<--- SIP read from UDP:172.30.0.197:5060 --->
INVITE sip:4116@scrubbed.com:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK223cfcb417A1EB67
From: "Yosh Shmenge" <sip:4115@scrubbed.com>;tag=B153786E-B0099A61
To: <sip:4116@scrubbed.com;user=phone>
CSeq: 1 INVITE
Call-ID: 7d7e327a-61d91ded-bcab79c0@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 294

v=0
o=- 1313069393 1313069393 IN IP4 172.30.0.197
s=Polycom IP Phone
c=IN IP4 172.30.0.197
t=0 0
a=sendrecv
m=audio 2254 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000

<------------->
[2011-08-11 06:29:57] --- (15 headers 13 lines) ---
[2011-08-11 06:29:57]   == Using UDPTL TOS bits 184
[2011-08-11 06:29:57]   == Using UDPTL CoS mark 5
[2011-08-11 06:29:57] Sending to 172.30.0.197:5060 (no NAT)
[2011-08-11 06:29:57] Using INVITE request as basis request - 7d7e327a-61d91ded-bcab79c0@172.30.0.197
[2011-08-11 06:29:57] Found peer '4115' for '4115' from 172.30.0.197:5060
[2011-08-11 06:29:57]
<--- Reliably Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK223cfcb417A1EB67;received=172.30.0.197
From: "Yosh Shmenge" <sip:4115@scrubbed.com>;tag=B153786E-B0099A61
To: <sip:4116@scrubbed.com;user=phone>;tag=as605f852a
Call-ID: 7d7e327a-61d91ded-bcab79c0@172.30.0.197
CSeq: 1 INVITE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3bb77946"
Content-Length: 0


<------------>
[2011-08-11 06:29:57] Scheduling destruction of SIP dialog '7d7e327a-61d91ded-bcab79c0@172.30.0.197' in 6400 ms (Method: INVITE)
[2011-08-11 06:29:57]
<--- SIP read from UDP:172.30.0.197:5060 --->
ACK sip:4116@scrubbed.com:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK223cfcb417A1EB67
From: "Yosh Shmenge" <sip:4115@scrubbed.com>;tag=B153786E-B0099A61
To: <sip:4116@scrubbed.com;user=phone>;tag=as605f852a
CSeq: 1 ACK
Call-ID: 7d7e327a-61d91ded-bcab79c0@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Content-Length: 0


<------------->
[2011-08-11 06:29:57] --- (12 headers 0 lines) ---
[2011-08-11 06:29:57]
<--- SIP read from UDP:172.30.0.197:5060 --->
INVITE sip:4116@scrubbed.com:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK7f0f91f3A8713286
From: "Yosh Shmenge" <sip:4115@scrubbed.com>;tag=B153786E-B0099A61
To: <sip:4116@scrubbed.com;user=phone>
CSeq: 2 INVITE
Call-ID: 7d7e327a-61d91ded-bcab79c0@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Authorization: Digest username="4115", realm="asterisk", nonce="3bb77946", uri="sip:4116@scrubbed.com:5060;user=phone", response="53308dbb3f8021b41542ce837032a9a3", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 294

v=0
o=- 1313069393 1313069393 IN IP4 172.30.0.197
s=Polycom IP Phone
c=IN IP4 172.30.0.197
t=0 0
a=sendrecv
m=audio 2254 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000

<------------->
[2011-08-11 06:29:57] --- (16 headers 13 lines) ---
[2011-08-11 06:29:57] Sending to 172.30.0.197:5060 (no NAT)
[2011-08-11 06:29:57] Using INVITE request as basis request - 7d7e327a-61d91ded-bcab79c0@172.30.0.197
[2011-08-11 06:29:57] Found peer '4115' for '4115' from 172.30.0.197:5060
[2011-08-11 06:29:57]   == Using SIP RTP TOS bits 184
[2011-08-11 06:29:57]   == Using SIP RTP CoS mark 5
[2011-08-11 06:29:57] Found RTP audio format 9
[2011-08-11 06:29:57] Found RTP audio format 0
[2011-08-11 06:29:57] Found RTP audio format 8
[2011-08-11 06:29:57] Found RTP audio format 18
[2011-08-11 06:29:57] Found RTP audio format 101
[2011-08-11 06:29:57] Found audio description format G722 for ID 9
[2011-08-11 06:29:57] Found audio description format PCMU for ID 0
[2011-08-11 06:29:57] Found audio description format PCMA for ID 8
[2011-08-11 06:29:57] Found audio description format G729 for ID 18
[2011-08-11 06:29:57] Found audio description format telephone-event for ID 101
[2011-08-11 06:29:57] Capabilities: us - 0x1004 (ulaw|g722), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1004 (ulaw|g722)
[2011-08-11 06:29:57] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2011-08-11 06:29:57] Peer audio RTP is at port 172.30.0.197:2254
[2011-08-11 06:29:57] Looking for 4116 in default-super (domain scrubbed.com:5060)
[2011-08-11 06:29:57] list_route: hop: <sip:4115@172.30.0.197:5060>
[2011-08-11 06:29:57]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK7f0f91f3A8713286;received=172.30.0.197
From: "Yosh Shmenge" <sip:4115@scrubbed.com>;tag=B153786E-B0099A61
To: <sip:4116@scrubbed.com;user=phone>
Call-ID: 7d7e327a-61d91ded-bcab79c0@172.30.0.197
CSeq: 2 INVITE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:4116@172.30.0.10:5060>
Content-Length: 0


