All pastes #354465 Raw Edit
Something

public text v1 · immutable
·published 2007-02-13 23:58 UTC
Here is the scenario of the problem:

Environment
	I am sending calls for external DIDs 204.16.49.121 via SIP

	This problem is reproducible every time.
	This configuration has worked in the past.
	We did not make any changes to this setup or the switch that we are aware
of that would coincide when this stopped working.


Problem
	A user dials 6366789921 which is a DID provisioned on my switch.
	On my switch I have this DID call a phone on my network and dial
            3142674833 which is a cell phone outside my network.
	The switch rings the phone on my network and sends a call back
            out to provider at 204.16.49.121 to ring the cell phone.
	Both of these events happen as expected today.
	If I pick up the call on the phone on my network the call
            connects without problem.
	If I pick up the call on the forwarded call to the cell phone
            the call immediately terminates with fast busy.
	Here is the console output from my server which includes the error 
            I get - (P1_100001 is the phone on my network)
		<snip>
		-- Called P1_100001
    		-- Called 3142674833@provider
    		-- SIP/P1_100001-86d5 is ringing
    		-- SIP/provider-aa90 is making progress passing it to SIP/204.16.49.121-0a057668
    		-- SIP/provider-aa90 answered SIP/204.16.49.121-0a057668
    		-- Attempting native bridge of SIP/204.16.49.121-0a057668 and SIP/provider-aa90
    		-- Got SIP response 480 "Temporarily Unavailable" back from 204.16.49.121
		== Spawn extension (origination, 6366789921, 2) exited non-zero on 'SIP/204.16.49.121-0a057668'


Other Observations
	This problem happens regardless of what number I use outside my network - not just the cell number.


Here is the debug output of one of these calls:

<snip>



voice1*CLI> 

<-- SIP read from 204.16.49.121:5060: 
INVITE sip:6366789921@63.246.27.240;user=phone;npdi=yes SIP/2.0

Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKh6hpmk000gigjesa4480.1

From: <sip:6368250852@204.16.49.121;user=phone>;tag=SDe9bu301-10000000-0-274604363

To: <sip:6366789921@63.246.27.240;user=phone>

CSeq: 1 INVITE

Min-SE: 1800

Session-Expires: 1800

Supported: timer

Contact: <sip:6368250852@204.16.49.121:5060;transport=udp>

Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1

Remote-Party-ID: <sip:6368250852@192.168.35.70;user=phone>;party=calling;id-type=subscriber;privacy=off;screen=yes

Max-Forwards: 66

Content-Type: application/sdp

Content-Length: 285



v=0

o=Sonus_UAC 31958 8114 IN IP4 204.16.49.10

s=SIP Media Capabilities

c=IN IP4 204.16.49.10

t=0 0

m=audio 32848 RTP/AVP 18 0 8 100

a=rtpmap:18 G729/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-15

a=sendrecv

a=maxptime:20


 
--- (14 headers 13 lines)---
 Using INVITE request as basis request - SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1
 Sending to 204.16.49.121 : 5060 (non-NAT)
 
[Kvoice1*CLI> 
Found peer 'provider'
 Found RTP audio format 18
 Found RTP audio format 0
 Found RTP audio format 8
 Found RTP audio format 100
 Peer audio RTP is at port 204.16.49.10:32848
 Found description format G729
 Found description format PCMU
 Found description format PCMA
 Found description format telephone-event
 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
 Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
 Looking for 6366789921 in origination (domain 63.246.27.240)
 list_route: hop: <sip:6368250852@204.16.49.121:5060;transport=udp>
 Transmitting (no NAT) to 204.16.49.121:5060:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKh6hpmk000gigjesa4480.1;received=204.16.49.121

From: <sip:6368250852@204.16.49.121;user=phone>;tag=SDe9bu301-10000000-0-274604363

