All pastes #167687 Raw Edit
Something

public text v1 · immutable
·published 2006-09-12 06:52 UTC
pbx*CLI> 

<-- SIP read from 70.34.104.240:5060: 
INVITE sip:*0@pbx.contractcsi.com SIP/2.0

Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-526eeb72

From: "Michael Azarian" <sip:30@pbx.contractcsi.com>;tag=103a80113b508f1ao0

To: <sip:*0@pbx.contractcsi.com>

Call-ID: 1229c09d-f20167fe@192.168.0.248

CSeq: 101 INVITE

Max-Forwards: 70

Contact: "Michael Azarian" <sip:30@192.168.0.248:5060>

Expires: 240

User-Agent: Sipura/SPA942-4.1.10(e)

Content-Length: 399

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

Content-Type: application/sdp



v=0

o=- 6210936 6210936 IN IP4 192.168.0.248

s=-

c=IN IP4 192.168.0.248

t=0 0

m=audio 16470 RTP/AVP 0 2 4 8 18 96 97 98 101

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:4 G723/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729a/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv


 
--- (13 headers 18 lines)---
 Using INVITE request as basis request - 1229c09d-f20167fe@192.168.0.248
 Sending to 192.168.0.248 : 5060 (non-NAT)
 Reliably Transmitting (NAT) to 70.34.104.240:5060:
SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-526eeb72;received=70.34.104.240

From: "Michael Azarian" <sip:30@pbx.contractcsi.com>;tag=103a80113b508f1ao0

To: <sip:*0@pbx.contractcsi.com>;tag=as7456533d

Call-ID: 1229c09d-f20167fe@192.168.0.248

CSeq: 101 INVITE

User-Agent: Asterisk PBX

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

Contact: <sip:*0@70.167.155.210>

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6871e45d"

Content-Length: 0




---
 Scheduling destruction of call '1229c09d-f20167fe@192.168.0.248' in 15000 ms
 Found user '30'
 
[Kpbx*CLI> 

<-- SIP read from 70.34.104.240:5060: 
ACK sip:*0@pbx.contractcsi.com SIP/2.0

Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-526eeb72

From: "Michael Azarian" <sip:30@pbx.contractcsi.com>;tag=103a80113b508f1ao0

To: <sip:*0@pbx.contractcsi.com>;tag=as7456533d

Call-ID: 1229c09d-f20167fe@192.168.0.248

CSeq: 101 ACK

Max-Forwards: 70

Contact: "Michael Azarian" <sip:30@192.168.0.248:5060>

User-Agent: Sipura/SPA942-4.1.10(e)

Content-Length: 0




 
--- (10 headers 0 lines)---
 
[Kpbx*CLI> 

<-- SIP read from 70.34.104.240:5060: 
INVITE sip:*0@pbx.contractcsi.com SIP/2.0

Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-e963f67e

From: "Michael Azarian" <sip:30@pbx.contractcsi.com>;tag=103a80113b508f1ao0

To: <sip:*0@pbx.contractcsi.com>

Call-ID: 1229c09d-f20167fe@192.168.0.248

CSeq: 102 INVITE

Max-Forwards: 70

Proxy-Authorization: Digest username="30",realm="asterisk",nonce="6871e45d",uri="sip:*0@pbx.contractcsi.com",algorithm=MD5,response="ebcabeaeccb17ac09e4a86dd9f13f5ff"

Contact: "Michael Azarian" <sip:30@192.168.0.248:5060>

Expires: 240

User-Agent: Sipura/SPA942-4.1.10(e)

Content-Length: 399

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

Content-Type: application/sdp



v=0

o=- 6210936 6210936 IN IP4 192.168.0.248

s=-

c=IN IP4 192.168.0.248

t=0 0

m=audio 16470 RTP/AVP 0 2 4 8 18 96 97 98 101

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:4 G723/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729a/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000


