OPTIONS sip:6002@192.168.30.107:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK1917812a Max-Forwards: 70 From: "asterisk" ;tag=as6bef9b54 To: Contact: Call-ID: 15f2cd6817ad99626a04643d3319ca08@192.168.30.254:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Date: Thu, 11 Aug 2011 13:25:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-08-11 09:25:47] <--- SIP read from UDP:192.168.30.107:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK1917812a From: "asterisk" ;tag=as6bef9b54 To: "Reception 6002" ;tag=9F303857-46B48DE6 CSeq: 102 OPTIONS Call-ID: 15f2cd6817ad99626a04643d3319ca08@192.168.30.254:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces,100rel,timer,replaces,norefersub User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Content-Length: 0 <------------> [2011-08-11 09:25:54] Really destroying SIP dialog 'fae3758a-4359f653-ccef25f4@192.168.30.150' Method: SUBSCRIBE [2011-08-11 09:25:55] <--- SIP read from UDP:192.168.30.194:5060 ---> INVITE sip:6002@scurbbed.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.194:5060;branch=z9hG4bK15685acf9B74CDA8 From: "Yosh Shmenge" ;tag=C43685B5-697E523E To: CSeq: 1 INVITE Call-ID: 7ad03b09-11b22532-9fac1063@192.168.30.194 Contact: 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: 298 v=0 o=- 1313069148 1313069148 IN IP4 192.168.30.194 s=Polycom IP Phone c=IN IP4 192.168.30.194 t=0 0 a=sendrecv m=audio 2226 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 09:25:55] --- (15 headers 13 lines) --- [2011-08-11 09:25:55] == Using UDPTL TOS bits 184 [2011-08-11 09:25:55] == Using UDPTL CoS mark 5 [2011-08-11 09:25:55] Sending to 192.168.30.194:5060 (no NAT) [2011-08-11 09:25:55] Using INVITE request as basis request - 7ad03b09-11b22532-9fac1063@192.168.30.194 [2011-08-11 09:25:55] Found peer '6010' for '6010' from 192.168.30.194:5060 [2011-08-11 09:25:55] <--- Reliably Transmitting (no NAT) to 192.168.30.194:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.30.194:5060;branch=z9hG4bK15685acf9B74CDA8;received=192.168.30.194 From: "Yosh Shmenge" ;tag=C43685B5-697E523E To: ;tag=as644314a9 Call-ID: 7ad03b09-11b22532-9fac1063@192.168.30.194 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="2a337f9a" Content-Length: 0 <------------> [2011-08-11 09:25:55] Scheduling destruction of SIP dialog '7ad03b09-11b22532-9fac1063@192.168.30.194' in 6400 ms (Method: INVITE) [2011-08-11 09:25:55] <--- SIP read from UDP:192.168.30.194:5060 ---> ACK sip:6002@scurbbed.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.194:5060;branch=z9hG4bK15685acf9B74CDA8 From: "Yosh Shmenge" ;tag=C43685B5-697E523E To: ;tag=as644314a9 CSeq: 1 ACK Call-ID: 7ad03b09-11b22532-9fac1063@192.168.30.194 Contact: 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 09:25:55] --- (12 headers 0 lines) --- [2011-08-11 09:25:55] <--- SIP read from UDP:192.168.30.194:5060 ---> INVITE sip:6002@scurbbed.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.30.194:5060;branch=z9hG4bK874d12dc119DE4DD From: "Yosh Shmenge" ;tag=C43685B5-697E523E To: CSeq: 2 INVITE Call-ID: 7ad03b09-11b22532-9fac1063@192.168.30.194 Contact: 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="6010", realm="asterisk", nonce="2a337f9a", uri="sip:6002@scurbbed.com:5060;user=phone", response="679c7ef1f70b6a70e3120a85ef2c4ef3", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 298 v=0 o=- 1313069148 1313069148 IN IP4 192.168.30.194 s=Polycom IP Phone c=IN IP4 192.168.30.194 t=0 0 a=sendrecv m=audio 2226 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 09:25:55] --- (16 headers 13 lines) --- [2011-08-11 09:25:55] Sending to 192.168.30.194:5060 (no NAT) [2011-08-11 09:25:55] Using INVITE request as basis request - 7ad03b09-11b22532-9fac1063@192.168.30.194 [2011-08-11 09:25:55] Found peer '6010' for '6010' from 192.168.30.194:5060 [2011-08-11 09:25:55] == Using SIP RTP TOS bits 184 [2011-08-11 09:25:55] == Using SIP RTP CoS mark 5 [2011-08-11 09:25:55] Found RTP audio format 9 [2011-08-11 09:25:55] Found RTP audio format 0 [2011-08-11 09:25:55] Found RTP audio format 8 [2011-08-11 09:25:55] Found RTP audio format 18 [2011-08-11 09:25:55] Found RTP audio format 101 [2011-08-11 09:25:55] Found audio description format G722 for ID 9 [2011-08-11 09:25:55] Found audio description format PCMU for ID 0 [2011-08-11 09:25:55] Found audio description format PCMA for ID 8 [2011-08-11 09:25:55] Found audio description format G729 for ID 18 [2011-08-11 09:25:55] Found audio description format telephone-event for ID 101 [2011-08-11 09:25:55] 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 09:25:55] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2011-08-11 09:25:55] Peer audio RTP is at port 192.168.30.194:2226 [2011-08-11 09:25:55] Looking for 6002 in default-super (domain scurbbed.com:5060) [2011-08-11 09:25:55] WARNING[31031]: res_odbc.c:1355 _ast_odbc_request_obj2: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 3.51 Driver]MySQL server has gone away [2011-08-11 09:25:55] NOTICE[31031]: res_odbc.c:1478 odbc_obj_connect: Re-connecting scopserv [2011-08-11 09:25:55] list_route: hop: [2011-08-11 09:25:55] <--- Transmitting (no NAT) to 192.168.30.194:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.194:5060;branch=z9hG4bK874d12dc119DE4DD;received=192.168.30.194 From: "Yosh Shmenge" ;tag=C43685B5-697E523E To: Call-ID: 7ad03b09-11b22532-9fac1063@192.168.30.194 CSeq: 2 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [2011-08-11 09:25:55] -- Executing [6002@default-super:1] Set("SIP/6010-0000006f", "LOCAL_EXTEN=6002") in new stack [2011-08-11 09:25:55] -- Executing [6002@default-super:2] Set("SIP/6010-0000006f", "LOCAL_LIMIT=1") in new stack [2011-08-11 09:25:55] -- Executing [6002@default-super:3] Gosub("SIP/6010-0000006f", "all-local-extension,s,1") in new stack [2011-08-11 09:25:55] -- Executing [s@all-local-extension:1] Set("SIP/6010-0000006f", "__PICKUPMARK=6002") in new stack [2011-08-11 09:25:55] -- Executing [s@all-local-extension:2] GotoIf("SIP/6010-0000006f", "0?4") in new stack [2011-08-11 09:25:55] -- Executing [s@all-local-extension:3] Set("SIP/6010-0000006f", "GROUP(OUTGOING)=6010") in new stack [2011-08-11 09:25:55] -- Executing [s@all-local-extension:4] Set("SIP/6010-0000006f", "OUTBOUND_GROUP_ONCE=6002@INCOMING") in new stack [2011-08-11 09:25:55] -- Executing [s@all-local-extension:5] GotoIf("SIP/6010-0000006f", "0?8") in new stack [2011-08-11 09:25:55] -- Executing [s@all-local-extension:6] Set("SIP/6010-0000006f", "GROUPCOUNT=0") in new stack [2011-08-11 09:25:55] -- Executing [s@all-local-extension:7] Set("SIP/6010-0000006f", "GROUPCOUNT2=0") in new stack [2011-08-11 09:25:55] -- Executing [s@all-local-extension:8] Return("SIP/6010-0000006f", "") in new stack [2011-08-11 09:25:55] -- Executing [6002@default-super:4] Set("SIP/6010-0000006f", "SCOPSERV_DBPUT(default/wrapup/6010/lastcall)=1313069155.111") in new stack [2011-08-11 09:25:55] NOTICE[31031]: res_odbc.c:1510 odbc_obj_connect: res_odbc: Connected to scopserv [scopserv] [2011-08-11 09:25:55] set_destination: Parsing for address/port to send to [2011-08-11 09:25:55] set_destination: set destination to 192.168.30.107:5060 [2011-08-11 09:25:55] Reliably Transmitting (no NAT) to 192.168.30.107:5060: --- [2011-08-11 09:25:55] == Extension Changed 6010[default-local] new state InUse for Notify User 6002 [2011-08-11 09:25:55] set_destination: Parsing for address/port to send to [2011-08-11 09:25:55] set_destination: set destination to 192.168.30.194:5060 [2011-08-11 09:25:55] Reliably Transmitting (no NAT) to 192.168.30.194:5060: --- [2011-08-11 09:25:55] == Extension Changed 6010[default-local] new