<------------>
[2011-08-11 06:29:57]     -- Executing [4116@default-super:1] Set("SIP/4115-00000002", "LOCAL_EXTEN=4116") in new stack
[2011-08-11 06:29:57]     -- Executing [4116@default-super:2] Gosub("SIP/4115-00000002", "all-local-extension,s,1") in new stack
[2011-08-11 06:29:57]     -- Executing [s@all-local-extension:1] Set("SIP/4115-00000002", "__PICKUPMARK=4116") in new stack
[2011-08-11 06:29:57]     -- Executing [s@all-local-extension:2] GotoIf("SIP/4115-00000002", "0?4") in new stack
[2011-08-11 06:29:57]     -- Executing [s@all-local-extension:3] Set("SIP/4115-00000002", "GROUP(OUTGOING)=4115") in new stack
[2011-08-11 06:29:57]     -- Executing [s@all-local-extension:4] Set("SIP/4115-00000002", "OUTBOUND_GROUP_ONCE=4116@INCOMING") in new stack
[2011-08-11 06:29:57]     -- Executing [s@all-local-extension:5] GotoIf("SIP/4115-00000002", "1?8") in new stack
[2011-08-11 06:29:57]     -- Goto (all-local-extension,s,8)
[2011-08-11 06:29:57]     -- Executing [s@all-local-extension:8] Return("SIP/4115-00000002", "") in new stack
[2011-08-11 06:29:57]     -- Executing [4116@default-super:3] Set("SIP/4115-00000002", "SCOPSERV_DBPUT(default/wrapup/4115/lastcall)=1313069397.2") in new stack
[2011-08-11 06:29:57]     -- Executing [4116@default-super:4] Set("SIP/4115-00000002", "SCOPSERV_DBPUT(default/wrapup/4116/lastcall)=1313069397.2") in new stack
[2011-08-11 06:29:57]     -- Executing [4116@default-super:5] Macro("SIP/4115-00000002", "default-dial,SIP/4116,4116,default,25,en,u4116@default,twWxXkK,,default,,Local/0@default-local/n,disa") in new stack
[2011-08-11 06:29:57]     -- Executing [s@macro-default-dial:1] NoOp("SIP/4115-00000002", ""CALL TO LOCAL EXTENSION FROM 4115(Yosh Shmenge)"") in new stack
[2011-08-11 06:29:57]     -- Executing [s@macro-default-dial:2] Set("SIP/4115-00000002", "__PICKUPMARK=4116") in new stack
[2011-08-11 06:29:57]     -- Executing [s@macro-default-dial:3] AGI("SIP/4115-00000002", "agi://127.0.0.1:4573/dial") in new stack
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'ARG1' result is SIP/4116
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'ARG2' result is 4116
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'ARG3' result is default
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'ARG4' result is 25
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'ARG5' result is en
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'ARG6' result is u4116@default
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'ARG7' result is twWxXkK
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'ARG8' result is
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'ARG9' result is default
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'ARG10' result is
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'ARG11' result is Local/0@default-local/n
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'ARG12' result is disa
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'EXT_ACCODE' result is not defined!!!
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'MACRO_PRIORITY' result is 5
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'CALLBACK_ON_HANGUP' result is not defined!!!
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'CLID_BLOCK' result is not defined!!!
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'CALENDAR_STATUS' result is not defined!!!
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'INCOMINGLINE' result is not defined!!!
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'MONITOR_OPTION' result is wW
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'AUTO_RECORDING' result is 4115
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'FORCE_RECORDING' result is not defined!!!
[2011-08-11 06:29:57]  agi://127.0.0.1:4573/dial: Variable 'LIMIT_IN_DEFAULT_4116' result is not defined!!!
[2011-08-11 06:29:57]     -- AGI Script Executing Application: (NoOp) Options: (STATUS:)
[2011-08-11 06:29:57]     -- AGI Script Executing Application: (Dial) Options: (SIP/4116,25,twWxXkKTwW,)
[2011-08-11 06:29:57]   == Using UDPTL TOS bits 184
[2011-08-11 06:29:57]   == Using UDPTL CoS mark 5
[2011-08-11 06:29:57]   == Using SIP RTP TOS bits 184
[2011-08-11 06:29:57]   == Using SIP RTP CoS mark 5
[2011-08-11 06:29:57] Audio is at 5060
[2011-08-11 06:29:57] Adding codec 0x1000 (g722) to SDP
[2011-08-11 06:29:57] Adding codec 0x4 (ulaw) to SDP
[2011-08-11 06:29:57] Adding non-codec 0x1 (telephone-event) to SDP
[2011-08-11 06:29:57] Reliably Transmitting (no NAT) to 172.30.0.199:5060:
INVITE sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2cb5fbe4
Max-Forwards: 70
From: "Yosh Shmenge" <sip:4115@172.30.0.10>;tag=as74d76455
To: <sip:4116@172.30.0.199:5060>
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 4e586e6f7396c0524687e6017e39e134@172.30.0.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX (asterisk)
Date: Thu, 11 Aug 2011 13:29:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
P-Asserted-Identity: "Yosh Shmenge" <sip:4115@172.30.0.10>
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 545391833 545391833 IN IP4 172.30.0.10
s=Asterisk PBX 1.8.5.0
c=IN IP4 172.30.0.10
t=0 0
m=audio 15794 RTP/AVP 9 0 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[2011-08-11 06:29:57]     -- Called SIP/4116
[2011-08-11 06:29:57]
<--- SIP read from UDP:172.30.0.199:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2cb5fbe4
From: "Yosh Shmenge" <sip:4115@172.30.0.10>;tag=as74d76455
To: "Stan Shmenge" <sip:4116@172.30.0.199:5060>;tag=128B4BD3-B847B356
CSeq: 102 INVITE
Call-ID: 4e586e6f7396c0524687e6017e39e134@172.30.0.10:5060
Contact: <sip:4116@172.30.0.199:5060>
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-08-11 06:29:57] --- (10 headers 0 lines) ---
[2011-08-11 06:29:57]
<--- SIP read from UDP:172.30.0.199:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2cb5fbe4
From: "Yosh Shmenge" <sip:4115@172.30.0.10>;tag=as74d76455
To: "Stan Shmenge" <sip:4116@172.30.0.199:5060>;tag=128B4BD3-B847B356
CSeq: 102 INVITE
Call-ID: 4e586e6f7396c0524687e6017e39e134@172.30.0.10:5060
Contact: <sip:4116@172.30.0.199:5060>
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Allow-Events: talk,hold,conference
Accept-Language: en-us,en;q=0.9
Content-Length: 0