To: <sip:6366789921@63.246.27.240;user=phone>

Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:6366789921@63.246.27.240>

Content-Length: 0




---
 
[Kvoice1*CLI> 
    -- Executing [1;36;40mMacro[0;37;40m("[1;35;40mSIP/204.16.49.121-09fe5f18[0;37;40m", "[1;35;40morig_global[0;37;40m")
     -- Executing [1;36;40mSetAccount[0;37;40m("[1;35;40mSIP/204.16.49.121-09fe5f18[0;37;40m", "[1;35;40m6366789921[0;37;40m") in new stack
     -- Executing [1;36;40mSet[0;37;40m("[1;35;40mSIP/204.16.49.121-09fe5f18[0;37;40m", "[1;35;40mfoo=EVT00001[0;37;40m") in new stack
     -- Executing [1;36;40mSetCDRUserField[0;37;40m("[1;35;40mSIP/204.16.49.121-09fe5f18[0;37;40m", "[1;35;40mEVT00001[0;37;40m") in new stack
 
[Kvoice1*CLI> 
    -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/204.16.49.121-09fe5f18[0;37;40m", "[1;35;40mSIP/P1_100001&SIP/3142674833@provider|30[0;37;40m")
     -- Called P1_100001
 
[Kvoice1*CLI> 
We're at 63.246.27.240 port 19642
 Adding codec 0x100 (g729) to SDP
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP
 13 headers, 13 lines
 Reliably Transmitting (no NAT) to 204.16.49.121:5060:
INVITE sip:3142674833@204.16.49.121 SIP/2.0

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1080f14c;rport

From: "6368250852" <sip:6368250852@63.246.27.240>;tag=as1c2568a6

To: <sip:3142674833@204.16.49.121>

Contact: <sip:6368250852@63.246.27.240>

C
[Kvoice1*CLI> 
all-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 13 Feb 2007 18:34:06 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 289



v=0

o=root 15800 15800 IN IP4 63.246.27.240

s=session

c=IN IP4 63.246.27.240

t=0 0

m=audio 19642 RTP/AVP 18 0 8 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
     -- Called 3142674833@provider
 
[Kvoice1*CLI> 

<-- SIP read from 204.16.49.121:5060: 
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1080f14c;rport=5060

From: "6368250852" <sip:6368250852@63.246.27.240>;tag=as1c2568a6

To: <sip:3142674833@204.16.49.121>

Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240

CSeq: 102 INVITE




 
--- (6 headers 0 lines)---
 
[Kvoice1*CLI> 
    -- SIP/P1_100001-40aa is ringing
 Transmitting (no NAT) to 204.16.49.121:5060:
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKh6hpmk000gigjesa4480.1;received=204.16.49.121

From: <sip:6368250852@204.16.49.121;user=phone>;tag=SDe9bu301-10000000-0-274604363

To: <sip:6366789921@63.246.27.240;user=phone>;tag=as4c1153d2

Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:6366789921@63.246.27.240>

Content-Length: 0




---
 
[Kvoice1*CLI> 

<-- SIP read from 204.16.49.121:5060: 
SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1080f14c;rport=5060

From: "6368250852" <sip:6368250852@63.246.27.240>;tag=as1c2568a6

To: <sip:3142674833@204.16.49.121>;tag=SDchsb499-31e00005-0-480760441

Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240

CSeq: 102 INVITE

Server: DC-SIP/1.2

Supported: timer

Contact: <sip:3142674833@204.16.49.121:5060;transport=udp>

Content-Type: application/sdp

Content-Length: 264



v=0

o
[Kvoice1*CLI> 
=- 3380380448 3380380469 IN IP4 204.16.49.10

s=-

c=IN IP4 204.16.49.10

t=0 0

m=audio 50254 RTP/AVP 18 101

a=sendrecv

a=ptime:20

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=fmtp:18 annexb=no

a=silenceSupp:off - - - -


 
--- (11 headers 13 lines)---
 Found RTP audio format 18
 Found RTP audio format 101
 Peer audio RTP is at port 204.16.49.10:50254
 Found description format G729
 Found description format telephone-event
 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
 Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
     -- SIP/provider-08d8 is making progress passing it to SIP/204.16.49.121-09fe5f18
 We're at 63.246.27.240 port 11876
 Adding codec 0x100 (g729) to SDP
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP
 Transmitting (no NAT) to 204.16.49.121:5060:
SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKh6hpmk000gigjesa4480.1;received=204.16.49.121