state InUse for Notify User 6010 [2011-08-11 09:25:55] set_destination: Parsing for address/port to send to [2011-08-11 09:25:55] set_destination: set destination to 192.168.30.152:5060 [2011-08-11 09:25:55] Reliably Transmitting (no NAT) to 192.168.30.152:5060: <-------------> [2011-08-11 09:25:55] --- (11 headers 0 lines) --- [2011-08-11 09:25:55] SIP Response message for INCOMING dialog NOTIFY arrived [2011-08-11 09:25:55] -- Executing [6002@default-super:5] Set("SIP/6010-0000006f", "SCOPSERV_DBPUT(default/wrapup/6002/lastcall)=1313069155.111") in new stack [2011-08-11 09:25:55] <-------------> [2011-08-11 09:25:55] --- (11 headers 0 lines) --- [2011-08-11 09:25:55] SIP Response message for INCOMING dialog NOTIFY arrived [2011-08-11 09:25:55] -- Executing [6002@default-super:6] Macro("SIP/6010-0000006f", "default-dial,SIP/6002,6002,default,20,en,b6002@default,twWxXkK,,g722,,Local/0@default-local/n,Local/s@default-aa-servicemainmenubutton1englishsubmenu") in new stack [2011-08-11 09:25:55] -- Executing [s@macro-default-dial:1] NoOp("SIP/6010-0000006f", ""CALL TO LOCAL EXTENSION FROM 6010(Yosh Shmenge)"") in new stack [2011-08-11 09:25:55] -- Executing [s@macro-default-dial:2] Set("SIP/6010-0000006f", "__PICKUPMARK=6002") in new stack [2011-08-11 09:25:55] -- Executing [s@macro-default-dial:3] AGI("SIP/6010-0000006f", "agi://127.0.0.1:4573/dial") in new stack [2011-08-11 09:25:55] <-------------> [2011-08-11 09:25:55] --- (11 headers 0 lines) --- [2011-08-11 09:25:55] SIP Response message for INCOMING dialog NOTIFY arrived [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'ARG1' result is SIP/6002 [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'ARG2' result is 6002 [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'ARG3' result is default [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'ARG4' result is 20 [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'ARG5' result is en [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'ARG6' result is b6002@default [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'ARG7' result is twWxXkK [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'ARG8' result is [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'ARG9' result is g722 [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'ARG10' result is [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'ARG11' result is Local/0@default-local/n [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'ARG12' result is Local/s@default-aa-servicemainmenubutton1englishsubmenu [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'EXT_ACCODE' result is not defined!!! [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'MACRO_PRIORITY' result is 6 [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'CALLBACK_ON_HANGUP' result is not defined!!! [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'CLID_BLOCK' result is not defined!!! [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'CALENDAR_STATUS' result is not defined!!! [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'INCOMINGLINE' result is not defined!!! [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'MONITOR_OPTION' result is wW [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'AUTO_RECORDING' result is 6010 [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'FORCE_RECORDING' result is not defined!!! [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'LIMIT_IN_DEFAULT_6002' result is 8 [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'GROUPCOUNT' result is 0 [2011-08-11 09:25:55] agi://127.0.0.1:4573/dial: Variable 'GROUPCOUNT2' result is 0 [2011-08-11 09:25:55] -- AGI Script Executing Application: (NoOp) Options: (STATUS:) [2011-08-11 09:25:55] -- AGI Script Executing Application: (Dial) Options: (SIP/6002,20,twWxXkKTwW,) [2011-08-11 