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Нет перевода звонка(tranfer) в связке AP2620-AP2640-APIP100
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Автор:  АдминАдм [ 02 апр 2012, 13:39 ]
Заголовок сообщения:  Нет перевода звонка(tranfer) в связке AP2620-AP2640-APIP100
В одной подсети находятся 3 устройства, которые прекрасно работаю сами по себе и отлично звонят друг другу. Конфигурацию устройств прикладываю (знаком «….» отметил строки, которые удалил из-за «лишней информативности»):
КОНФИГ AP2640:
Код:
version 8.30W
!
hostname AP2620
!
!
no bridge spanning-tree
!
!
ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
!
interface ether0.0
 ip address 192.168.14.4 255.255.255.0
!
interface ether1.0
 no ip address
!
!
snmp name AP2620
snmp enable-trap dn-register 300 forcely-block
!
no arp reset
!
service ez-setup
service tftpd
!
auto-script autorun.inf
!
!
!
!
!
! VoIP configuration.
!
!
! Controller configuration.
!
controller e1 0/0
 clock-source  slave
 channel-group timeslots 1-31  0
 out-barred-group timeslots 16
 chan-number-order ascending
!
!
!
! Voice service voip configuration.
!
voice service voip
 fax protocol bypass
 fax rate 9600
 h323 call start fast
 h323 call tunnel enable
 busyout monitor gatekeeper
 busyout monitor sip-server
 no busyout monitor callagent
 busyout monitor voip-interface
!
!
! Voice port configuration.
!
! E1
voice-port 0/0 0
!
!
!
!
! Pots peer configuration.
!
....
!
dial-peer voice 90 pots
 destination-pattern 9[0123]...
 port 0/0  0
 forward-digits from  1
 huntstop
!
dial-peer voice 901 pots
 destination-pattern 90T
 port 0/0  0
 forward-digits from  1
 huntstop
!
dial-peer voice 902 pots
 destination-pattern 0T
 port 0/0  0
 forward-digits from  1
 huntstop
!
!
!
! Voip peer configuration.
!
....
!
dial-peer voice 831 voip
 destination-pattern 831..
 session target 192.168.14.53
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay h245-alphanumeric
 translate-outgoing called-number 2083
 preference 1
 huntstop
 fax protocol bypass
! fax protocol t38 redundancy 0
 fax rate 9600
!
....
!
dial-peer voice 3105 voip
 destination-pattern 83105
 session target 192.168.14.55
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing called-number 2083
 preference 1
 huntstop
!
....
!
dial-peer voice 3135 voip
 destination-pattern 83135
 session target 192.168.14.55
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing called-number 2083
 huntstop
!
....
!
dial-peer voice 9172 voip
 destination-pattern 9172
 session target 192.168.14.53
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing called-number 2083
 huntstop
!
dial-peer voice 9173 voip
 destination-pattern 9173
 session target 192.168.14.53
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing called-number 2083
 huntstop
!
dial-peer voice 9174 voip
 destination-pattern 9174
 session target 192.168.14.53
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing called-number 2083
 huntstop
!
!
!
dial-peer ipaddr-prefix n
dial-peer call-pickup *4
dial-peer call-hold h
dial-peer call-transfer h
!
dial-peer hunt 2
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
 h323-id voip.192.168.14.4
 no ignore-msg-from-other-gk
!
!
! Translation Rule configuration.
!
translation-rule 2083
 rule 0      83...                    ...
 rule 1      84...                    ...
 rule 2      849..                    ..
!
!
!
! SIP UA configuration.
!
sip-ua
 sip-username ap2620
 sip-password router
 sip-server 192.168.14.50
 call-transfer-mode attended
 remote-party-id
 register e164
!
!
! MGCP configuration.
!
mgcp
 codec g711ulaw
 vad
!
!
! Tones
!
!
!
voip-interface ether0.0
!