[Kpbx*CLI> 
a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv


 
--- (14 headers 18 lines)---
 Using INVITE request as basis request - 1229c09d-f20167fe@192.168.0.248
 Sending to 192.168.0.248 : 5060 (NAT)
 Found user '30'
 Found RTP audio format 0
 Found RTP audio format 2
 
[Kpbx*CLI> 
Found RTP audio format 4
 Found RTP audio format 8
 Found RTP audio format 18
 Found RTP audio format 96
 Found RTP audio format 97
 Found RTP audio format 98
 Found RTP audio format 101
 Peer audio RTP is at port 192.168.0.248:16470
 Found description format PCMU
 Found description format G726-32
 Found description format G723
 Found description format PCMA
 Found description format G729a
 Found description format G726-40
 Found description format G726-24
 Found description format G726-16
 Found description format telephone-event
 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
 Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
 Looking for *0 in internal-admin (domain pbx.contractcsi.com)
 list_route: hop: <sip:30@192.168.0.248:5060>
 Transmitting (NAT) to 70.34.104.240:5060:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-e963f67e;received=70.34.104.240

From: "Michael Azarian" <sip:30@pbx.contractcsi.com>;tag=103a80113b508f1ao0

To: <sip:*0@pbx.contractcsi.com>

Call-ID: 1229c09d-f20167fe@192.168.0.248

CSeq: 102 INVITE

User-Agent: Asterisk PBX

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

Contact: <sip:*0@70.167.155.210>

Content-Length: 0




---
 
[Kpbx*CLI> 
    -- Executing [1;36;40mSet[0;37;40m("[1;35;40mSIP/30-093a52d0[0;37;40m", "[1;35;40mTIMEOUT(absolute) = 25[0;37;40m") in new stack
     -- Executing [1;36;40mAGI[0;37;40m("[1;35;40mSIP/30-093a52d0[0;37;40m", "[1;35;40mpage.agi[0;37;40m") in new stack
     -- Launched AGI Script /var/lib/asterisk/agi-bin/page.agi
 
[Kpbx*CLI> 
    -- Remote UNIX connection
 
[Kpbx*CLI> 
    -- Remote UNIX connection disconnected
 
[Kpbx*CLI> 
    -- AGI Script page.agi completed, returning 0
     -- Executing [1;36;40mSetCallerID[0;37;40m("[1;35;40mSIP/30-093a52d0[0;37;40m", "[1;35;40m"PAGE: Michael Azarian" <[0;37;40m") in new stack
     -- Executing [1;36;40mSet[0;37;40m("[1;35;40mSIP/30-093a52d0[0;37;40m", "[1;35;40m_ALERT_INFO="Ring Answer"[0;37;40m") in new stack
     -- Executing [1;36;40mSIPAddHeader[0;37;40m("[1;35;40mSIP/30-093a52d0[0;37;40m", "[1;35;40mCall-Info: <sip:192.168.16.50>;answer-after=0[0;37;40m") in new stack
     -- Executing [1;36;40mPage[0;37;40m("[1;35;40mSIP/30-093a52d0[0;37;40m", "[1;35;40mSIP/18&SIP/17&SIP/16&SIP/13&SIP/12&SIP/11[0;37;40m") in new stack
 
[Kpbx*CLI> 
We're at 70.167.155.210 port 10132
 
[Kpbx*CLI> 
Adding codec 0x4 (ulaw) to SDP
 
[Kpbx*CLI> 
Adding codec 0x8 (alaw) to SDP
 
[Kpbx*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP
 
[Kpbx*CLI> 
Transmitting (NAT) to 70.34.104.240:5060:
SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-e963f67e;received=70.34.104.240