09:25:55] == Using UDPTL TOS bits 184 [2011-08-11 09:25:55] == Using UDPTL CoS mark 5 [2011-08-11 09:25:55] == Using SIP RTP TOS bits 184 [2011-08-11 09:25:55] == Using SIP RTP CoS mark 5 [2011-08-11 09:25:55] Audio is at 5060 [2011-08-11 09:25:55] Adding codec 0x1000 (g722) to SDP [2011-08-11 09:25:55] Adding codec 0x4 (ulaw) to SDP [2011-08-11 09:25:55] Adding non-codec 0x1 (telephone-event) to SDP [2011-08-11 09:25:55] Reliably Transmitting (no NAT) to 192.168.30.107:5060: INVITE sip:6002@192.168.30.107:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4f8600d4 Max-Forwards: 70 From: "Yosh Shmenge" ;tag=as76fe3239 To: Contact: Call-ID: 3f79e8441a0fab137278355e27d477a3@192.168.30.254:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX (asterisk) Date: Thu, 11 Aug 2011 13:25:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "Yosh Shmenge" Content-Type: application/sdp Content-Length: 264 v=0 o=root 1332346097 1332346097 IN IP4 192.168.30.254 s=Asterisk PBX 1.8.5.0 c=IN IP4 192.168.30.254 t=0 0 m=audio 13510 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 09:25:55] -- Called SIP/6002 [2011-08-11 09:25:55] <--- SIP read from UDP:192.168.30.107:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4f8600d4 From: "Yosh Shmenge" ;tag=as76fe3239 To: "Reception 6002" ;tag=B6E79D70-EB92518B CSeq: 102 INVITE Call-ID: 3f79e8441a0fab137278355e27d477a3@192.168.30.254:5060 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Content-Length: 0 <-------------> [2011-08-11 09:25:55] --- (10 headers 0 lines) --- [2011-08-11 09:25:56] set_destination: Parsing for address/port to send to [2011-08-11 09:25:56] set_destination: set destination to 192.168.30.107:5060 [2011-08-11 09:25:56] Reliably Transmitting (no NAT) to 192.168.30.107:5060: --- [2011-08-11 09:25:56] == Extension Changed 6002[default-local] new state Ringing for Notify User 6002 [2011-08-11 09:25:56] set_destination: Parsing for address/port to send to [2011-08-11 09:25:56] set_destination: set destination to 192.168.30.194:5060 [2011-08-11 09:25:56] Reliably Transmitting (no NAT) to 192.168.30.194:5060: --- [2011-08-11 09:25:56] == Extension Changed 6002[default-local] new state Ringing for Notify User 6010 [2011-08-11 09:25:56] set_destination: Parsing for address/port to send to [2011-08-11 09:25:56] set_destination: set destination to 192.168.30.152:5060 [2011-08-11 09:25:56] Reliably Transmitting (no NAT) to 192.168.30.152:5060: <-------------> [2011-08-11 09:25:56] --- (11 headers 0 lines) --- [2011-08-11 09:25:56] SIP Response message for INCOMING dialog NOTIFY arrived [2011-08-11 09:25:56] <-------------> [2011-08-11 09:25:56] --- (11 headers 0 lines) --- <-------------> [2011-08-11 09:25:56] --- (11 headers 0 lines) --- [2011-08-11 09:25:56] SIP Response message for INCOMING dialog NOTIFY arrived [2011-08-11 09:25:56] <--- SIP read from UDP:192.168.30.107:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4f8600d4 From: "Yosh Shmenge" ;tag=as76fe3239 To: "Reception 6002" ;tag=B6E79D70-EB92518B CSeq: 102 INVITE Call-ID: 3f79e8441a0fab137278355e27d477a3@192.168.30.254:5060 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508 Allow-Events: talk,hold,conference Accept-Language: en-us,en;q=0.9 Content-Length: 0 <---------------------------------------------------------------------------------------------------------------------------------> [2011-08-11 09:25:56] --- (11 headers 0 lines) --- [2011-08-11 09:25:56] -- SIP/6002-00000070 is ringing [2011-08-11 09:25:56] <--- Transmitting (no NAT) to 192.168.30.194:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.30.194:5060;branch=z9hG4bK874d12dc119DE4DD;received=192.168.30.194 From: "Yosh Shmenge" ;tag=C43685B5-697E523E To: ;tag=as78815d33 Call-ID: 7ad03b09-11b22532-9fac1063@192.168.30.194 CSeq: 2 