КОНФИГ AP2640:
Код:
!
version 8.30U
!
hostname AP2640
!
!
no bridge spanning-tree
!
!
ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
!
interface ether0.0
 ip address 192.168.14.53 255.255.255.0
!
interface ether1.0
 no ip address
!
snmp name AP2640
snmp enable-trap dn-register 300 forcely-block
!
no arp reset
!
service ez-setup
!
auto-script autorun.inf
!
!
!
!
!
! VoIP configuration.
!
!
! Controller configuration.
!
!
!
! Voice service voip configuration.
!
!
voice service voip
 fax protocol bypass
 fax rate 9600
 h323 call start fast
 h323 call tunnel enable
 timeout tohdt 900
 timeout tdhf 500
 busyout monitor gatekeeper
 busyout monitor sip-server
 no busyout monitor callagent
 busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 0/1
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 0/2
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 0/3
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 1/0
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 1/1
 caller-id enable
 caller-id type etsi
!
!
!
! FXS
voice-port 1/2
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 1/3
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 2/0
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 2/1
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 2/2
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 2/3
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 3/0
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 3/1
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 3/2
 caller-id enable
 caller-id type etsi
!
!
!
! FXS
voice-port 3/3
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 4/0
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 4/1
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 4/2
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 4/3
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 5/0
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 5/1
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 5/2
 caller-id enable
 caller-id type etsi
!
!
! FXS
voice-port 5/3
 caller-id enable
 caller-id type etsi
!
!
! FXO
voice-port 6/0
 connection plar 102
 description 2239617
 ring detect-timeout 80
 caller-id enable
 caller-id type etsi
!
!
! FXO
voice-port 6/1
 connection plar 110
 ring detect-timeout 80
 caller-id enable
 caller-id type etsi
!
!
! FXO
voice-port 6/2
 connection plar 103
 ring detect-timeout 80
 caller-id enable
 caller-id type etsi
!
!
! FXO
voice-port 6/3
 connection plar 107
 ring detect-timeout 80
 caller-id enable
 caller-id type etsi
!
!
! FXO
voice-port 7/0
 connection plar 108
 ring detect-timeout 80
 caller-id enable
 caller-id type etsi
!
!
! FXO
voice-port 7/1
 connection plar 111
 ring detect-timeout 80
 caller-id enable
 caller-id type etsi
!
!
! FXO
voice-port 7/2
 caller-id enable
 caller-id type etsi
!
!
! FXO
voice-port 7/3
 caller-id enable
 caller-id type etsi
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 100 pots
 destination-pattern 101
 port 0/0
 display-name Galina_Proskuryakova
!
dial-peer voice 101 pots
 destination-pattern 102
 port 0/1
 display-name Polina_Gomzina
!
dial-peer voice 102 pots
 destination-pattern 103
 port 0/2
 display-name Elina_Shakirova
!
dial-peer voice 103 pots
 destination-pattern 125
 port 0/3
 display-name Iluza_Emasheva
!
dial-peer voice 110 pots
 destination-pattern 126
 port 1/0
 display-name Alsu_Kutlubaeva
!
dial-peer voice 111 pots
 destination-pattern 106
 port 1/1
 display-name Olesya_Kuznetsova
!
dial-peer voice 112 pots
 destination-pattern 107
 port 1/2
 display-name Svetlana_Livshits
!
dial-peer voice 113 pots
 destination-pattern 108
 port 1/3
 display-name Masha_Viktorova
!
dial-peer voice 120 pots
 destination-pattern 123
 port 2/0
 display-name Roman_Nurgaliev
!
dial-peer voice 121 pots
 destination-pattern 110
 port 2/1
 display-name Olga_Larionova
!
dial-peer voice 122 pots
 destination-pattern 111
 port 2/2
 display-name Tatyana_Chernitcina
!
dial-peer voice 123 pots
 destination-pattern 112
 port 2/3
 display-name Lidiya_Akchurina
!
dial-peer voice 130 pots
 destination-pattern 113
 port 3/0
 display-name Elena_Yamakaeva
!
dial-peer voice 131 pots
 destination-pattern 109
 port 3/1
 display-name Guzel_Damirovna_Meschaninova
!
dial-peer voice 132 pots
 destination-pattern 115
 port 3/2
 display-name Ekaterina_Safina
!
dial-peer voice 133 pots
 destination-pattern 116
 port 3/3
 display-name Olga_Savitckaya
!
dial-peer voice 140 pots
 destination-pattern 117
 port 4/0
 to-display-name non-quoted
!
dial-peer voice 141 pots
 destination-pattern 118
 port 4/1
 display-name Lidiya_Akchurina
!
dial-peer voice 142 pots
 destination-pattern 119
 port 4/2
 display-name Kseniya_Leontyeva
!
!
dial-peer voice 143 pots
 destination-pattern 120
 port 4/3
 display-name A_Zhirnova
!
dial-peer voice 150 pots
 destination-pattern 121
 port 5/0
 display-name S_Mitkina
!
dial-peer voice 151 pots
 destination-pattern 122
 port 5/1
 display-name Grigory_Popov
!