From: "Michael Azarian" <sip:30@pbx.contractcsi.com>;tag=103a80113b508f1ao0

To: <sip:*0@pbx.contractcsi.com>;tag=as2456f08e

Call-ID: 1229c09d-f20167fe@192.168.0.248

CSeq: 102 INVITE

User-Agent: Asterisk PBX

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

Contact: <sip:*0@70.167.155.210>

Content-Type: application/sdp

Content-Length: 242



v=0

o=root 2501 2501 IN IP4 70.167.155.210

s=session

c=IN IP4 70.167.155.210

t=0 0

m=audio 10132 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 - - - -


---
 
[Kpbx*CLI> 
    -- Playing 'beep' (language 'en')
 
[Kpbx*CLI> 
We're at 192.168.16.50 port 18620
 
[Kpbx*CLI> 
Adding codec 0x4 (ulaw) to SDP
 
[Kpbx*CLI> 
Adding codec 0x8 (alaw) to SDP
 
[Kpbx*CLI> 
Adding codec 0x2 (gsm) to SDP
 
[Kpbx*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP
 
[Kpbx*CLI> 
15 headers, 12 lines
 
[Kpbx*CLI> 
Reliably Transmitting (no NAT) to 192.168.16.80:5060:
INVITE sip:18@192.168.16.80:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as20cf0eac

To: <sip:18@192.168.16.80:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 18620 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
We're at 192.168.16.50 port 13810
 
[Kpbx*CLI> 
Adding codec 0x4 (ulaw) to SDP
 
[Kpbx*CLI> 
Adding codec 0x8 (alaw) to SDP
 
[Kpbx*CLI> 
Adding codec 0x2 (gsm) to SDP
 
[Kpbx*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP
 
[Kpbx*CLI> 
15 headers, 12 lines
 
[Kpbx*CLI> 
Reliably Transmitting (no NAT) to 192.168.16.81:5060:
INVITE sip:17@192.168.16.81:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as295fd97d

To: <sip:17@192.168.16.81:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 13810 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
We're at 192.168.16.50 port 19550
 
[Kpbx*CLI> 
Adding codec 0x4 (ulaw) to SDP
 
[Kpbx*CLI> 
Adding codec 0x8 (alaw) to SDP
 
[Kpbx*CLI> 
Adding codec 0x2 (gsm) to SDP
 
[Kpbx*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP
 
[Kpbx*CLI> 
15 headers, 12 lines
 
[Kpbx*CLI> 
Reliably Transmitting (NAT) to 192.168.16.13:5060:
INVITE sip:13@192.168.16.13:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as335b3254

To: <sip:13@192.168.16.13:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 19550 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
We're at 192.168.16.50 port 16732
 
[Kpbx*CLI> 
Adding codec 0x4 (ulaw) to SDP
 
[Kpbx*CLI> 
Adding codec 0x8 (alaw) to SDP
 
[Kpbx*CLI> 
Adding codec 0x2 (gsm) to SDP
 
[Kpbx*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP
 
[Kpbx*CLI> 
15 headers, 12 lines
 
[Kpbx*CLI> 
Reliably Transmitting (NAT) to 192.168.16.72:5060:
INVITE sip:12@192.168.16.72:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as3a3f0c93

To: <sip:12@192.168.16.72:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 16732 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
We're at 192.168.16.50 port 14124
 
[Kpbx*CLI> 
Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding codec 0x2 (gsm) to SDP
 
[Kpbx*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP
 15 headers, 12 lines
 
[Kpbx*CLI> 
Reliably Transmitting (NAT) to 192.168.16.84:5060:
INVITE sip:11@192.168.16.84:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as5beed5a7

To: <sip:11@192.168.16.84:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 14124 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 We're at 192.168.16.50 port 17244
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding codec 0x2 (gsm) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP
 15 headers, 12 lines
 
[Kpbx*CLI> 
Reliably Transmitting (NAT) to 192.168.16.16:5060:
INVITE sip:16@192.168.16.16:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as06705b60

To: <sip:16@192.168.16.16:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 17244 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #1 (NAT) to 192.168.16.84:5060:
INVITE sip:11@192.168.16.84:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as5beed5a7