<------------->
[2011-08-11 06:29:57] --- (11 headers 0 lines) ---
[2011-08-11 06:29:57]     -- SIP/4116-00000003 is ringing
[2011-08-11 06:29:57]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK7f0f91f3A8713286;received=172.30.0.197
From: "Yosh Shmenge" <sip:4115@scrubbed.com>;tag=B153786E-B0099A61
To: <sip:4116@scrubbed.com;user=phone>;tag=as7729a7d3
Call-ID: 7d7e327a-61d91ded-bcab79c0@172.30.0.197
CSeq: 2 INVITE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:4116@172.30.0.10:5060>
Content-Length: 0


<------------>
[2011-08-11 06:29:59]
<--- SIP read from UDP:172.30.0.199:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK2cb5fbe4
From: "Yosh Shmenge" <sip:4115@172.30.0.10>;tag=as74d76455
To: "Stan Shmenge" <sip:4116@172.30.0.199:5060>;tag=128B4BD3-B847B356
CSeq: 102 INVITE
Call-ID: 4e586e6f7396c0524687e6017e39e134@172.30.0.10:5060
Contact: <sip:4116@172.30.0.199:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Content-Type: application/sdp
Content-Length: 211

v=0
o=- 1313069395 1313069395 IN IP4 172.30.0.199
s=Polycom IP Phone
c=IN IP4 172.30.0.199
t=0 0
a=sendrecv
m=audio 2236 RTP/AVP 9 101
a=sendrecv
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000