dial-peer voice 152 pots
 destination-pattern 130
 port 5/2
 display-name 130_unused
!
dial-peer voice 153 pots
 destination-pattern 124
 port 5/3
 display-name Darya_Romanova
!
dial-peer voice 224 pots
 port 6/0
!
dial-peer voice 225 pots
 port 6/1
!
dial-peer voice 226 pots
 port 6/2
!
dial-peer voice 227 pots
 port 6/3
!
dial-peer voice 228 pots
 destination-pattern 994T
 port 7/0
!
dial-peer voice 229 pots
 port 7/1
!
dial-peer voice 230 pots
 port 7/2
!
dial-peer voice 231 pots
 port 7/3
!
dial-peer voice 300 pots
 destination-pattern 9172
 port 0/0
 preference 1
!
dial-peer voice 301 pots
 destination-pattern 9172
 port 0/1
 preference 2
!
dial-peer voice 302 pots
 destination-pattern 9174
 port 0/2
 preference 2
!
dial-peer voice 311 pots
 destination-pattern 9174
 port 1/1
 preference 4
!
dial-peer voice 312 pots
 destination-pattern 9174
 port 1/2
 preference 1
!
dial-peer voice 313 pots
 destination-pattern 9172
 port 1/3
 preference 4
!
dial-peer voice 321 pots
 destination-pattern 9173
 port 2/1
 preference 1
!
dial-peer voice 323 pots
 destination-pattern 9174
 port 2/3
 preference 3
!
dial-peer voice 332 pots
 destination-pattern 9173
 port 3/2
 preference 5
!
dial-peer voice 333 pots
 destination-pattern 9172
 port 3/3
 preference 3
!
dial-peer voice 343 pots
 destination-pattern 9173
 port 4/3
 preference 4
!
dial-peer voice 350 pots
 destination-pattern 9172
 port 5/0
 preference 5
!
dial-peer voice 351 pots
 destination-pattern 9173
 port 5/1
 preference 2
!
dial-peer voice 352 pots
 destination-pattern 9173
 port 5/2
 preference 3
!
!
!
! Voip peer configuration.
!
....
!
dial-peer voice 90 voip
 destination-pattern 9[0123]...
 session target 192.168.14.4
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing calling-number 2083
 huntstop
 fax protocol bypass
! fax protocol t38 redundancy 0
 fax rate 9600
!
....
!
dial-peer voice 906 voip
 destination-pattern 906..
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing calling-number 2083
 huntstop
 shutdown
!
dial-peer voice 1000 voip
 destination-pattern 0T
 session target 192.168.14.4
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing called-number 2641
 translate-outgoing calling-number 2000
 huntstop
 fax protocol bypass
! fax protocol t38 redundancy 0
 fax rate 9600
!
dial-peer voice 1100 voip
 destination-pattern 1..
 session target 192.168.14.53
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing calling-number 2083
 preference 1
 huntstop
!
....
!
dial-peer voice 1105 voip
 destination-pattern 105
 session target 192.168.14.55
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing calling-number 2083
 huntstop
!
....
!
dial-peer voice 1135 voip
 destination-pattern 135
 session target 192.168.14.55
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing calling-number 2083
 huntstop
!
....
!
dial-peer voice 9010 voip
 destination-pattern 90101
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing calling-number 2083
 huntstop
 shutdown
!
!
!
dial-peer ipaddr-prefix n
dial-peer call-pickup *4
dial-peer call-hold h
dial-peer call-transfer h
dial-peer pstn-switch n
dial-peer switch-to-pstn-on-call none
dial-peer switch-to-voip-on-call none
!
dial-peer hunt 2
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
 h323-id voip.192.168.14.53
 no ignore-msg-from-other-gk
!
!
! Translation Rule configuration.
!
translation-rule 2083
 rule 0      T                        83T
!
translation-rule 9835
 rule 0      83T                      T
!
translation-rule 2084
 rule 0      T                        0505%98
!
translation-rule 2641
 rule 0      0T                       641T
!
translation-rule 2000
 rule 1      101                      9172%98
 rule 2      102                      9172%98
 rule 3      104                      9172%98
 rule 4      108                      9172%98
 rule 5      116                      9172%98
 rule 6      105                      9173%98
 rule 7      210                      9173%98
 rule 8      215                      9173%98
 rule 9      217                      9173%98
 rule 10     218                      9173%98
 rule 11     219                      9173%98
 rule 12     103                      9174%98
 rule 13     106                      9174%98
 rule 14     107                      9174%98
 rule 15     109                      9174%98
 rule 16     111                      9174%98
 rule 17     112                      9174%98
 rule 18     113                      9174%98
 rule 19     114                      9174%98
 rule 20     124                      9174%98
 rule 21     125                      9174%98
 rule 22     126                      9174%98
 rule 23     127                      9174%98
 rule 24     128                      9174%98
 rule 25     129                      9174%98
!
!
!
! SIP UA configuration.
!
sip-ua
 sip-username ap2640
 sip-password router
 sip-server 192.168.14.50
 call-transfer-mode attended
!
!
! MGCP configuration.
!
mgcp
 no codec
 vad
!
!
! Tones
!