To: <sip:11@192.168.16.84:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 14124 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #1 (no NAT) to 192.168.16.80:5060:
INVITE sip:18@192.168.16.80:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as20cf0eac

To: <sip:18@192.168.16.80:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 18620 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #1 (NAT) to 192.168.16.13:5060:
INVITE sip:13@192.168.16.13:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as335b3254

To: <sip:13@192.168.16.13:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 19550 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #1 (NAT) to 192.168.16.72:5060:
INVITE sip:12@192.168.16.72:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as3a3f0c93

To: <sip:12@192.168.16.72:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 16732 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #1 (no NAT) to 192.168.16.81:5060:
INVITE sip:17@192.168.16.81:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as295fd97d

To: <sip:17@192.168.16.81:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 13810 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #1 (NAT) to 192.168.16.16:5060:
INVITE sip:16@192.168.16.16:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as06705b60

To: <sip:16@192.168.16.16:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 17244 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #2 (NAT) to 192.168.16.84:5060:
INVITE sip:11@192.168.16.84:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as5beed5a7

To: <sip:11@192.168.16.84:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 14124 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #2 (NAT) to 192.168.16.13:5060:
INVITE sip:13@192.168.16.13:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as335b3254

To: <sip:13@192.168.16.13:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 19550 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #2 (no NAT) to 192.168.16.80:5060:
INVITE sip:18@192.168.16.80:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as20cf0eac

To: <sip:18@192.168.16.80:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 18620 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #2 (NAT) to 192.168.16.72:5060:
INVITE sip:12@192.168.16.72:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as3a3f0c93

To: <sip:12@192.168.16.72:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 16732 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #2 (NAT) to 192.168.16.16:5060:
INVITE sip:16@192.168.16.16:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as06705b60

To: <sip:16@192.168.16.16:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 17244 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #2 (no NAT) to 192.168.16.81:5060:
INVITE sip:17@192.168.16.81:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as295fd97d

To: <sip:17@192.168.16.81:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 13810 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #3 (NAT) to 192.168.16.84:5060:
INVITE sip:11@192.168.16.84:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as5beed5a7

To: <sip:11@192.168.16.84:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 14124 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #3 (NAT) to 192.168.16.13:5060:
INVITE sip:13@192.168.16.13:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as335b3254

To: <sip:13@192.168.16.13:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 19550 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #3 (no NAT) to 192.168.16.80:5060:
INVITE sip:18@192.168.16.80:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as20cf0eac

To: <sip:18@192.168.16.80:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 18620 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #3 (NAT) to 192.168.16.72:5060:
INVITE sip:12@192.168.16.72:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as3a3f0c93

To: <sip:12@192.168.16.72:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 16732 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #3 (NAT) to 192.168.16.16:5060:
INVITE sip:16@192.168.16.16:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as06705b60

To: <sip:16@192.168.16.16:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 17244 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #3 (no NAT) to 192.168.16.81:5060:
INVITE sip:17@192.168.16.81:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as295fd97d

To: <sip:17@192.168.16.81:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 13810 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #4 (NAT) to 192.168.16.84:5060:
INVITE sip:11@192.168.16.84:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as5beed5a7

To: <sip:11@192.168.16.84:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 14124 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #4 (NAT) to 192.168.16.13:5060:
INVITE sip:13@192.168.16.13:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as335b3254

To: <sip:13@192.168.16.13:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 19550 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #4 (no NAT) to 192.168.16.80:5060:
INVITE sip:18@192.168.16.80:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as20cf0eac

To: <sip:18@192.168.16.80:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 18620 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #4 (NAT) to 192.168.16.72:5060:
INVITE sip:12@192.168.16.72:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as3a3f0c93

To: <sip:12@192.168.16.72:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 16732 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #4 (NAT) to 192.168.16.16:5060:
INVITE sip:16@192.168.16.16:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as06705b60