<------------->
[2011-08-11 06:29:59] --- (13 headers 10 lines) ---
[2011-08-11 06:29:59] Found RTP audio format 9
[2011-08-11 06:29:59] Found RTP audio format 101
[2011-08-11 06:29:59] Found audio description format G722 for ID 9
[2011-08-11 06:29:59] Found audio description format telephone-event for ID 101
[2011-08-11 06:29:59] Capabilities: us - 0x1004 (ulaw|g722), peer - audio=0x1000 (g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1000 (g722)
[2011-08-11 06:29:59] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2011-08-11 06:29:59] Peer audio RTP is at port 172.30.0.199:2236
[2011-08-11 06:29:59] list_route: hop: <sip:4116@172.30.0.199:5060>
[2011-08-11 06:29:59] set_destination: Parsing <sip:4116@172.30.0.199:5060> for address/port to send to
[2011-08-11 06:29:59] set_destination: set destination to 172.30.0.199:5060
[2011-08-11 06:29:59] Transmitting (no NAT) to 172.30.0.199:5060:
ACK sip:4116@172.30.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK3c776028
Max-Forwards: 70
From: "Yosh Shmenge" <sip:4115@172.30.0.10>;tag=as74d76455
To: <sip:4116@172.30.0.199:5060>;tag=128B4BD3-B847B356
Contact: <sip:4115@172.30.0.10:5060>
Call-ID: 4e586e6f7396c0524687e6017e39e134@172.30.0.10:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX (asterisk)
Content-Length: 0


---
[2011-08-11 06:29:59]     -- SIP/4116-00000003 answered SIP/4115-00000002
[2011-08-11 06:29:59] Audio is at 5060
[2011-08-11 06:29:59] Adding codec 0x1000 (g722) to SDP
[2011-08-11 06:29:59] Adding codec 0x4 (ulaw) to SDP
[2011-08-11 06:29:59] Adding non-codec 0x1 (telephone-event) to SDP
[2011-08-11 06:29:59]
<--- Reliably Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK7f0f91f3A8713286;received=172.30.0.197
From: "Yosh Shmenge" <sip:4115@scrubbed.com>;tag=B153786E-B0099A61
To: <sip:4116@scrubbed.com;user=phone>;tag=as7729a7d3
Call-ID: 7d7e327a-61d91ded-bcab79c0@172.30.0.197
CSeq: 2 INVITE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:4116@172.30.0.10:5060>
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 1194124156 1194124156 IN IP4 172.30.0.10
s=Asterisk PBX 1.8.5.0
c=IN IP4 172.30.0.10
t=0 0
m=audio 17172 RTP/AVP 9 0 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[2011-08-11 06:29:59]
<--- SIP read from UDP:172.30.0.197:5060 --->
ACK sip:4116@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bKc473be7fD7EF7892
From: "Yosh Shmenge" <sip:4115@scrubbed.com>;tag=B153786E-B0099A61
To: <sip:4116@scrubbed.com;user=phone>;tag=as7729a7d3
CSeq: 2 ACK
Call-ID: 7d7e327a-61d91ded-bcab79c0@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Max-Forwards: 70
Content-Length: 0


<------------->
[2011-08-11 06:29:59] --- (12 headers 0 lines) ---
[2011-08-11 06:30:00]
<--- SIP read from UDP:172.30.0.197:5060 --->
BYE sip:4116@172.30.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK2fda3705AE78C5D8
From: "Yosh Shmenge" <sip:4115@scrubbed.com>;tag=B153786E-B0099A61
To: <sip:4116@scrubbed.com;user=phone>;tag=as7729a7d3
CSeq: 3 BYE
Call-ID: 7d7e327a-61d91ded-bcab79c0@172.30.0.197
Contact: <sip:4115@172.30.0.197:5060>
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en-us,en;q=0.9
Authorization: Digest username="4115", realm="asterisk", nonce="3bb77946", uri="sip:4116@scrubbed.com:5060;user=phone", response="ccd1be1c0ad9d40a5b10053f27066083", algorithm=MD5
Max-Forwards: 70
Content-Length: 0