КОНФИГ IP100:
Код:
!
version 8.41.081
!
hostname IP100-105
!
username root password router administrator
!
!
interface Loopback0
 ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
 ip address 192.168.14.55 255.255.255.0
 ip nat outside
 speed auto
 no qos-control
!
interface FastEthernet0/1
 ip address 192.168.10.1 255.255.255.0
 ip nat inside
 speed auto
 no qos-control
!
access-list 100 permit ip 192.168.10.0 0.0.0.255 any
!
!
ip nat inside source list 100 interface FastEthernet0/0  overload
!
!
!
ftp server
http server
!
logging event 0-emergency
!
! IP PHONE OSD configuration.
!
osd
 language english
 network signaling sip
 network sscp disable
 phone lcd-type graphic
 phone ring-type 1
 phone volume ring 4
 phone volume input 6
 phone volume output 5
 phone volume micbooster disable
 phone auto-hook-on disable
 phone display-name 105-A_A_Lyadkov
 phone voice-codec 0
 phone dnd-mode silence
 phone pbx-mode general
 phone auto-answer disable
 phone save-mode always
 phone forward-status disable
 phone conference-status disable
 phone password 2337
 phone password-status disable
 phone admin-lock factory status disable
 phone admin-lock internet status disable
 phone admin-lock voip status disable
 phone admin-lock service status disable
 phone admin-lock auto-upgrade status disable
 phone admin-lock sscp status disable
 phone privacy-password 0000
 phone privacy-status disable
 phone privacy-lock menu status disable
 phone privacy-lock incoming status disable
 phone privacy-lock outgoing status disable
!
! SSCP configuration.!
!
!
! SSCP Static CM List
sscp
!
! SSCP Dynamic CM List
sscp
!
!
sscp
 call-manager broadcast port 8855
 logger disable
 logger level info
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
 fax protocol t38 redundancy 0
 fax rate disable
 h323 call start fast
 h323 call tunnel enable
 timeout tmohdt 300
 call-barring unconfigured-ip-address
!
!
! Voice port configuration.
!
! SPEECH
voice-port 0/0
 caller-id type etsi
!
!
! FXS
voice-port 0/1
 caller-id enable
 caller-id type etsi
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 104 pots
 destination-pattern 105
 port 0/0
 display-name A_A_Lyadkov
!
dial-peer voice 105 pots
 destination-pattern 105
 port 0/1
 display-name A_A_Lyadkov
!
dial-peer voice 134 pots
 destination-pattern 135
 port 0/0
 display-name A_A_Lyadkov
!
dial-peer voice 135 pots
 destination-pattern 135
 port 0/1
 display-name A_A_Lyadkov
!
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
 destination-pattern 0T
 session target ip 192.168.14.4
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing called-number 2641
 translate-outgoing calling-number 2000
 huntstop
!
dial-peer voice 1001 voip
 session target ras
 voice-class codec 0
 no vad
 dtmf-relay h245-alphanumeric
!
....
!
dial-peer voice 1090 voip
 destination-pattern 9[0123]...
 session target ip 192.168.14.4
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing calling-number 2083
 huntstop
!
dial-peer voice 1100 voip
 destination-pattern 1..
 session target ip 192.168.14.53
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 preference 1
 huntstop
!
....
!
dial-peer voice 1105 voip
 destination-pattern 105
 session target ip 192.168.14.55
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing calling-number 2083
 huntstop
!
....
!
dial-peer voice 1135 voip
 destination-pattern 135
 session target ip 192.168.14.55
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing calling-number 2083
 huntstop
!
....
!
dial-peer voice 1901 voip
 destination-pattern 90101
 session target ras
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing calling-number 2083
 huntstop
 shutdown
!
dial-peer voice 1906 voip
 destination-pattern 906..
 session target ras
 session protocol sip
 codec g711alaw
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
 translate-outgoing calling-number 2083
 huntstop
 shutdown
!
!
!
dial-peer ipaddr-prefix n
dial-peer call-pickup *4
dial-peer call-hold h
dial-peer call-transfer h
!
dial-peer hunt 2
!
!
! Gateway configuration.
!
gateway
 h323-id voip.192.168.14.55
 no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g726r32
 codec preference 4 g726r16
 codec preference 5 g729
 codec preference 6 g7231r63
!
!
!
! Translation Rule configuration.
!
translation-rule 2083
 rule 0      T                        83T
!
translation-rule 9835
 rule 0      83T                      T
!
translation-rule 2084
 rule 0      T                        0505%98
!
translation-rule 2641
 rule 0      0T                       641T
!
translation-rule 2000
 rule 0      105                      9173%98
 rule 1      135                      9173%98
!
!
!
! SIP UA configuration.
!
sip-ua
 fault-tolerance 10 500
 sip-username ap2640
 sip-password router
 sip-server 192.168.14.50
 rport enable
 call-transfer-mode attended
 remote-party-id
!
!
! Tones
!
!
!
!
line console
!
line vty
!
!
sms
 quota 30
!
Автор:  АдминАдм [ 02 апр 2012, 13:43 ]
Заголовок сообщения:  Нет перевода звонка(tranfer) в связке AP2620-AP2640-APIP100
У нас не работает перевод звонков в следующем алгоритме.
Есть станция Nortel(префикс станции 90) c абонентом 339 соединенная по Е1 с AP2620. Данный абонент совершает звонок через АР2620 на AP2640(префикс 83), где его абонент 102 переводит вызов на AP IP100. В тот момент, когда абонент AP2620(Nortel) должен соединиться с AP IP100 и слышить его голос, в трубке с обоих концов «тишина». Если в совершать звонок с PSTN на FXO-порт AP2640, то дальнейший перевод на AP IP100 прекрасно работает.
Дэбаг с АР IP100:
Код:
        Received SIP PDU from ( 192.168.14.50:5060 )
OPTIONS sip:192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.50:5060;branch=z9hG4bK7a6af8c8;rport
From: "Unknown" <sip:Unknown@192.168.14.50>;tag=as0ccd2e44
To: <sip:192.168.14.55>
Contact: <sip:Unknown@192.168.14.50>
Call-ID: 607328b90b3422b215ba7ad00c4cd3b5@192.168.14.50
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 Mar 2012 09:57:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0