To: <sip:16@192.168.16.16:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 17244 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #4 (no NAT) to 192.168.16.81:5060:
INVITE sip:17@192.168.16.81:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as295fd97d

To: <sip:17@192.168.16.81:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 13810 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #5 (NAT) to 192.168.16.84:5060:
INVITE sip:11@192.168.16.84:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as5beed5a7

To: <sip:11@192.168.16.84:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 14124 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #5 (NAT) to 192.168.16.13:5060:
INVITE sip:13@192.168.16.13:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as335b3254

To: <sip:13@192.168.16.13:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 19550 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #5 (no NAT) to 192.168.16.80:5060:
INVITE sip:18@192.168.16.80:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as20cf0eac

To: <sip:18@192.168.16.80:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 18620 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #5 (NAT) to 192.168.16.72:5060:
INVITE sip:12@192.168.16.72:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as3a3f0c93

To: <sip:12@192.168.16.72:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 16732 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 Retransmitting #5 (NAT) to 192.168.16.16:5060:
INVITE sip:16@192.168.16.16:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as06705b60

To: <sip:16@192.168.16.16:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 17244 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #5 (no NAT) to 192.168.16.81:5060:
INVITE sip:17@192.168.16.81:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as295fd97d

To: <sip:17@192.168.16.81:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 13810 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
We're at 70.167.155.210 port 10132
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP
 Reliably Transmitting (NAT) to 70.34.104.240:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-e963f67e;received=70.34.104.240

From: "Michael Azarian" <sip:30@pbx.contractcsi.com>;tag=103a80113b508f1ao0

To: <sip:*0@pbx.contractcsi.com>;tag=as2456f08e

Call-ID: 1229c09d-f20167fe@192.168.0.248

CSeq: 102 INVITE

User-Agent: Asterisk PBX

A
[Kpbx*CLI> 
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:*0@70.167.155.210>

Content-Type: application/sdp

Content-Length: 242



v=0

o=root 2501 2502 IN IP4 70.167.155.210

s=session

c=IN IP4 70.167.155.210

t=0 0

m=audio 10132 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 - - - -


---
     -- Created MeetMe conference 1023 for conference '2093530406d'
 
[Kpbx*CLI> 

<-- SIP read from 70.34.104.240:5060: 
ACK sip:*0@70.167.155.210 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-b45543d3

From: "Michael Azarian" <sip:30@pbx.contractcsi.com>;tag=103a80113b508f1ao0

To: <sip:*0@pbx.contractcsi.com>;tag=as2456f08e

Call-ID: 1229c09d-f20167fe@192.168.0.248

CSeq: 102 ACK

Max-Forwards: 70

Proxy-Authorization: Digest username="30",realm="asterisk",nonce="6871e45d",uri="sip:*0@70.167.155.210",algorithm=MD5,response="1ba3e86baa691428ac0899e060178a87"

Contact: "Michael Azarian" <sip:30@192.168.0.248:5060>

User-Agent: Sipura/SPA942-4.1.10(e)

Content-Length: 0




 
--- (11 headers 0 lines)---
 
[Kpbx*CLI> 
Retransmitting #6 (NAT) to 192.168.16.84:5060:
INVITE sip:11@192.168.16.84:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as5beed5a7

To: <sip:11@192.168.16.84:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 14124 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #6 (NAT) to 192.168.16.13:5060:
INVITE sip:13@192.168.16.13:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as335b3254

To: <sip:13@192.168.16.13:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 19550 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #6 (no NAT) to 192.168.16.80:5060:
INVITE sip:18@192.168.16.80:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as20cf0eac

To: <sip:18@192.168.16.80:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 18620 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #6 (NAT) to 192.168.16.72:5060:
INVITE sip:12@192.168.16.72:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as3a3f0c93

To: <sip:12@192.168.16.72:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 16732 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #6 (NAT) to 192.168.16.16:5060:
INVITE sip:16@192.168.16.16:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as06705b60

