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Откуда: almaty
Сообщений: 76
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Адпак-ФХО астериск проблемы
Здравствуйте! Такая вот проблема- есть пара филиалов связанная с головным офисом через VPN. Поставили в офисе Asterisk 1.4.21.2, в одном филиале Panasonic KX -TD1232RU соединен через ap1100fb по FXS с * - звонки идут без проблем. В двух других филиалах Panasonic KX -TD1232RU и Coral FlexiCom 400 соединены через ap1100fc по FXO с * - в обоих случаях связь работает то в одном направлении то в другом, т.е. если я звоню слышим друг-друга нормально, от-туда звонят я их не слышу, через некоторое время все наоборот.
Спасибо за помощь!
Вот конфиг адпака и дебаги:
Using 2666 out of 130868 bytes
!
version 8.30U
!
hostname taraz_voip
!
!
no bridge spanning-tree
!
dhcp-list 1 type server
dhcp-list 1 address server 10.1.1.2 10.1.1.126 255.255.255.128
!
!
ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address 192.168.160.241 255.255.255.0
!
interface ether1.0
no ip address
!
snmp name AP1100F
snmp enable-trap dn-register 300 forcely-block
!
no arp reset
!
route 0.0.0.0 0.0.0.0 192.168.160.1
!
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call channel early
h323 call tunnel enable
!
!
! Voice port configuration.
!
! FXO
voice-port 0/0
no caller-id enable
!
!
! FXO
voice-port 0/1
no caller-id enable
!
!
! FXO
voice-port 0/2
no caller-id enable
!
!
! FXO
voice-port 0/3
no caller-id enable
!
!
! FXO
voice-port 1/0
no caller-id enable
!
!
! FXO
voice-port 1/1
no caller-id enable
!
!
! FXO
voice-port 1/2
no caller-id enable
!
!
! FXO
voice-port 1/3
no caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
!
dial-peer voice 10 pots
destination-pattern 7262...
port 0/0
forward-digits last 3
shutdown
!
dial-peer voice 20 pots
destination-pattern 7262...
port 0/1
forward-digits last 3
shutdown
!
dial-peer voice 30 pots
destination-pattern 7262...
port 0/2
forward-digits last 3
shutdown
!
dial-peer voice 40 pots
destination-pattern 7262...
port 0/3
forward-digits last 3
shutdown
!
dial-peer voice 50 pots
destination-pattern 7262...
port 1/0
forward-digits last 3
!
dial-peer voice 60 pots
destination-pattern 7262...
port 1/1
forward-digits last 3
shutdown
!
dial-peer voice 70 pots
destination-pattern 7262...
port 1/2
forward-digits last 3
shutdown
!
dial-peer voice 80 pots
destination-pattern 7262...
port 1/3
forward-digits last 3
shutdown
!
!
!
! Voip peer configuration.
!
dial-peer voice 100 voip
destination-pattern .T
session target 192.168.1.9
session protocol sip
codec g729
no vad
dtmf-relay rtp-2833
!
!
!
!
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.160.241
no ignore-msg-from-other-gk
shutdown
!
!
! SIP UA configuration.
!
sip-ua
!
!
! MGCP configuration.
!
mgcp
no codec
vad
!
!
! Tones
!
!
!
!