        Sending SIP PDU to ( 192.168.14.50:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.50:5060;branch=z9hG4bK7a6af8c8;rport=5060
From: "Unknown" <sip:Unknown@192.168.14.50>;tag=as0ccd2e44
To: <sip:192.168.14.55>
Call-ID: 607328b90b3422b215ba7ad00c4cd3b5@192.168.14.50
CSeq: 102 OPTIONS
User-Agent: AddPac SIP Gateway
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Length: 0



        Received SIP PDU from ( 192.168.14.50:5060 )
OPTIONS sip:192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.50:5060;branch=z9hG4bK07ff2ae1;rport
From: "Unknown" <sip:Unknown@192.168.14.50>;tag=as543b9593
To: <sip:192.168.14.55>
Contact: <sip:Unknown@192.168.14.50>
Call-ID: 17840b577041bba44e71c27165bcffc8@192.168.14.50
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 Mar 2012 09:58:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0



        Sending SIP PDU to ( 192.168.14.50:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.50:5060;branch=z9hG4bK07ff2ae1;rport=5060
From: "Unknown" <sip:Unknown@192.168.14.50>;tag=as543b9593
To: <sip:192.168.14.55>
Call-ID: 17840b577041bba44e71c27165bcffc8@192.168.14.50
CSeq: 102 OPTIONS
User-Agent: AddPac SIP Gateway
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Length: 0



        Received SIP PDU from ( 192.168.14.53:5060 )
INVITE sip:105@192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49754
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9754 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Mon, 26 Mar 2012 15:56:19 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:83102@192.168.14.53>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 234
Max-Forwards: 70

v=0
o=83102 1332777379 1332777379 IN IP4 192.168.14.53
s=AddPac Gateway SDP
c=IN IP4 192.168.14.53
t=1332777379 0
m=audio 23640 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20


        Sending SIP PDU to ( 192.168.14.53:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49754
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9754 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