From: <sip:6368250852@204.16.49.121;user=phone>;tag=SDe9bu301-10000000-0-274604363

To: <sip:6366789921@63.246.27.240;user=phone>;tag=as4c1153d2

Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:6366789921@63.246.27.240>

Content-Type: application/sdp

Content-Length: 289



v=0

o=root 15800 15800 IN IP4 63.246.27.240

s=session

c=IN IP4 63.246.27.240

t=0 0

m=audio 11876 RTP/AVP 18 0 8 100

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-16

a=silenceSupp:off - - - -


---
 
[Kvoice1*CLI> 

<-- SIP read from 204.16.49.121:5060: 
SIP/2.0 200 OK

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1080f14c;rport=5060

From: "6368250852" <sip:6368250852@63.246.27.240>;tag=as1c2568a6

To: <sip:3142674833@204.16.49.121>;tag=SDchsb499-31e00005-0-480760441

Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240

CSeq: 102 INVITE

Server: DC-SIP/1.2

Supported: timer

Session-Expires: 1800;refresher=uas

Contact: <sip:3142674833@204.16.49.121:5060;transport=udp>

Content-Type: application/sdp

Content-Length: 264



v=0

o=- 3380380448 3380380469 IN IP4 204.16.49.10

s=-

c=IN IP4 204.16.49.10

t=0 0

m=audio 50254 RTP/AVP 18 101

a=sendrecv

a=ptime:20

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=fmtp:18 annexb=no

a=silenceSupp:off - - - -


 
--- (12 headers 13 lines)---
 Found RTP audio format 18
 Found RTP audio format 101
 Peer audio RTP is at port 204.16.49.10:50254
 Found description format G729
 Found description format telephone-event
 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
 Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
 list_route: hop: <sip:3142674833@204.16.49.121:5060;transport=udp>
 set_destination: Parsing <sip:3142674833@204.16.49.121:5060;transport=udp> for address/port to send to
 set_destination: set destination to 204.16.49.121, port 5060
 Transmitting (no NAT) to 204.16.49.121:5060:
ACK sip:3142674833@204.16.49.121:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK2b003b61;rport

From: "6368250852" <sip:6368250852@63.246.27.240>;tag=as1c2568a6

To: <sip:3142674833@204.16.49.121>;tag=SDchsb499-31e00005-0-480760441

Contact: <sip:6368250852@63.246.27.240>

Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---
     -- SIP/provider-08d8 answered SIP/204.16.49.121-09fe5f18
 We're at 63.246.27.240 port 11876
 Adding codec 0x100 (g729) to SDP
 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 204.16.49.121:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKh6hpmk000gigjesa4480.1;received=204.16.49.121

From: <sip:6368250852@204.16.49.121;user=phone>;tag=SDe9bu301-10000000-0-274604363

To: <sip:6366789921@63.246.27.240;user=phone>;tag=as4c1153d2

Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:6366789921@63.246.27.240>

Content-Type: application/sdp

Content-Length: 289



v=0

o=root 15800 15801 IN IP4 63.246.27.240

s=session

c=IN IP4 63.246.27.240

t=0 0

m=audio 11876 RTP/AVP 18 0 8 100

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-16

a=silenceSupp:off - - - -


---
     -- Attempting native bridge of SIP/204.16.49.121-09fe5f18 and SIP/provider-08d8
 set_destination: Parsing <sip:3142674833@204.16.49.121:5060;transport=udp> for address/port to send to
 set_destination: set destination to 204.16.49.121, port 5060
 We're at 63.246.27.240 port 19642
 Adding codec 0x100 (g729) to SDP
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP
 13 headers, 13 lines
 Reliably Transmitting (no NAT) to 204.16.49.121:5060:
INVITE sip:3142674833@204.16.49.121:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK0446fd3c;rport

