[asterisk-users] Redirecting Calls

Tue Nov 14 01:17:25 MST 2006

Hello All.

I am stumped, please help me out..

I have the following setup:

VOIP provider = VOIP GW (asterisk GW1) = VOIP server (asterisk - VS1)

The gateway is there to get around the limitations running on the VOIP
server.

I can call out from and receive calls VS1 no problems at all. However,
when I try
and redirect an inbound call out via the GW, it drops out.

I have found that if I redirect the call at the gateway everything
works, but this
is not an option as we can't keep a track of it in our system.

Here is some logs that might help.
As far as I can tell I get "SIP/2.0 183 Session Progress" and the
"CANCEL" back from
the provider.

fone2 is SV1 above.

Any help at all would be greatly appreciated.

-- Executing Macro("SIP/fone2-6a06", "voip|090xxxxxxx3|japan") in new stack
-- Executing Macro("SIP/fone2-6a06", "japan|090xxxxxxx3") in new stack
-- Executing Set("SIP/fone2-6a06", "Var_FROM="090xxxxxxx4"
<sip:090xxxxxxx3 at 203.83.244.199>;tag=
as10d01d24") in new stack
-- Executing NoOp("SIP/fone2-6a06", ""090xxxxxxx4"
<sip:090xxxxxxx3 at 203.83.244.199>;tag=as10d01d
24)}") in new stack
-- Executing GotoIf("SIP/fone2-6a06", "0?4:6") in new stack
-- Goto (macro-japan,s,6)
-- Executing GotoIf("SIP/fone2-6a06", "0?7:10") in new stack
-- Goto (macro-japan,s,10)
-- Executing SetCallerID("SIP/fone2-6a06", "050xxxxxxx") in new stack
-- Executing SetCallerPres("SIP/fone2-6a06", "prohib_passed_screen") in
new stack
-- Executing Dial("SIP/fone2-6a06", "SIP/090xxxxxxx3 at 050xxxxxxx|720|tT")
in new stack
We're at 211.129.117.89 port 17262
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
15 headers, 10 lines
Reliably Transmitting (no NAT) to 210.227.109.232:5060:
INVITE sip:090xxxxxxx3 at 210.227.109.232 SIP/2.0
Via: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK638c731a
From: "050xxxxxxx" <sip:anonymous at localhost>;tag=as0d659f34
To: <sip:090xxxxxxx3 at 210.227.109.232>
Contact: <sip:anonymous at 211.129.117.89>
Call-ID: 3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp
CSeq: 102 INVITE
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Max-Forwards: 70
Proxy-Require: privacy
Remote-Party-ID: "050xxxxxxx"
<sip:050xxxxxxx at ocn.ne.jp>;privacy=full;screen=pass
Date: Tue, 14 Nov 2006 07:12:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 2761 2761 IN IP4 211.129.117.89
s=session
c=IN IP4 211.129.117.89
t=0 0
m=audio 17262 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 090xxxxxxx3 at 050xxxxxxx
denwa*CLI>
<-- SIP read from 210.227.109.232:5060:
SIP/2.0 100 Trying
v: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK638c731a
From: "050xxxxxxx" <sip:anonymous at localhost>;tag=as0d659f34
To: <sip:090xxxxxxx3 at 210.227.109.232>
Call-ID: 3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp
CSeq: 102 INVITE
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Date: Tue, 14 Nov 2006 07:12:44 GMT
l: 0

--- (9 headers 0 lines)---
denwa*CLI>
<-- SIP read from 210.227.109.232:5060:
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK638c731a
From: "050xxxxxxx" <sip:anonymous at localhost>;tag=as0d659f34
To: <sip:090xxxxxxx3 at 210.227.109.232>
Call-ID: 3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp
CSeq: 102 INVITE
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Date: Tue, 14 Nov 2006 07:12:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
l: 0
Proxy-Authenticate: Digest realm="nc01.ipp.biglobe.ne.jp",
domain="sip:210.227.109.232", nonce="1163
487165", opaque="", stale=FALSE, algorithm=MD5

--- (11 headers 0 lines)---
Transmitting (no NAT) to 210.227.109.232:5060:
ACK sip:090xxxxxxx3 at 210.227.109.232 SIP/2.0
Via: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK638c731a
From: "050xxxxxxx" <sip:anonymous at localhost>;tag=as0d659f34
To: <sip:090xxxxxxx3 at 210.227.109.232>
Contact: <sip:anonymous at 211.129.117.89>
Call-ID: 3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp
CSeq: 102 ACK
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Max-Forwards: 70
Proxy-Require: privacy
Remote-Party-ID: "050xxxxxxx"
<sip:050xxxxxxx at ocn.ne.jp>;privacy=full;screen=pass
Content-Length: 0