To: <sip:16@192.168.16.16:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 17244 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Retransmitting #6 (no NAT) to 192.168.16.81:5060:
INVITE sip:17@192.168.16.81:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport

From: "PAGE: Michael Azarian" <" <sip:CSI@192.168.16.50>;tag=as295fd97d

To: <sip:17@192.168.16.81:5060>

Contact: <sip:CSI@192.168.16.50>

Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 12 Sep 2006 06:50:22 GMT

Alert-Info: "Ring Answer"

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

Call-Info: <sip:192.168.16.50>;answer-after=0

Content-Type: application/sdp

Content-Length: 263



v=0

o=root 2501 2501 IN IP4 192.168.16.50

s=session

c=IN IP4 192.168.16.50

t=0 0

m=audio 13810 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
 
[Kpbx*CLI> 
Sep 11 23:50:23 [1;31;40mWARNING[0;37;40m[2541]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m1217[0;37;40m [1;37;40mretrans_pkt[0;37;40m:  Maximum retries exceeded on transmission 4359e5a262f4e0773c88216517033e67@192.168.16.50 for seqno 102 (Critical Request)
 
[Kpbx*CLI> 
Sep 11 23:50:23 [1;31;40mWARNING[0;37;40m[2541]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m1234[0;37;40m [1;37;40mretrans_pkt[0;37;40m:  Hanging up call 4359e5a262f4e0773c88216517033e67@192.168.16.50 - no reply to our critical packet.
 
[Kpbx*CLI> 
Sep 11 23:50:23 [1;31;40mWARNING[0;37;40m[2541]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m1217[0;37;40m [1;37;40mretrans_pkt[0;37;40m:  Maximum retries exceeded on transmission 22caddf1659a1715126ea5a71869692e@192.168.16.50 for seqno 102 (Critical Request)
 
[Kpbx*CLI> 
Sep 11 23:50:23 [1;31;40mWARNING[0;37;40m[2541]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m1234[0;37;40m [1;37;40mretrans_pkt[0;37;40m:  Hanging up call 22caddf1659a1715126ea5a71869692e@192.168.16.50 - no reply to our critical packet.
 
[Kpbx*CLI> 
Destroying call '4359e5a262f4e0773c88216517033e67@192.168.16.50'
 
[Kpbx*CLI> 
Sep 11 23:50:24 [1;31;40mWARNING[0;37;40m[2541]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m1217[0;37;40m [1;37;40mretrans_pkt[0;37;40m:  Maximum retries exceeded on transmission 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50 for seqno 102 (Critical Request)
 
[Kpbx*CLI> 
Sep 11 23:50:24 [1;31;40mWARNING[0;37;40m[2541]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m1234[0;37;40m [1;37;40mretrans_pkt[0;37;40m:  Hanging up call 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50 - no reply to our critical packet.
 
[Kpbx*CLI> 
Destroying call '22caddf1659a1715126ea5a71869692e@192.168.16.50'
 
[Kpbx*CLI> 
Destroying call '5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50'
 
[Kpbx*CLI> 
Sep 11 23:50:24 [1;31;40mWARNING[0;37;40m[2541]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m1217[0;37;40m [1;37;40mretrans_pkt[0;37;40m:  Maximum retries exceeded on transmission 226cd87216c32baf3e48049921efe777@192.168.16.50 for seqno 102 (Critical Request)
 
[Kpbx*CLI> 
Sep 11 23:50:24 [1;31;40mWARNING[0;37;40m[2541]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m1234[0;37;40m [1;37;40mretrans_pkt[0;37;40m:  Hanging up call 226cd87216c32baf3e48049921efe777@192.168.16.50 - no reply to our critical packet.
 