From: "6368250852" <sip:6368250852@63.246.27.240>;tag=as1c2568a6

To: <sip:3142674833@204.16.49.121>;tag=SDchsb499-31e00005-0-480760441

Contact: <sip:6368250852@63.246.27.240>

Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240

CSeq: 103 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

X-asterisk-info: SIP re-invite (RTP bridge)

Content-Type: application/sdp

Content-Length: 287



v=0

o=root 15800 15801 IN IP4 204.16.49.10

s=session

c=IN IP4 204.16.49.10

t=0 0

m=audio 32848 RTP/AVP 18 0 8 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kvoice1*CLI> 

<-- SIP read from 204.16.49.121:5060: 
ACK sip:6366789921@63.246.27.240 SIP/2.0

Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKu23a89206o4h9dk8g5c1.1

Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1

From: <sip:6368250852@204.16.49.121;user=phone>;tag=SDe9bu301-10000000-0-274604363

To: <sip:6366789921@63.246.27.240;user=phone>;tag=as4c1153d2

CSeq: 1 ACK

Contact: <sip:6368250852@204.16.49.121:5060;transport=udp>

Supported: timer

Content-Length: 0

Max-Forwards: 66




 
--- (10 headers 0 lines)---
 set_destination: Parsing <sip:6368250852@204.16.49.121:5060;transport=udp> for address/port to send to
 set_destination: set destination to 204.16.49.121, port 5060
 We're at 63.246.27.240 port 11876
 Adding codec 0x100 (g729) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP
 13 headers, 11 lines
 Reliably Transmitting (no NAT) to 204.16.49.121:5060:
INVITE sip:6368250852@204.16.49.121:5060 SIP/2.0

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1bb72d82;rport

From: <sip:6366789921@63.246.27.240;user=phone>;tag=as4c1153d2

To: <sip:6368250852@204.16.49.121;user=phone>;tag=SDe9bu301-10000000-0-274604363

Contact: <sip:6366789921@63.246.27.240>

Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

X-asterisk-info: SIP re-invite (RTP bridge)

Content-Type: application/sdp

Content-Length: 239



v=0

o=root 15800 15802 IN IP4 204.16.49.10

s=session

c
[Kvoice1*CLI> 
=IN IP4 204.16.49.10

t=0 0

m=audio 50254 RTP/AVP 18 100

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-16

a=silenceSupp:off - - - -


---
 
[Kvoice1*CLI> 

<-- SIP read from 204.16.49.121:5060: 
SIP/2.0 200 OK

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK0446fd3c;rport=5060

From: "6368250852" <sip:6368250852@63.246.27.240>;tag=as1c2568a6

To: <sip:3142674833@204.16.49.121>;tag=SDchsb499-31e00005-0-480760441

Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240

CSeq: 103 INVITE

Server: DC-SIP/1.2

Supported: timer

Session-Expires: 1800;refresher=uas

Contact: <sip:3142674833@204.16.49.121:5060;transport=udp>

Content-Type: application/sdp

Content-Length: 264



v=0

o=- 3380380453 3380380527 IN IP4 204.16.49.10

s=-

c=IN IP4 204.16.49.10

t=0 0

m=audio 50254 RTP/AVP 18 101

a=sendrecv

a=ptime:20

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=fmtp:18 annexb=no

a=silenceSupp:off - - - -


 
--- (12 headers 13 lines)---
 Found RTP audio format 18
 Found RTP audio format 101
 Peer audio RTP is at port 204.16.49.10:50254
 Found description format G729
 Found description format telephone-event
 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
 Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
 set_destination: Parsing <sip:3142674833@204.16.49.121:5060;transport=udp> for address/port to send to
 set_destination: set destination to 204.16.49.121, port 5060
 Transmitting (no NAT) to 204.16.49.121:5060:
ACK sip:3142674833@204.16.49.121:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK76c43541;rport