1       <Call   4>      : ******  Call Created status(InitiatedByNet) ver(8.28:2
006-02-06-00-00) time(1734) ****
2       <SIP    4>      : Receive INVITE Request
3       <NetCon 4>      : Found inbound voip peer by IP address id(1100)
4       <Call   4>      : From Net - calledParty(105) callingParty(83102)
5       <Call   4>      : MatchedPerfect
6       <Call   4>      : MatchAllProcess After Sorted
                          <0>  id(105) dest(105) prefer(0) selected(1)
                          <1>  id(104) dest(105) prefer(0) selected(2)
7       <Call   4>      : Initiate callee with dial-peer(105) status(CalleeDeter
minedAll) id(00000000-0000-0000-0000-000000000000)
8       <CEP    000100> : InitiateOutCall :  calledNum(), callingNum(83102), cal
lerPort(ffffffff) type(FXS)
[1742.525] RTA(0/1/0) Rx CC_RING_REQ [80 21 01 08 30 31 30 31 30 30 32 38 02 05
38 33 31 30 32 07 0e 50 6f 6c 69 6e 61 5f 47 6f 6d 7a 69 6e 61 ] peerId(-1)
[1742.525] VM(0/1/0) DaTime [L=8] 30 31 30 31 30 30 32 38
[1742.525] VM(0/1/0) CgNumb [L=5] 38 33 31 30 32
[1742.525] VM(0/1/0) CgName [L=14] 50 6f 6c 69 6e 61 5f 47 6f 6d 7a 69 6e 61
[1742.525] VM(0/1/0) Line Reverse
[1742.525] VM(0/1/0) Start ring actv
[1742.525] VM(0/1/0) SW to -72V
[1742.525] VM(0/1/0) FXS input block
9       <CEP    000100> : Outbound call to CEP callId(00000000-0000-0000-0000-00
0000000000) callNum(4)
[1742.525] VM(0/1/0) set T38 disable
[1742.525] VM(0/1/0) set T38 mode STD
[1742.525] VM(0/1/0) Fax rate disab
10      <SIP    4>      : SetAlerting

        Sending SIP PDU to ( 192.168.14.53:5060 ) from 5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49754
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>;tag=c600b703a4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9754 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:105@192.168.14.55
Content-Length: 0


[1743.525] VM(0/1/0) Gen ring idle
[1743.525] VM(0/1/0) slic normal mode
[1744.025] VM(0/1/0) Tx CID enable
[1744.025] VP(0/1/0) use line
[1744.025] VP(0/0/0) add line
[1744.025] VP(0/1/0) CallerId enable, std/gain 1/6
[1744.025] VP(0/1/0) open channel
[1744.025] VM(0/1/0) play mute
[1744.025] VP(0/1/0) Tx IBS signal 2/0
[1744.025] VP(0/1/0) Tx IBS dir 0
[1744.085] VM(0/1/0) Tx CID DATA [L=35] 80 21 01 08 30 31 30 31 30 30 32 38 02 0
5 38 33 31 30 32 07 0e 50 6f 6c 69 6e 61 5f 47 6f 6d 7a 69 6e 61
[1744.085] VP(0/1/0) play CallerId
[1744.095] VP(0/1/0) GeneralEvent IBS gen end
[1745.085] VM(0/1/0) Tx CID fin
[1745.085] VP(0/1/0) stop CallerId
[1745.085] VM(0/1/0) vopp idle
[1745.085] VP(0/1/0) close channel
[1745.525] VM(0/1/0) Gen ring actv
[1745.525] VM(0/1/0) slic ring mode
[1745.815] VM(0/1/0) vmOffHook
[1745.875] VM(0/1/0) vmTmoOffHook
[1745.875] VM(0/1/0) SW to -48V
[1745.875] VM(0/1/0) FXS input pass
[1745.875] VM(0/1/0) Line Forward
[1745.935] VM(0/1/0) vmTmoOffHook
[1745.935] VM(0/1/0) Rx OffHook
[1745.935] VP(0/1/0) use line
[1745.935] VP(0/0/0) add line
[1745.935] VP(0/1/0) open channel
[1745.935] VM(0/1/0) T38 Fax disabled
[1745.935] VM(0/1/0) Tx CONNECT_CNF
11      <Call   4>      : Connected from(100)
[1745.935] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
   VAD_CTRL 0
[1745.935] VM(0/1/0) VAD disable
[1745.935] VP(0/1/0) update VAD 0
[1745.935] VM(0/1/0) SID enable by CCC
12      <SIP    4>      : SetConnected
13      <SIP    4>      : Add Local Audio MediaFormat : 8