[Kpbx*CLI> 
Sep 11 23:50:24 [1;31;40mWARNING[0;37;40m[2541]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m1217[0;37;40m [1;37;40mretrans_pkt[0;37;40m:  Maximum retries exceeded on transmission 0603149d18b7109627d0efe84987bbb5@192.168.16.50 for seqno 102 (Critical Request)
 
[Kpbx*CLI> 
Sep 11 23:50:24 [1;31;40mWARNING[0;37;40m[2541]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m1234[0;37;40m [1;37;40mretrans_pkt[0;37;40m:  Hanging up call 0603149d18b7109627d0efe84987bbb5@192.168.16.50 - no reply to our critical packet.
 
[Kpbx*CLI> 
Destroying call '0603149d18b7109627d0efe84987bbb5@192.168.16.50'
 
[Kpbx*CLI> 
Destroying call '226cd87216c32baf3e48049921efe777@192.168.16.50'
 
[Kpbx*CLI> 
Sep 11 23:50:24 [1;31;40mWARNING[0;37;40m[2541]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m1217[0;37;40m [1;37;40mretrans_pkt[0;37;40m:  Maximum retries exceeded on transmission 11bafa99125866ac60c082113ef7d971@192.168.16.50 for seqno 102 (Critical Request)
 
[Kpbx*CLI> 
Sep 11 23:50:24 [1;31;40mWARNING[0;37;40m[2541]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m1234[0;37;40m [1;37;40mretrans_pkt[0;37;40m:  Hanging up call 11bafa99125866ac60c082113ef7d971@192.168.16.50 - no reply to our critical packet.
 
[Kpbx*CLI> 
Destroying call '11bafa99125866ac60c082113ef7d971@192.168.16.50'
 
[Kpbx*CLI> 

<-- SIP read from 70.34.104.240:5060: 
BYE sip:*0@70.167.155.210 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-d2c5d1ac

From: "Michael Azarian" <sip:30@pbx.contractcsi.com>;tag=103a80113b508f1ao0

To: <sip:*0@pbx.contractcsi.com>;tag=as2456f08e

Call-ID: 1229c09d-f20167fe@192.168.0.248

CSeq: 103 BYE

Max-Forwards: 70

Proxy-Authorization: Digest username="30",realm="asterisk",nonce="6871e45d",uri="sip:*0@70.167.155.210",algorithm=MD5,response="e4ada15de33eaa86a331e23546f25827"

User-Agent: Sipura/SPA942-4.1.10(e)

Content-Length: 0




 
[Kpbx*CLI> 

--- (10 headers 0 lines)---
 
[Kpbx*CLI> 
Sending to 192.168.0.248 : 5060 (NAT)
 
[Kpbx*CLI> 
Transmitting (NAT) to 70.34.104.240:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-d2c5d1ac;received=70.34.104.240

From: "Michael Azarian" <sip:30@pbx.contractcsi.com>;tag=103a80113b508f1ao0

To: <sip:*0@pbx.contractcsi.com>;tag=as2456f08e

Call-ID: 1229c09d-f20167fe@192.168.0.248

CSeq: 103 BYE

User-Agent: Asterisk PBX

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

Contact: <sip:*0@70.167.155.210>

Content-Length: 0




---
 
[Kpbx*CLI> 
    -- Hungup 'Zap/pseudo-1544701011'
 
[Kpbx*CLI> 
  == Spawn extension (internal-admin, *0, 6) exited non-zero on 'SIP/30-093a52d0'
 
[Kpbx*CLI> 
    -- Executing [1;36;40mHangup[0;37;40m("[1;35;40mSIP/30-093a52d0[0;37;40m", "[1;35;40m[0;37;40m") in new stack
   == Spawn extension (internal-admin, h, 1) exited non-zero on 'SIP/30-093a52d0'
 
[Kpbx*CLI> 
Destroying call '1229c09d-f20167fe@192.168.0.248'
 
[Kpbx*CLI> sip no debug

pbx*CLI> 
SIP Debugging Disabled

[Kpbx*CLI>