---
We're at 211.129.117.89 port 17262
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 210.227.109.232:5060:
INVITE sip:090xxxxxxx3 at 210.227.109.232 SIP/2.0
Via: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK743bca2b
From: "050xxxxxxx" <sip:anonymous at localhost>;tag=as0d659f34
To: <sip:090xxxxxxx3 at 210.227.109.232>
Contact: <sip:anonymous at 211.129.117.89>
Call-ID: 3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp
CSeq: 103 INVITE
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Max-Forwards: 70
Proxy-Require: privacy
Remote-Party-ID: "050xxxxxxx"
<sip:050xxxxxxx at ocn.ne.jp>;privacy=full;screen=pass
Proxy-Authorization: Digest username="LNRTKR4U",
realm="nc01.ipp.biglobe.ne.jp", algorithm=MD5, uri=
"sip:210.227.109.232", nonce="1163487165",
response="9260f7cbd216bba6be90f544db4c45b2", opaque=""
Date: Tue, 14 Nov 2006 07:12:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 2761 2762 IN IP4 211.129.117.89
s=session
c=IN IP4 211.129.117.89
t=0 0
m=audio 17262 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
denwa*CLI>
<-- SIP read from 210.227.109.232:5060:
SIP/2.0 100 Trying
v: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK743bca2b
From: "050xxxxxxx" <sip:anonymous at localhost>;tag=as0d659f34
To: <sip:090xxxxxxx3 at 210.227.109.232>
Call-ID: 3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp
CSeq: 103 INVITE
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Date: Tue, 14 Nov 2006 07:12:44 GMT
l: 0

--- (9 headers 0 lines)---
denwa*CLI>
<-- SIP read from 210.227.109.232:5060:
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK743bca2b
f: "050xxxxxxx" <sip:anonymous at localhost>;tag=as0d659f34
t: <sip:090xxxxxxx3 at 210.227.109.232>;tag=a0ca110d
i: 3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp
Cseq: 103 INVITE
c: application/sdp
l: 125

v=0
o=- 1 1 IN IP4 221.184.3.141
s=SIP-Call
c=IN IP4 221.184.3.141
t=0 0
m=audio 10466 RTP/AVP 0
a=rtpmap:0 PCMU/8000

--- (8 headers 7 lines)---
Found RTP audio format 0
Peer audio RTP is at port 221.184.3.141:10466
Found description format PCMU
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
-- SIP/050xxxxxxx-b715 is making progress passing it to SIP/fone2-6a06
We're at 211.129.117.89 port 10600
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 203.83.244.199:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
203.83.244.199:5060;branch=z9hG4bK335c9b00;rport;received=203.83.244.199
From: "090xxxxxxx3" <sip:090xxxxxxx3 at 203.83.244.199>;tag=as10d01d24
To: <sip:090xxxxxxx3 at 211.129.117.89>;tag=as2a60f646
Call-ID: 133bebc65adbb3f4477ee637425c5efd at 203.83.244.199
CSeq: 102 INVITE
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:090xxxxxxx3 at 211.129.117.89>
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 2761 2761 IN IP4 211.129.117.89
s=session
c=IN IP4 211.129.117.89
t=0 0
m=audio 10600 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
denwa*CLI>
<-- SIP read from 203.83.244.199:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
211.129.117.89:5060;branch=z9hG4bK14a82980;received=211.129.117.89
From: "090xxxxxxx3" <sip:090xxxxxxx3 at 211.129.117.89>;tag=as1e4d43f1
To: <sip:050xxxxxxx at fone.tsukaeru.net>;tag=as06dbf71f
Call-ID: 443e05cc04ca8b093ec7aca524e1de36 at 211.129.117.89
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:050xxxxxxx at 203.83.244.199>
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 9837 9837 IN IP4 203.83.244.199
s=session
c=IN IP4 203.83.244.199
t=0 0
m=audio 18982 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (11 headers 10 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 203.83.244.199:18982
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (te
lephone-event)
-- SIP/fone.tsukaeru.net-d53e is making progress passing it to
SIP/ECPVRSM6-7438
We're at 211.129.117.89 port 15152
Adding codec 0x4 (ulaw) to SDP
Transmitting (no NAT) to 210.227.109.232:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
210.227.109.232:5060;branch=NTTNCA00045596c66000c036c1;received=210.227.109.232
Via: SIP/2.0/UDP 10.124.237.43:5060
From: <sip:090xxxxxxx3 at 10.124.237.44;user=phone>;tag=678cbf0d
To: <sip:050xxxxxxx at 211.129.117.89:5060;user=phone>;tag=as36d84ca6
Call-ID: 678cbf0d89e103384dd000021f6 at 10.124.237.44
CSeq: 1 INVITE
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:050xxxxxxx at 211.129.117.89>
Content-Type: application/sdp
Content-Length: 162