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: P-Asserted-Identity: "Reception 6002" Content-Length: 0 <------------> [2011-08-11 09:25:57] <--- SIP read from UDP:192.168.30.107:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4f8600d4 From: "Yosh Shmenge" ;tag=as76fe3239 To: "Reception 6002" ;tag=B6E79D70-EB92518B CSeq: 102 INVITE Call-ID: 3f79e8441a0fab137278355e27d477a3@192.168.30.254:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Content-Type: application/sdp Content-Length: 215 v=0 o=- 1313069151 1313069151 IN IP4 192.168.30.107 s=Polycom IP Phone c=IN IP4 192.168.30.107 t=0 0 a=sendrecv m=audio 2228 RTP/AVP 9 101 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 <-------------> [2011-08-11 09:25:57] --- (13 headers 10 lines) --- [2011-08-11 09:25:57] Found RTP audio format 9 [2011-08-11 09:25:57] Found RTP audio format 101 [2011-08-11 09:25:57] Found audio description format G722 for ID 9 [2011-08-11 09:25:57] Found audio description format telephone-event for ID 101 [2011-08-11 09:25:57] Capabilities: us - 0x1004 (ulaw|g722), peer - audio=0x1000 (g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1000 (g722) [2011-08-11 09:25:57] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2011-08-11 09:25:57] Peer audio RTP is at port 192.168.30.107:2228 [2011-08-11 09:25:57] list_route: hop: [2011-08-11 09:25:57] set_destination: Parsing for address/port to send to [2011-08-11 09:25:57] set_destination: set destination to 192.168.30.107:5060 [2011-08-11 09:25:57] Transmitting (no NAT) to 192.168.30.107:5060: ACK sip:6002@192.168.30.107:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK00a388fe Max-Forwards: 70 From: "Yosh Shmenge" ;tag=as76fe3239 To: ;tag=B6E79D70-EB92518B Contact: Call-ID: 3f79e8441a0fab137278355e27d477a3@192.168.30.254:5060 CSeq: 102 ACK User-Agent: Asterisk PBX (asterisk) Content-Length: 0 --- [2011-08-11 09:25:57] set_destination: Parsing for address/port to send to [2011-08-11 09:25:57] set_destination: set destination to 192.168.30.107:5060 [2011-08-11 09:25:57] Reliably Transmitting (no NAT) to 192.168.30.107:5060: ------------------------------------------------------------------------------------------------------------------------------------------- [2011-08-11 09:25:57] -- SIP/6002-00000070 answered SIP/6010-0000006f [2011-08-11 09:25:57] Audio is at 5060 [2011-08-11 09:25:57] Adding codec 0x1000 (g722) to SDP [2011-08-11 09:25:57] Adding codec 0x4 (ulaw) to SDP [2011-08-11 09:25:57] Adding non-codec 0x1 (telephone-event) to SDP [2011-08-11 09:25:57] <--- Reliably Transmitting (no NAT) to 192.168.30.194:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.194:5060;branch=z9hG4bK874d12dc119DE4DD;received=192.168.30.194 From: "Yosh Shmenge" ;tag=C43685B5-697E523E To: ;tag=as78815d33 Call-ID: 7ad03b09-11b22532-9fac1063@192.168.30.194 CSeq: 2 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: P-Asserted-Identity: "Reception 6002" Content-Type: application/sdp Content-Length: 264 v=0 o=root 1549351726 1549351726 IN IP4 192.168.30.254 s=Asterisk PBX 1.8.5.0 c=IN IP4 192.168.30.254 t=0 0 m=audio 18284 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 09:25:57] == Extension Changed 6002[default-local] new state InUse for Notify User 6002 [2011-08-11 09:25:57] set_destination: Parsing for address/port to send to [2011-08-11 09:25:57] set_destination: set destination to 192.168.30.194:5060 [2011-08-11 09:25:57] Reliably Transmitting (no NAT) to 192.168.30.194:5060: --- [2011-08-11 09:25:57] == Extension Changed 6002[default-local] new state InUse for Notify User 6010 [2011-08-11 09:25:57] set_destination: Parsing for address/port to send to [2011-08-11 09:25:57] set_destination: set destination to 192.168.30.152:5060 [2011-08-11 09:25:57] Reliably Transmitting (no NAT) to 192.168.30.152:5060: --- [2011-08-11 09:25:57] <--- SIP