 M5T SIP-UA SAFE/v3.6.6.10


--- (9 headers 0 lines) ---
trixbox*CLI>
<-- SIP read from 200.200.200.41:5060:
INVITE sip:600@200.200.200.11 SIP/2.0
From: 163 <sip:163@200.200.200.41>;tag=b175082a51435db
To: sip:600@200.200.200.11;tag=as1bc531fd
Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bK0d1c2a983
Call-ID: 76712560dda5b8d
CSeq: 2009506259 INVITE
Max-Forwards: 70
Content-Length: 366
Content-Type: application/sdp
Supported: replaces
Contact: sip:163@200.200.200.41:5060
User-Agent: HiPath 3000 V7.0 M5T SIP-UA SAFE/v3.6.6.10

v=0
o=MxSIP 0 2 IN IP4 200.200.200.168
s=SIP Call
c=IN IP4 200.200.200.168
t=0 0
m=audio 5004 RTP/AVP 0 8 9 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:4 annexa=no
a=fmtp:101 0-15
a=inactive

--- (12 headers 17 lines) ---
Using INVITE request as basis request - 76712560dda5b8d
Sending to 200.200.200.41 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 200.200.200.168:5004
Found description format PCMU
Found description format PCMA
Found description format G722
Found description format G729
Found description format G723
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
We're at 200.200.200.11 port 11088
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 200.200.200.41:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bK0d1c2a983;received=200.200.200.41
From: 163 <sip:163@200.200.200.41>;tag=b175082a51435db
To: sip:600@200.200.200.11;tag=as1bc531fd
Call-ID: 76712560dda5b8d
CSeq: 2009506259 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@200.200.200.11>
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 2802 2803 IN IP4 200.200.200.11
s=session
c=IN IP4 200.200.200.11
t=0 0
m=audio 11088 RTP/AVP 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 - - - -

---
trixbox*CLI>
<-- SIP read from 200.200.200.41:5060:
ACK sip:600@200.200.200.11 SIP/2.0
From: 163 <sip:163@200.200.200.41>;tag=b175082a51435db
To: sip:600@200.200.200.11;tag=as1bc531fd
Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bK95f8b9aa6
Call-ID: 76712560dda5b8d
CSeq: 2009506259 ACK
Max-Forwards: 70
Content-Length: 0
User-Agent: HiPath 3000 V7.0 M5T SIP-UA SAFE/v3.6.6.10


--- (9 headers 0 lines) ---
trixbox*CLI>
<-- SIP read from 200.200.200.41:5060:
BYE sip:600@200.200.200.11 SIP/2.0
From: 163 <sip:163@200.200.200.41>;tag=b175082a51435db
To: sip:600@200.200.200.11;tag=as1bc531fd
Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bK378992d6f
Call-ID: 76712560dda5b8d
CSeq: 2009506260 BYE
Max-Forwards: 70
Content-Length: 0
Reason:Q.850; cause=16; text="Normal call clearing"
Supported: replaces
User-Agent: HiPath 3000 V7.0 M5T SIP-UA SAFE/v3.6.6.10


--- (11 headers 0 lines) ---
Sending to 200.200.200.41 : 5060 (non-NAT)
Transmitting (no NAT) to 200.200.200.41:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 200.200.200.41:5060;branch=z9hG4bK378992d6f;received=200.200.200.41
From: 163 <sip:163@200.200.200.41>;tag=b175082a51435db
To: sip:600@200.200.200.11;tag=as1bc531fd
Call-ID: 76712560dda5b8d
CSeq: 2009506260 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@200.200.200.11>
Content-Length: 0


---
Destroying call '76712560dda5b8d'
trixbox*CLI>