v=0
o=root 2761 2761 IN IP4 211.129.117.89
s=session
c=IN IP4 211.129.117.89
t=0 0
m=audio 15152 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
denwa*CLI>
<-- SIP read from 210.227.109.232:5060:
CANCEL sip:050xxxxxxx at 211.129.117.89:5060 SIP/2.0
i: 678cbf0d89e103384dd000021f6 at 10.124.237.44
l: 0
f: <sip:090xxxxxxx3 at 10.124.237.44;user=phone>;tag=678cbf0d
Cseq: 1 CANCEL
t: <sip:050xxxxxxx at 211.129.117.89:5060;user=phone>
v: SIP/2.0/UDP 210.227.109.232:5060;branch=NTTNCA00045596c66000c036c1
Date: Tue, 14 Nov 2006 07:12:45 GMT

--- (8 headers 0 lines)---
Sending to 210.227.109.232 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 210.227.109.232:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
210.227.109.232:5060;branch=NTTNCA00045596c66000c036c1;received=210.227.109.232
Via: SIP/2.0/UDP 10.124.237.43:5060
From: <sip:090xxxxxxx3 at 10.124.237.44;user=phone>;tag=678cbf0d
To: <sip:050xxxxxxx at 211.129.117.89:5060;user=phone>;tag=as36d84ca6
Call-ID: 678cbf0d89e103384dd000021f6 at 10.124.237.44
CSeq: 1 INVITE
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:050xxxxxxx at 211.129.117.89>
Content-Length: 0

---
Transmitting (no NAT) to 210.227.109.232:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
210.227.109.232:5060;branch=NTTNCA00045596c66000c036c1;received=210.227.109.232
From: <sip:090xxxxxxx3 at 10.124.237.44;user=phone>;tag=678cbf0d
To: <sip:050xxxxxxx at 211.129.117.89:5060;user=phone>;tag=as36d84ca6
Call-ID: 678cbf0d89e103384dd000021f6 at 10.124.237.44
CSeq: 1 CANCEL
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:050xxxxxxx at 211.129.117.89>
Content-Length: 0

---
Reliably Transmitting (NAT) to 203.83.244.199:5060:
CANCEL sip:050xxxxxxx at fone.tsukaeru.net SIP/2.0
Via: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK14a82980
From: "090xxxxxxx3" <sip:090xxxxxxx3 at 211.129.117.89>;tag=as1e4d43f1
To: <sip:050xxxxxxx at fone.tsukaeru.net>
Contact: <sip:090xxxxxxx3 at 211.129.117.89>
Call-ID: 443e05cc04ca8b093ec7aca524e1de36 at 211.129.117.89
CSeq: 102 CANCEL
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Max-Forwards: 70
Content-Length: 0

---
Scheduling destruction of call
'443e05cc04ca8b093ec7aca524e1de36 at 211.129.117.89' in 15000 ms
== Spawn extension (fromocn, 050xxxxxxx, 6) exited non-zero on
'SIP/ECPVRSM6-7438'
-- Executing Hangup("SIP/ECPVRSM6-7438", "") in new stack
== Spawn extension (fromocn, h, 1) exited non-zero on 'SIP/ECPVRSM6-7438'
denwa*CLI>
<-- SIP read from 210.227.109.232:5060:
ACK sip:050xxxxxxx at 211.129.117.89:5060 SIP/2.0
i: 678cbf0d89e103384dd000021f6 at 10.124.237.44
l: 0
f: <sip:090xxxxxxx3 at 10.124.237.44;user=phone>;tag=678cbf0d
Cseq: 1 ACK
t: <sip:050xxxxxxx at 211.129.117.89:5060;user=phone>;tag=as36d84ca6
v: SIP/2.0/UDP 210.227.109.232:5060;branch=NTTNCA00045596c66000c036c1
Date: Tue, 14 Nov 2006 07:12:45 GMT