--------------------
taraz_voip(config)#
Received SIP PDU from ( 192.168.1.9:5060 )
INVITE sip:7262330@192.168.160.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1b815811;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>
Contact: <sip:727151@192.168.1.9:5060>
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "151" <sip:727151@192.168.1.9>;privacy=off;screen=no
Date: Fri, 04 Dec 2009 11:11:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 20457 20457 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 6086 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Sending SIP PDU to ( 192.168.1.9:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1b815811;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 102 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0
38 <Call 7> : ****************** Call Created status(InitiatedByNet) *******************
39 <SIP 7> : Receive INVITE Request
40 <NetCon 7> : Found inbound voip peer by dest-pattern id(100)
41 <Call 7> : From Net - calledParty(7262330) callingParty(727151)
42 <Call 7> : MatchedPerfect
43 <Call 7> : MatchAllProcess After Sorted
<0> id(10) dest(7262...) prefer(0) selected(1)
<1> id(30) dest(7262...) prefer(0) selected(0)
<2> id(60) dest(7262...) prefer(0) selected(0)
<3> id(40) dest(7262...) prefer(0) selected(0)
<4> id(50) dest(7262...) prefer(0) selected(0)
<5> id(80) dest(7262...) prefer(0) selected(0)
<6> id(20) dest(7262...) prefer(0) selected(1)
<7> id(70) dest(7262...) prefer(0) selected(0)
44 <Call 7> : Initiate callee with dial-peer(7262...) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
45 <Call 7> : Fail by Shutdowned dial peer
46 <Call 7> : Initiate callee with dial-peer(7262...) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
47 <Call 7> : Fail by Shutdowned dial peer
48 <Call 7> : Initiate callee with dial-peer(7262...) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
49 <Call 7> : Fail by Shutdowned dial peer
50 <Call 7> : Initiate callee with dial-peer(7262...) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
51 <Call 7> : Fail by Shutdowned dial peer
52 <Call 7> : Initiate callee with dial-peer(7262...) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
53 <CEP 010000> : InitiateOutCall : calledNum(330), callingNum(727151), callerPort(ffffffff) type(FXO)
[113823.615] RTA(1/0/0) Rx CC_OFFHOOK_REQ [33 33 30 ] peerId(-1)
[113823.615] VM(1/0/0) FXO OffHook
[113823.615] VM(1/0/0) vopp enable
54 <CEP 010000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(7)
[113823.620] VM(1/0/0) set T38 mode STD
[113823.620] VM(1/0/0) Fax rate 9600
55 <SIP 7> : SetAlerting
56 <Call 7> : PreConnected from(10000)
[113823.620] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[113823.620] VM(1/0/0) VAD disable
[113823.620] VM(1/0/0) SID enable by CCC
57 <SIP 7> : Add Local Audio MediaFormat : 18
Sending SIP PDU to ( 192.168.1.9:5060 ) from 5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1b815811;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>;tag=3d4b010ca4
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 102 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:7262330@192.168.160.241
Content-Type: application/sdp
Content-Length: 256
v=0
o=7262330 1259957566 1259957566 IN IP4 192.168.160.241
s=AddPac Gateway SDP
c=IN IP4 192.168.160.241
t=1259957566 0
m=audio 23010 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
[113823.660] RTA(1/0/0) Rx RS_OPEN_REQ callId=7 ssId=1 G729A
peer=192.168.1.9 mp=23010/23011 hp=6086/6087
[113823.660] VM(1/0/0) vopp idle
[113823.665] VM(1/0/0) VoPP ready
[113823.665] VM(1/0/0) start codec replace timer to G729A
[113823.725] VM(1/0/0) vopp enable
[113823.725] VM(1/0/0) codec replaced to G729A
[113823.875] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113823.885] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113823.910] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113823.940] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113823.950] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113823.980] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.000] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.030] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.060] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.070] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.100] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.110] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.140] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.165] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.175] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.210] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.220] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.245] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.275] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.285] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.315] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.325] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.350] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.385] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.395] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.415] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.430] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.460] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.475] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.500] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.525] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.555] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.585] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.595] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.605] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.615] VM(1/0/0) play digit '3'