From: "6368250852" <sip:6368250852@63.246.27.240>;tag=as1c2568a6

To: <sip:3142674833@204.16.49.121>;tag=SDchsb499-31e00005-0-480760441

Contact: <sip:6368250852@63.246.27.240>

Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240

CSeq: 103 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---
 
[Kvoice1*CLI> 

<-- SIP read from 204.16.49.121:5060: 
SIP/2.0 480 Temporarily Unavailable

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1bb72d82;rport=5060

From: <sip:6366789921@63.246.27.240;user=phone>;tag=as4c1153d2

To: <sip:6368250852@204.16.49.121;user=phone>;tag=SDe9bu301-10000000-0-274604363

Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1

CSeq: 102 INVITE

Server: DC-SIP/1.2

Supported: timer

Contact: <sip:6368250852@204.16.49.121>

Content-Length: 0




 
--- (10 headers 0 lines)---
     -- Got SIP response 480 "Temporarily Unavailable" back from 204.16.49.121
 set_destination: Parsing <sip:6368250852@204.16.49.121:5060;transport=udp> for address/port to send to
 set_destination: set destination to 204.16.49.121, port 5060
 Transmitting (no NAT) to 204.16.49.121:5060:
ACK sip:6368250852@204.16.49.121:5060 SIP/2.0

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1bb72d82;rport

From: <sip:6366789921@63.246.27.240;user=phone>;tag=as4c1153d2

To: <sip:6368250852@204.16.49.121;user=phone>;tag=SDe9bu301-10000000-0-274604363

Contact: <sip:6366789921@63.246.27.240>

Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---
 
[Kvoice1*CLI> 
set_destination: Parsing <sip:3142674833@204.16.49.121:5060;transport=udp> for address/port to send to
 set_destination: set destination to 204.16.49.121, port 5060
 Reliably Transmitting (no NAT) to 204.16.49.121:5060:
BYE sip:3142674833@204.16.49.121:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK7d5b1c1c;rport

From: "6368250852" <sip:6368250852@63.246.27.240>;tag=as1c2568a6

To: <sip:3142674833@204.16.49.121>;tag=SDchsb499-31e00005-0-480760441

Contact: <sip:6368250852@63.246.27.240>

Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240

CSeq: 104 BYE

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




---
   == Spawn extension (origination, 6366789921, 2) exited non-zero on 'SIP/204.16.49.121-09fe5f18'
 
[Kvoice1*CLI> 

<-- SIP read from 204.16.49.121:5060: 
BYE sip:6366789921@63.246.27.240 SIP/2.0

Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKu23a89206o4h9dk8g5c1cd0000010.1

Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1

From: <sip:6368250852@204.16.49.121;user=phone>;tag=SDe9bu301-10000000-0-274604363

To: <sip:6366789921@63.246.27.240;user=phone>;tag=as4c1153d2

CSeq: 2 BYE

Supported: timer

Content-Length: 0

Max-Forwards: 69




 
--- (9 headers 0 lines)---
 Sending to 204.16.49.121 : 5060 (non-NAT)
 Transmitting (no NAT) to 204.16.49.121:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKu23a89206o4h9dk8g5c1cd0000010.1;received=204.16.49.121

From: <sip:6368250852@204.16.49.121;user=phone>;tag=SDe9bu301-10000000-0-274604363

To: <sip:6366789921@63.246.27.240;user=phone>;tag=as4c1153d2

Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1

CSeq: 2 BYE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:6366789921@63.246.27.240>

Content-Length: 0

X-Asterisk-HangupCause: Normal Clearing




---
 
[Kvoice1*CLI> 

<-- SIP read from 204.16.49.121:5060: 
SIP/2.0 200 OK

Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK7d5b1c1c;rport=5060

From: "6368250852" <sip:6368250852@63.246.27.240>;tag=as1c2568a6

To: <sip:3142674833@204.16.49.121>;tag=SDchsb499-31e00005-0-480760441

Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240

CSeq: 104 BYE

Server: DC-SIP/1.2

Supported: timer

Content-Length: 0




 
--- (9 headers 0 lines)---
 Destroying call '7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240'
 Destroying call 'SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1'
 
[Kvoice1*CLI> 

voice1*CLI> 

voice1*CLI>