--- (8 headers 0 lines)---
Destroying call '678cbf0d89e103384dd000021f6 at 10.124.237.44'
denwa*CLI>
<-- SIP read from 203.83.244.199:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
211.129.117.89:5060;branch=z9hG4bK14a82980;received=211.129.117.89
From: "090xxxxxxx3" <sip:090xxxxxxx3 at 211.129.117.89>;tag=as1e4d43f1
To: <sip:050xxxxxxx at fone.tsukaeru.net>;tag=as06dbf71f
Call-ID: 443e05cc04ca8b093ec7aca524e1de36 at 211.129.117.89
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

--- (9 headers 0 lines)---
Transmitting (NAT) to 203.83.244.199:5060:
ACK sip:050xxxxxxx at fone.tsukaeru.net SIP/2.0
Via: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK14a82980
From: "090xxxxxxx3" <sip:090xxxxxxx3 at 211.129.117.89>;tag=as1e4d43f1
To: <sip:050xxxxxxx at fone.tsukaeru.net>;tag=as06dbf71f
Contact: <sip:090xxxxxxx3 at 211.129.117.89>
Call-ID: 443e05cc04ca8b093ec7aca524e1de36 at 211.129.117.89
CSeq: 102 ACK
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Max-Forwards: 70
Content-Length: 0

---
Destroying call '443e05cc04ca8b093ec7aca524e1de36 at 211.129.117.89'

<-- SIP read from 203.83.244.199:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
211.129.117.89:5060;branch=z9hG4bK14a82980;received=211.129.117.89
From: "090xxxxxxx3" <sip:090xxxxxxx3 at 211.129.117.89>;tag=as1e4d43f1
To: <sip:050xxxxxxx at fone.tsukaeru.net>;tag=as06dbf71f
Call-ID: 443e05cc04ca8b093ec7aca524e1de36 at 211.129.117.89
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:050xxxxxxx at 203.83.244.199>
Content-Length: 0

--- (10 headers 0 lines)---
Destroying call '443e05cc04ca8b093ec7aca524e1de36 at 211.129.117.89'

<-- SIP read from 203.83.244.199:5060:
CANCEL sip:090xxxxxxx3 at 211.129.117.89 SIP/2.0
Via: SIP/2.0/UDP 203.83.244.199:5060;branch=z9hG4bK335c9b00;rport
From: "090xxxxxxx3" <sip:090xxxxxxx3 at 203.83.244.199>;tag=as10d01d24
To: <sip:090xxxxxxx3 at 211.129.117.89>
Contact: <sip:090xxxxxxx3 at 203.83.244.199>
Call-ID: 133bebc65adbb3f4477ee637425c5efd at 203.83.244.199
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

--- (10 headers 0 lines)---
Sending to 203.83.244.199 : 5060 (NAT)
Reliably Transmitting (NAT) to 203.83.244.199:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
203.83.244.199:5060;branch=z9hG4bK335c9b00;received=203.83.244.199;rport=5060
From: "090xxxxxxx3" <sip:090xxxxxxx3 at 203.83.244.199>;tag=as10d01d24
To: <sip:090xxxxxxx3 at 211.129.117.89>;tag=as2a60f646
Call-ID: 133bebc65adbb3f4477ee637425c5efd at 203.83.244.199
CSeq: 102 INVITE
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:090xxxxxxx3 at 211.129.117.89>
Content-Length: 0

---
Transmitting (NAT) to 203.83.244.199:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
203.83.244.199:5060;branch=z9hG4bK335c9b00;received=203.83.244.199;rport=5060
From: "090xxxxxxx3" <sip:090xxxxxxx3 at 203.83.244.199>;tag=as10d01d24
To: <sip:090xxxxxxx3 at 211.129.117.89>;tag=as2a60f646
Call-ID: 133bebc65adbb3f4477ee637425c5efd at 203.83.244.199
CSeq: 102 CANCEL
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:090xxxxxxx3 at 211.129.117.89>
Content-Length: 0

---
Reliably Transmitting (no NAT) to 210.227.109.232:5060:
CANCEL sip:090xxxxxxx3 at 210.227.109.232 SIP/2.0
Via: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK743bca2b
From: "050xxxxxxx" <sip:anonymous at localhost>;tag=as0d659f34
To: <sip:090xxxxxxx3 at 210.227.109.232>
Contact: <sip:anonymous at 211.129.117.89>
Call-ID: 3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp
CSeq: 103 CANCEL
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Max-Forwards: 70
Proxy-Require: privacy
Remote-Party-ID: "050xxxxxxx"
<sip:050xxxxxxx at ocn.ne.jp>;privacy=full;screen=pass
Proxy-Authorization: Digest username="LNRTKR4U",
realm="nc01.ipp.biglobe.ne.jp", algorithm=MD5, uri=
"sip:210.227.109.232", nonce="1163487165",
response="7716940e4166f33f505e107c30b62d69", opaque=""
Content-Length: 0