[113824.620] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.650] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.675] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.685] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.725] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.740] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.760] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.765] VM(1/0/0) play mute
[113824.795] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.805] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.815] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.825] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.855] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.865] VM(1/0/0) play digit '3'
[113824.875] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.885] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.920] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.930] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.945] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113824.980] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.015] VM(1/0/0) play mute
[113825.065] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.075] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.085] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.095] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.110] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.115] VM(1/0/0) play digit '0'
[113825.120] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.120] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.130] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.145] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.155] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.180] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.195] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.215] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.250] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.265] VM(1/0/0) play mute
[113825.265] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.295] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.305] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.325] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.350] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.360] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
[113825.365] VM(1/0/0) Fax enable
[113825.365] VM(1/0/0) play mute
[113825.365] VM(1/0/0) Tx CONNECT_CNF
58 <Call 7> : Connected from(10000)
[113825.365] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[113825.365] VM(1/0/0) VAD disable
[113825.365] VM(1/0/0) SID enable by CCC
59 <SIP 7> : SetConnected
60 <SIP 7> : Add Local Audio MediaFormat : 18
Sending SIP PDU to ( 192.168.1.9:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1b815811;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>;tag=3d4b010ca4
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 102 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:7262330@192.168.160.241
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 256
v=0
o=7262330 1259957567 1259957567 IN IP4 192.168.160.241
s=AddPac Gateway SDP
c=IN IP4 192.168.160.241
t=1259957567 0
m=audio 23010 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
[113825.410] RTA(1/0/0) Rx RS_LISTEN_REQ callId=7 ssId=1 G729A
peer=192.168.1.9 mp=23010/23011 hp=6086/6087
[113825.410] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[113825.410] VM(1/0/0) DTMF_RTP_RFC2833 enable
[113825.410] VM(1/0/0) DTMF_RTP_RFC2833 TxPT=0x65, RxPT=0x65
Received SIP PDU from ( 192.168.1.9:5060 )
ACK sip:7262330@192.168.160.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5ec90f73;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>;tag=3d4b010ca4
Contact: <sip:727151@192.168.1.9:5060>
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "151" <sip:727151@192.168.1.9>;privacy=off;screen=no
Content-Length: 0
61 <SIP 7> : ACK received
62 <SIP 7> : Receive ACK Request
63 <SIP 7> : Set Terminated Success for 102 INVITE
[113828.175] RTA(1/0/0) RTP play loss, xCnt=6
[113828.185] RTA(1/0/0) RTP play loss, xCnt=7
[113835.815] RTA(1/0/0) RTP play loss, xCnt=2
[113835.825] RTA(1/0/0) RTP play loss, xCnt=3
[113835.855] RTA(1/0/0) RTP play loss, xCnt=4
[113835.865] RTA(1/0/0) RTP play loss, xCnt=4
[113869.915] RTA(1/0/0) RTP play loss, xCnt=4
[113869.925] RTA(1/0/0) RTP play loss, xCnt=4
[113876.365] RTA(1/0/0) RTP play loss, xCnt=1
[113876.515] RTA(1/0/0) RTP play loss, xCnt=6
[113876.525] RTA(1/0/0) RTP play loss, xCnt=6
Received SIP PDU from ( 192.168.1.9:5060 )
BYE sip:7262330@192.168.160.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5689c435;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>;tag=3d4b010ca4
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "151" <sip:727151@192.168.1.9>;privacy=off;screen=no
Content-Length: 0
64 <SIP 7> : Receive BYE Request
Sending SIP PDU to ( 192.168.1.9:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5689c435;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>;tag=3d4b010ca4
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 103 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0
65 <SIP 7> : ReleaseWithNothing
[113882.925] RTA(1/0/0) Rx RS_CLOSE_REQ callId=7 ssId=1 dir=reve
[113882.925] RTA(1/0/0) Rx RS_CLOSE_REQ callId=7 ssId=1 dir=forw
[113882.925] RTA(1/0/0) close Media socket
[113882.925] RTA(1/0/0) close RTCP socket
66 <Call 7> : Terminated from(fffffffe) this(Remote:CallClear) before(NULL) forced(0)
67 <CEP 010000> : DisconnectCall at Busy
68 <CEP 010000> : StopSignal
[113882.925] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[113882.925] VM(1/0/0) play mute
69 <CEP 010000> : Disconnect (0)