<------------->
[2011-08-11 06:30:00] --- (12 headers 0 lines) ---
[2011-08-11 06:30:00] Sending to 172.30.0.197:5060 (no NAT)
[2011-08-11 06:30:00] Scheduling destruction of SIP dialog '7d7e327a-61d91ded-bcab79c0@172.30.0.197' in 6400 ms (Method: BYE)
[2011-08-11 06:30:00]
<--- Transmitting (no NAT) to 172.30.0.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.197:5060;branch=z9hG4bK2fda3705AE78C5D8;received=172.30.0.197
From: "Yosh Shmenge" <sip:4115@scrubbed.com>;tag=B153786E-B0099A61
To: <sip:4116@scrubbed.com;user=phone>;tag=as7729a7d3
Call-ID: 7d7e327a-61d91ded-bcab79c0@172.30.0.197
CSeq: 3 BYE
Server: Asterisk PBX (asterisk)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

cat /etc/asterisk/sip.conf
;
; DESCRIPTION: Configuration for Asterisk
; GENERATOR:   ScopServ-VoIP v2.0
; TIMESTAMP:   2011-08-10 13:43:57
;


[general]
context         =  default-incoming-guest
callevents      =  yes
alwaysauthreject =  yes
tcpenable       =  yes
tcpbindaddr     =  0.0.0.0:5060
tlsenable       =  no
jbenable        =  yes
jbforce         =  no
jbmaxsize       =  200
jbresyncthreshold =  1000
jbimpl          =  adaptive
jblog           =  no
t38pt_udptl     =  yes
progressinband  =  never
bindport        =  5060
bindaddr        =  0.0.0.0
rtpkeepalive    =  0
limitonpeers    =  yes
notifyringing   =  yes
notifyhold      =  yes
realm           =  asterisk
useragent       =  Asterisk PBX (asterisk)
maxexpirey      =  3600
defaultexpirey  =  120
recordhistory   =  no
autocreatepeers =  no
srvlookup       =  yes
videosupport    =  yes
directrtpsetup  =  no
disallow        =  all
allow           =  g722
allow           =  ulaw
allow           =  alaw
allow           =  g723
allow           =  g726
allow           =  g729
allow           =  slin
allow           =  gsm
allow           =  ilbc
allow           =  lpc10
allow           =  speex
allow           =  adpcm
allow           =  h261
allow           =  h263
allow           =  h263p
allow           =  h264
tos_sip         =  CS0
tos_audio       =  ef
tos_video       =  CS0
pedantic        =  no
allowexternaldomains =  no
allowexternalinvites =  no
autodomain      =  no
relaxdtmf       =  no
promiscredir    =  no
usereqphone     =  yes
compactheaders  =  no
trustrpid       =  no
sendrpid        =  pai
#include "sip-register.conf"
#include "default/sip.conf"
#include "default/sip-extras.conf"
#include "sip-extras.conf"


[4115]
type            =  friend
transport       =  udp
mohinterpret    =  default
mohsuggest      =  default
subscribecontext =  default-local
accountcode     =  4115
parkinglot      =  parkinglot_default
defaultuser     =  4115
secret          =  scrubbed
host            =  dynamic
language        =  en
t38pt_udptl     =  no
dtmfmode        =  rfc2833
qualify         =  2000
trustrpid       =  no
sendrpid        =  pai
nat             =  no
canreinvite     =  no
mailbox         =  4115@default
disallow        =  all
allow           =  g722
allow           =  ulaw
context         =  default-super
cc_agent_policy =  generic
cc_monitor_policy =  generic
cc_offer_timer  =  20
setvar          =  EXTCONTEXT=default-super
setvar          =  TRANSFER_CONTEXT=default-super
setvar          =  AUTO_RECORDING=4115
setvar          =  MONITOR_OPTION=wW
setvar          =  INBOUND_GROUP=4115@INCOMING
call-limit      =  8
limitonpeers    =  yes
notifyringing   =  yes
notifyhold      =  yes


[4116]
type            =  friend
transport       =  udp
mohinterpret    =  default
mohsuggest      =  default
subscribecontext =  default-local
accountcode     =  4116
parkinglot      =  parkinglot_default
defaultuser     =  4116
secret          =  scrubbed
host            =  dynamic
language        =  en
t38pt_udptl     =  no
dtmfmode        =  rfc2833
qualify         =  2000
trustrpid       =  no
sendrpid        =  pai
nat             =  no
canreinvite     =  no
mailbox         =  4116@default
disallow        =  all
allow           =  g722
allow           =  ulaw
context         =  default-super
cc_agent_policy =  generic
cc_monitor_policy =  generic
cc_offer_timer  =  20
setvar          =  EXTCONTEXT=default-super
setvar          =  TRANSFER_CONTEXT=default-super
setvar          =  AUTO_RECORDING=4116
setvar          =  MONITOR_OPTION=wW
setvar          =  INBOUND_GROUP=4116@INCOMING
call-limit      =  8
limitonpeers    =  yes
notifyringing   =  yes
notifyhold      =  yes