        Sending SIP PDU to ( 192.168.14.53:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49754
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>;tag=c600b703a4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9754 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:105@192.168.14.55
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 215

v=0
o=105 1738 1738 IN IP4 192.168.14.55
s=AddPac Gateway SDP
c=IN IP4 192.168.14.55
t=0 0
m=audio 23014 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

[1745.945] RTA(0/1/0) Rx RS_LISTEN_REQ callId=4 ssId=1 G711A
        peer=192.168.14.53 mp=23014/23015 hp=23640/23641
[1745.945] VM(0/1/0) vopp idle
[1745.945] VP(0/1/0) close channel
[1745.945] VM(0/1/0) start codec replace timer to G711A
[1745.945] RTA(0/1/0) Rx RS_OPEN_REQ callId=4 ssId=1 G711A
        peer=192.168.14.53 mp=23014/23015 hp=23640/23641
[1745.945] VM(0/1/0) under codec replace to G711A
[1745.945] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
   DTMF_CTRL 1
[1745.945] VM(0/1/0) DTMF enable
[1745.945] VM(0/1/0) DTMF_RTP_RFC2833 enable
[1745.945] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
   Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65

        Received SIP PDU from ( 192.168.14.53:5060 )
ACK sip:105@192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49754
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>;tag=c600b703a4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9754 ACK
Content-Length: 0
Max-Forwards: 70


14      <SIP    4>      : ACK received
15      <SIP    4>      : Receive ACK Request
16      <SIP    4>      : Set Terminated Success for 9754 INVITE
[1746.005] VP(0/1/0) open channel
[1746.005] VM(0/1/0) codec replaced to G711A
[1746.005] VM(0/1/0) play mute
[1746.005] VP(0/1/0) Tx IBS signal 2/0
[1746.005] VP(0/1/0) Tx IBS dir 0
[1746.075] VP(0/1/0) GeneralEvent IBS gen end

        Received SIP PDU from ( 192.168.14.53:5060 )
INVITE sip:105@192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49755
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>;tag=c600b703a4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9755 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Mon, 26 Mar 2012 15:56:29 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:83102@192.168.14.53>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 244
Max-Forwards: 70

v=0
o=83102 1332777379 1332777379 IN IP4 192.168.14.53
s=AddPac Gateway SDP
c=IN IP4 0.0.0.0
t=1332777379 0
a=sendonly
m=audio 23640 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=ptime:20

17      <SIP    4>      : Receive INVITE Request
18      <SIP    4>      : Add Local Audio MediaFormat : 8
[1751.895] RTA(0/1/0) Rx RS_CLOSE_REQ callId=4 ssId=1 dir=all
[1751.895] RTA(0/1/0) close Media socket
[1751.895] RTA(0/1/0) close RTCP socket
[1751.895] RTA(0/1/0) Rx RS_LISTEN_REQ callId=4 ssId=1 G711A
        peer=192.168.14.53 mp=23014/23015 hp=23640/23641
[1751.895] VM(0/1/0) codec same G711A
19      <Call   4>      : Hold Request from Network.
[1751.895] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
   TONE_PLAY Status1
[1751.895] VM(0/1/0) play Status1 tone
[1751.895] VP(0/1/0) Tx IBS signal 6/7
[1751.895] VP(0/1/0) Tx IBS dir 0

        Sending SIP PDU to ( 192.168.14.53:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49755
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>;tag=c600b703a4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9755 INVITE
User-Agent: AddPac SIP Gateway
Contact: sip:105@192.168.14.55
Content-Type: application/sdp
Content-Length: 227

v=0
o=105 1744 1744 IN IP4 192.168.14.55
s=AddPac Gateway SDP
c=IN IP4 192.168.14.55
t=0 0
a=recvonly
m=audio 23014 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15


        Received SIP PDU from ( 192.168.14.53:5060 )
ACK sip:105@192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49755
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>;tag=c600b703a4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9755 ACK
Content-Length: 0
Max-Forwards: 70


20      <SIP    4>      : ACK received
21      <SIP    4>      : Receive ACK Request
22      <SIP    4>      : Set Terminated Success for 9755 INVITE