[113882.930] RTA(1/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[113882.930] VM(1/0/0) vopp idle
[113882.930] VM(1/0/0) VoPP ready
[113882.930] VM(1/0/0) FXO OnHook
[113882.930] VM(1/0/0) Tx DISCONN_CNF
70 <NetEP 7> : Call TO <151> terminated reason(Remote:CallClear)
71 <CEP 010000> : Disconnected(16) at Disconnecting
72 <SIP 7> : Set Terminated Success for 103 BYE
taraz_voip(config)# [113992.510] VM(1/0/0) Rx FXO Ring Actv
[113992.510] VM(1/0/0) Tx RING_IND
73 <CEP 010000> : Call Received
[113993.590] VM(1/0/0) Rx FXO Ring Idle
[113993.590] VM(1/0/0) FXO OffHook
[113993.590] VM(1/0/0) vopp enable
[113993.590] VM(1/0/0) play Dial tone
[113993.590] VM(1/0/0) Tx OFFHOOK_IND
74 <CEP 010000> : Call Initiated : calledNumber() crv(0) total(0)
75 <Call 8> : ****************** Call Created status(InitiatedByFXO) *******************
76 <CEP 010000> : Calling number()
77 <CEP 010000> : Call id(e76d194b-3ac1-f04e-800e-0002a406e022) callNum(8)
[113998.430] VM(1/0/0) Tx DIGIT_IND '7'
[113998.430] VM(1/0/0) play mute
78 <Call 8> : Digit(7) at InitiatedByFXO
79 <Call 8> : MatchedAll
[113998.740] VM(1/0/0) Tx DIGIT_IND '2'
80 <Call 8> : Digit(2) at CalleeDeterminedWaitDigit
81 <Call 8> : MatchedAll
[113999.030] VM(1/0/0) Tx DIGIT_IND '7'
82 <Call 8> : Digit(7) at CalleeDeterminedWaitDigit
83 <Call 8> : MatchedAll
[113999.640] VM(1/0/0) Tx DIGIT_IND '1'
84 <Call 8> : Digit(1) at CalleeDeterminedWaitDigit
85 <Call 8> : MatchedAll
[113999.920] VM(1/0/0) Tx DIGIT_IND '5'
86 <Call 8> : Digit(5) at CalleeDeterminedWaitDigit
87 <Call 8> : MatchedAll
[114000.130] VM(1/0/0) Tx DIGIT_IND '1'
88 <Call 8> : Digit(1) at CalleeDeterminedWaitDigit
89 <Call 8> : MatchedAll
90 <Time 8> : Inter digit timer timeout.
91 <Call 8> : Digit(#) at CalleeDeterminedWaitDigit
92 <Call 8> : digitsReceived(727151)
93 <Call 8> : MatchAllProcess After Sorted
<0> id(100) dest(.T) prefer(0) selected(2)
94 <Call 8> : Initiate callee with dial-peer(.T) status(CalleeDeterminedAll) id(e76d194b-3ac1-f04e-800e-0002a406e022)
95 <NetEP 8> : InitiateOutCall: calledNum(727151) callingNum() target(192.168.1.9)
96 <NetEP 8> : DoCall: calledAddr(sip:727151@192.168.1.9) callingAddr()
[114003.135] VM(1/0/0) set T38 mode STD
[114003.135] VM(1/0/0) Fax rate 9600
97 <SIP 0> : No authentication information available
98 <SIP 8> : Send INVITE Request
Sending SIP PDU to ( 192.168.1.9:5060 ) from 5060
INVITE sip:727151@192.168.1.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
CSeq: 3 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Fri, 04 Dec 2009 20:15:45 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:192.168.160.241>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 236
Max-Forwards: 70
v=0
o=- 1259957745 1259957745 IN IP4 192.168.160.241
s=AddPac Gateway SDP
c=IN IP4 192.168.160.241
t=1259957745 0
m=audio 23012 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
[114003.180] RTA(1/0/0) Rx RS_LISTEN_REQ callId=8 ssId=1 G711U
peer=0.0.0.0 mp=23012/23013 hp=0/0
[114003.185] VM(1/0/0) codec same G711U
Received SIP PDU from ( 192.168.1.9:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.1.9
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9:5060>
Content-Length: 0
99 <SIP 8> : Receive 100 Trying
100 <SIP 8> : Transaction (3 INVITE) proceeding
Received SIP PDU from ( 192.168.1.9:5060 )
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.1.9
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9:5060>
Content-Length: 0
101 <SIP 8> : Receive 180 Ringing
102 <SIP 8> : Transaction (3 INVITE) proceeding
Received SIP PDU from ( 192.168.1.9:5060 )
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.1.9
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9:5060>
Content-Type: application/sdp
Content-Length: 257
v=0
o=root 20457 20457 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 0 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[114003.500] VM(1/0/0) vopp idle
[114003.500] VM(1/0/0) VoPP ready
[114003.500] VM(1/0/0) start codec replace timer to G729A
[114003.500] VM(1/0/0) Rx RTP replace codec to G729A
103 <SIP 8> : Receive 183 Session Progress
104 <SIP 8> : Transaction (3 INVITE) proceeding
[114003.560] VM(1/0/0) vopp enable
[114003.560] VM(1/0/0) codec replaced to G729A
Received SIP PDU from ( 192.168.1.9:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.1.9
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9:5060>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
105 <SIP 8> : Receive 200 OK
106 <SIP 8> : Set Terminated Success for 3 INVITE
Received SIP PDU from ( 192.168.1.9:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.1.9
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9:5060>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Received SIP PDU from ( 192.168.1.9:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.1.9
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9:5060>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Received SIP PDU from ( 192.168.1.9:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.1.9
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9:5060>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Received SIP PDU from ( 192.168.1.9:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.1.9
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9:5060>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Received SIP PDU from ( 192.168.1.9:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.1.9
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9:5060>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
taraz_voip(config)#
Received SIP PDU from ( 192.168.1.9:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.1.9
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9:5060>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
taraz_voip(config)# 107 <Time 0> : SIP INVITE Expire timer timeout.