---
Scheduling destruction of call
'3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp' in 15000 ms
== Spawn extension (macro-japan, s, 12) exited non-zero on
'SIP/fone2-6a06' in macro 'japan'
== Spawn extension (macro-japan, s, 12) exited non-zero on
'SIP/fone2-6a06' in macro 'voip'
== Spawn extension (macro-japan, s, 12) exited non-zero on 'SIP/fone2-6a06'
denwa*CLI>
<-- SIP read from 210.227.109.232:5060:
SIP/2.0 200 OK
v: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK743bca2b
From: "050xxxxxxx" <sip:anonymous at localhost>;tag=as0d659f34
To: <sip:090xxxxxxx3 at 210.227.109.232>;tag=a0ca110d
Call-ID: 3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp
CSeq: 103 CANCEL
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Remote-Party-ID: "050xxxxxxx"
<sip:050xxxxxxx at ocn.ne.jp>;privacy=full;screen=pass
Content-Length: 0
Date: Tue, 14 Nov 2006 07:12:45 GMT

--- (10 headers 0 lines)---
denwa*CLI>
<-- SIP read from 203.83.244.199:5060:
ACK sip:090xxxxxxx3 at 211.129.117.89 SIP/2.0
Via: SIP/2.0/UDP 203.83.244.199:5060;branch=z9hG4bK335c9b00;rport
From: "090xxxxxxx3" <sip:090xxxxxxx3 at 203.83.244.199>;tag=as10d01d24
To: <sip:090xxxxxxx3 at 211.129.117.89>;tag=as2a60f646
Contact: <sip:090xxxxxxx3 at 203.83.244.199>
Call-ID: 133bebc65adbb3f4477ee637425c5efd at 203.83.244.199
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

--- (10 headers 0 lines)---
Destroying call '133bebc65adbb3f4477ee637425c5efd at 203.83.244.199'
denwa*CLI>
<-- SIP read from 210.227.109.232:5060:
SIP/2.0 487 Request Terminated
v: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK743bca2b
f: "050xxxxxxx" <sip:anonymous at localhost>;tag=as0d659f34
t: <sip:090xxxxxxx3 at 210.227.109.232>;tag=a0ca110d
i: 3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp
Cseq: 103 INVITE
l: 0

--- (7 headers 0 lines)---
Transmitting (no NAT) to 210.227.109.232:5060:
ACK sip:090xxxxxxx3 at 210.227.109.232 SIP/2.0
Via: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK743bca2b
From: "050xxxxxxx" <sip:anonymous at localhost>;tag=as0d659f34
To: <sip:090xxxxxxx3 at 210.227.109.232>;tag=a0ca110d
Contact: <sip:anonymous at 211.129.117.89>
Call-ID: 3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp
CSeq: 103 ACK
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Max-Forwards: 70
Proxy-Require: privacy
Remote-Party-ID: "050xxxxxxx"
<sip:050xxxxxxx at ocn.ne.jp>;privacy=full;screen=pass
Content-Length: 0

---
Destroying call '3cea8d85269ea4d0269c629a2720af7c at ocn.ne.jp'
denwa*CLI> e
<-- SIP read from 58.88.135.159:61096:
NOTIFY sip:sip.tsukaeru.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK-e803c402
From: <sip:2202 at sip.tsukaeru.net>;tag=2a1573c85bc51449o0
To: <sip:sip.tsukaeru.net>
Call-ID: e19c2ee8-676893a9 at 192.168.0.102
CSeq: 237692 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA841-3.1.4(a)
Content-Length: 0

--- (10 headers 0 lines)---
Transmitting (NAT) to 58.88.135.159:61096:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.102:5060;branch=z9hG4bK-e803c402;received=58.88.135.159
From: <sip:2202 at sip.tsukaeru.net>;tag=2a1573c85bc51449o0
To: <sip:sip.tsukaeru.net>;tag=as7e862511
Call-ID: e19c2ee8-676893a9 at 192.168.0.102
CSeq: 237692 NOTIFY
User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
Destroying call 'e19c2ee8-676893a9 at 192.168.0.102'
denwa*CLI> exit
Executing last minute cleanups