read from UDP:192.168.30.194:5060 ---> ACK sip:6002@192.168.30.254:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.194:5060;branch=z9hG4bKaa32389093BC5331 From: "Yosh Shmenge" ;tag=C43685B5-697E523E To: ;tag=as78815d33 CSeq: 2 ACK Call-ID: 7ad03b09-11b22532-9fac1063@192.168.30.194 Contact: 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 09:25:57] --- (12 headers 0 lines) --- [2011-08-11 09:25:57] <--- SIP read from UDP:192.168.30.107:5060 ---> <-------------> [2011-08-11 09:25:57] --- (11 headers 0 lines) --- [2011-08-11 09:25:57] SIP Response message for INCOMING dialog NOTIFY arrived [2011-08-11 09:25:57] <-------------> [2011-08-11 09:25:57] --- (11 headers 0 lines) --- [2011-08-11 09:25:57] SIP Response message for INCOMING dialog NOTIFY arrived [2011-08-11 09:25:58] <--- SIP read from UDP:192.168.30.194:5060 ---> BYE sip:6002@192.168.30.254:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.194:5060;branch=z9hG4bK66a3849a25CBA10B From: "Yosh Shmenge" ;tag=C43685B5-697E523E To: ;tag=as78815d33 CSeq: 3 BYE Call-ID: 7ad03b09-11b22532-9fac1063@192.168.30.194 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Authorization: Digest username="6010", realm="asterisk", nonce="2a337f9a", uri="sip:6002@scurbbed.com:5060;user=phone", response="1926a6662ea505c8bf163d05d0696dc5", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [2011-08-11 09:25:58] --- (12 headers 0 lines) --- [2011-08-11 09:25:58] Sending to 192.168.30.194:5060 (no NAT) [2011-08-11 09:25:58] Scheduling destruction of SIP dialog '7ad03b09-11b22532-9fac1063@192.168.30.194' in 6400 ms (Method: BYE) [2011-08-11 09:25:58] <--- Transmitting (no NAT) to 192.168.30.194:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.194:5060;branch=z9hG4bK66a3849a25CBA10B;received=192.168.30.194 From: "Yosh Shmenge" ;tag=C43685B5-697E523E To: ;tag=as78815d33 Call-ID: 7ad03b09-11b22532-9fac1063@192.168.30.194 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 15:11:50 ; [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 externip = 68.179.105.41 localnet = 192.168.0.0/255.255.0.0 localnet = 10.0.0.0/255.0.0.0 localnet = 172.16.0.0/12 localnet = 169.254.0.0/255.255.0.0 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 = g726 allow = g723 allow = gsm allow = g729 allow = slin 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" [6010] type = friend transport = udp mohinterpret = g722 mohsuggest = g722 subscribecontext = default-local accountcode = 6010 amaflags = default parkinglot = parkinglot_default defaultuser = 6010 secret = scrubbed host = dynamic language = en callgroup = 1 pickupgroup = 1 t38pt_udptl = no dtmfmode = rfc2833 qualify = 2000 trustrpid = no sendrpid = pai nat = no canreinvite = no mailbox = 6010@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=6010 setvar = MONITOR_OPTION=wW setvar = INBOUND_GROUP=6010@INCOMING setvar = SPYGROUP=default call-limit = 8 limitonpeers = yes notifyringing = yes notifyhold = yes [6002] type = friend transport = udp mohinterpret = g722 mohsuggest = g722 subscribecontext = default-local accountcode = 6002 amaflags = default parkinglot = parkinglot_default defaultuser = 6002 secret = scrubbed host = dynamic language = en callgroup = 1 pickupgroup = 1 t38pt_udptl = no dtmfmode = rfc2833 callerid = "Reception 6002" <6002> qualify = 2000 trustrpid = yes sendrpid = pai nat = no canreinvite = no mailbox = 6002@default disallow = all allow = g722 allow = ulaw context = default-notlogged cc_agent_policy = generic cc_monitor_policy = generic cc_offer_timer = 20 setvar = EXTCONTEXT=default-notlogged setvar = TRANSFER_CONTEXT=default-notlogged setvar = AUTO_RECORDING=6002 setvar = MONITOR_OPTION=wW setvar = INBOUND_GROUP=6002@INCOMING setvar = SPYGROUP=default call-limit = 8 limitonpeers = yes notifyringing = yes notifyhold = yes