108 <SIP 8> : ReleaseWithCANCEL for 0 INVITEs)
[114183.185] RTA(1/0/0) Rx RS_CLOSE_REQ callId=8 ssId=1 dir=reve
[114183.185] RTA(1/0/0) close Media socket
[114183.185] RTA(1/0/0) close RTCP socket
109 <Call 8> : Terminated from(fffffffe) this(Remote:NoAnswerFromUser) before(NULL) forced(0)
110 <CEP 010000> : DisconnectCall at Busy
111 <CEP 010000> : StopSignal
[114183.185] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[114183.185] VM(1/0/0) play mute
112 <CEP 010000> : Disconnect (0)
[114183.185] RTA(1/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[114183.185] VM(1/0/0) vopp idle
[114183.190] VM(1/0/0) VoPP ready
[114183.190] VM(1/0/0) FXO OnHook
[114183.190] VM(1/0/0) Tx DISCONN_CNF
113 <NetEP 8> : Call TO <sip:727151@192.168.1.9> terminated reason(Remote:NoAnswerFromUser)
114 <CEP 010000> : Disconnected(16) at Disconnecting
taraz_voip(config)#
-----------------------------
SIP Debugging Enabled for IP: 192.168.160.241:5060
-- Executing [7262330@default:1] Dial("SIP/voip-08edce18", "SIP/taraz/7262330") in new stack
Audio is at 192.168.1.9 port 6086
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.160.241:5060:
INVITE sip:7262330@192.168.160.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1b815811;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>
Contact: <sip:727151@192.168.1.9>
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "151" <sip:727151@192.168.1.9>;privacy=off;screen=no
Date: Fri, 04 Dec 2009 11:11:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 20457 20457 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 6086 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called taraz/7262330
Asterisk*CLI>
<--- SIP read from 192.168.160.241:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1b815811;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 102 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 192.168.160.241:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1b815811;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>;tag=3d4b010ca4
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 102 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:7262330@192.168.160.241
Content-Type: application/sdp
Content-Length: 256
v=0
o=7262330 1259957566 1259957566 IN IP4 192.168.160.241
s=AddPac Gateway SDP
c=IN IP4 192.168.160.241
t=1259957566 0
m=audio 23010 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (11 headers 11 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.160.241:23010
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.160.241:23010
-- SIP/taraz-08eef750 is making progress passing it to SIP/voip-08edce18
<--- SIP read from 192.168.160.241:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1b815811;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>;tag=3d4b010ca4
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 102 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:7262330@192.168.160.241
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 256
v=0
o=7262330 1259957567 1259957567 IN IP4 192.168.160.241
s=AddPac Gateway SDP
c=IN IP4 192.168.160.241
t=1259957567 0
m=audio 23010 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.160.241:23010
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.160.241:23010
list_route: hop: <sip:7262330@192.168.160.241>
set_destination: Parsing <sip:7262330@192.168.160.241> for address/port to send to
set_destination: set destination to 192.168.160.241, port 5060
Transmitting (no NAT) to 192.168.160.241:5060:
ACK sip:7262330@192.168.160.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5ec90f73;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>;tag=3d4b010ca4
Contact: <sip:727151@192.168.1.9>
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "151" <sip:727151@192.168.1.9>;privacy=off;screen=no
Content-Length: 0
---
-- SIP/taraz-08eef750 answered SIP/voip-08edce18
-- Packet2Packet bridging SIP/voip-08edce18 and SIP/taraz-08eef750
Scheduling destruction of SIP dialog '1170e4100a9016a24ad822437c2096aa@192.168.1.9' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:7262330@192.168.160.241> for address/port to send to
set_destination: set destination to 192.168.160.241, port 5060
Reliably Transmitting (no NAT) to 192.168.160.241:5060:
BYE sip:7262330@192.168.160.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5689c435;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>;tag=3d4b010ca4
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "151" <sip:727151@192.168.1.9>;privacy=off;screen=no
Content-Length: 0
---
== Spawn extension (default, 7262330, 1) exited non-zero on 'SIP/voip-08edce18'
Asterisk*CLI>
<--- SIP read from 192.168.160.241:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK5689c435;rport
From: "151" <sip:727151@192.168.1.9>;tag=as6563257c
To: <sip:7262330@192.168.160.241>;tag=3d4b010ca4
Call-ID: 1170e4100a9016a24ad822437c2096aa@192.168.1.9
CSeq: 103 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1170e4100a9016a24ad822437c2096aa@192.168.1.9' Method: INVITE
== Spawn extension (default, 727240, 1) exited non-zero on 'SIP/192.168.170.241-08ef4f08'
== Spawn extension (default, 7112101, 1) exited non-zero on 'SIP/voip-08ece4d8'
Asterisk*CLI>
<--- SIP read from 192.168.160.241:5060 --->
INVITE sip:727151@192.168.1.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
CSeq: 3 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Fri, 04 Dec 2009 20:15:45 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:192.168.160.241>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 236
Max-Forwards: 70
v=0
o=- 1259957745 1259957745 IN IP4 192.168.160.241
s=AddPac Gateway SDP
c=IN IP4 192.168.160.241
t=1259957745 0
m=audio 23012 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
--- (16 headers 10 lines) ---
Sending to 192.168.160.241 : 5060 (no NAT)
Using INVITE request as basis request - f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
Found peer 'taraz'
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.160.241:23012
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.160.241:23012
Looking for 727151 in default (domain 192.168.1.9)
list_route: hop: <sip:192.168.160.241>
<--- Transmitting (no NAT) to 192.168.160.241:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9>
Content-Length: 0
<------------>
-- Executing [727151@default:1] Dial("SIP/192.168.160.241-08edca10", "SIP/alcatel/151") in new stack
-- Called alcatel/151
-- SIP/alcatel-08eecbf8 is ringing
Asterisk*CLI>
<--- Transmitting (no NAT) to 192.168.160.241:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9>
Content-Length: 0
<------------>
-- SIP/alcatel-08eecbf8 is making progress passing it to SIP/192.168.160.241-08edca10
Audio is at 192.168.1.9 port 22644
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 192.168.160.241:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20457 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- SIP/alcatel-08eecbf8 answered SIP/192.168.160.241-08edca10
Audio is at 192.168.1.9 port 22644
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.160.241:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- Packet2Packet bridging SIP/192.168.160.241-08edca10 and SIP/alcatel-08eecbf8
Retransmitting #1 (no NAT) to 192.168.160.241:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #2 (no NAT) to 192.168.160.241:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #3 (no NAT) to 192.168.160.241:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #4 (no NAT) to 192.168.160.241:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
== Spawn extension (default, 727151, 1) exited non-zero on 'SIP/192.168.160.241-08edca10'
Scheduling destruction of SIP dialog 'f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241' in 32000 ms (Method: INVITE)
Retransmitting #5 (no NAT) to 192.168.160.241:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #6 (no NAT) to 192.168.160.241:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.160.241:5060;branch=z9hG4bKf14b6d0fa43;received=192.168.160.241
From: <sip:192.168.160.241>;tag=f14b6d0fa4
To: <sip:727151@192.168.1.9>;tag=as5180a785
Call-ID: f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:727151@192.168.1.9>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 20457 20458 IN IP4 192.168.1.9
s=session
c=IN IP4 192.168.1.9
t=0 0
m=audio 22644 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 4 17:15:10] WARNING[20482]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission f16d194b-1874-6d5d-800f-0002a406e022@192.168.160.241 for seqno 3 (Critical Response)
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