        Received SIP PDU from ( 192.168.14.4:5060 )
INVITE sip:105@192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.4:5060;branch=z9hG4bKce4fa1f3a4514
From: <sip:90339@192.168.14.4>;tag=ce4fa1f3a4
To: <sip:105@192.168.14.55>
Call-ID: cebc704f-f21b-a184-8bf3-0002a40864fc@192.168.14.4
CSeq: 514 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Mon, 26 Mar 2012 19:00:30 GMT
User-Agent: AddPac AP2620 8.30W
Contact: <sip:90339@192.168.14.4>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 232
Replaces: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53;to-tag=c600b703a4;f
rom-tag=a34f143fa4
Max-Forwards: 70
Remote-Party-ID: <sip:90339@192.168.14.4>;screen=yes;party=calling

v=0
o=90339 1332788430 1332788430 IN IP4 192.168.14.4
s=AddPac Gateway SDP
c=IN IP4 192.168.14.4
t=1332788430 0
m=audio 23098 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20


        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.14.4:5060;branch=z9hG4bKce4fa1f3a4514
From: <sip:90339@192.168.14.4>;tag=ce4fa1f3a4
To: <sip:105@192.168.14.55>
Call-ID: cebc704f-f21b-a184-8bf3-0002a40864fc@192.168.14.4
CSeq: 514 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


23      <Call   5>      : ******  Call Created status(InitiatedByNet) ver(8.28:2
006-02-06-00-00) time(1744) ****
24      <SIP    4>      : Find Matching Connection with Replace HEADER
25      <SIP    5>      : Receive INVITE Request
26      <NetCon 5>      : Found inbound voip peer by dest-pattern id(1090)
27      <Call   5>      : Replace Request received From Net - Called(105) Callin
g(90339)
28      <CEP    000100> : StopSignal
[1751.965] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
   DTMF_STOP
[1751.965] VM(0/1/0) stop WT or WT1 tone
[1751.965] VM(0/1/0) play mute
[1751.965] VP(0/1/0) Tx IBS signal 2/0
[1751.965] VP(0/1/0) Tx IBS dir 0
[1751.965] VM(0/1/0) set T38 disable
[1751.965] VM(0/1/0) set T38 mode STD
[1751.965] VM(0/1/0) Fax rate disab
[1751.965] RTA(0/1/0) Rx RS_CLOSE_REQ callId=4 ssId=1 dir=reve
[1751.965] RTA(0/1/0) close Media socket
[1751.965] RTA(0/1/0) close RTCP socket
29      <Call   4>      : Terminated from(100) this(Local:CallClear) before(NULL
) forced(0) time(1744)
30      <SIP    4>      : ReleaseWithBYE
31      <SIP    4>      : Send BYE Request

        Sending SIP PDU to ( 192.168.14.53:5060 ) from 5060
BYE sip:83102@192.168.14.53 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.55:5060;branch=z9hG4bKc600b703a42
From: <sip:105@192.168.14.55>;tag=c600b703a4
To: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 2 BYE
Date: Thu, 01 Jan 1970 00:29:04 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:105@192.168.14.55>
Content-Length: 0
Max-Forwards: 70


32      <NetEP  4>      : Call FROM <Polina_Gomzina> terminated reason(Local:Cal
lClear)
33      <SIP    5>      : SetConnected
34      <SIP    5>      : Add Local Audio MediaFormat : 8

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.4:5060;branch=z9hG4bKce4fa1f3a4514
From: <sip:90339@192.168.14.4>;tag=ce4fa1f3a4
To: <sip:105@192.168.14.55>;tag=d0008504a4
Call-ID: cebc704f-f21b-a184-8bf3-0002a40864fc@192.168.14.4
CSeq: 514 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:105@192.168.14.55
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 215

v=0
o=105 1744 1744 IN IP4 192.168.14.55
s=AddPac Gateway SDP
c=IN IP4 192.168.14.55
t=0 0
m=audio 23018 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

[1751.980] RTA(0/1/0) Rx RS_LISTEN_REQ callId=5 ssId=1 G711A
        peer=192.168.14.4 mp=23018/23019 hp=23098/23099
[1751.980] VM(0/1/0) codec same G711A
[1751.980] RTA(0/1/0) Rx RS_OPEN_REQ callId=5 ssId=1 G711A
        peer=192.168.14.4 mp=23018/23019 hp=23098/23099
[1751.980] VM(0/1/0) codec same G711A
[1751.980] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
   DTMF_CTRL 1
[1751.985] VM(0/1/0) DTMF enable
[1751.985] VM(0/1/0) DTMF_RTP_RFC2833 enable
[1751.985] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
   Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65

        Received SIP PDU from ( 192.168.14.53:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.55:5060;branch=z9hG4bKc600b703a42
From: <sip:105@192.168.14.55>;tag=c600b703a4
To: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 2 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0


35      <SIP    4>      : Receive 200 OK
36      <SIP    4>      : Transaction (2 BYE) completed

        Received SIP PDU from ( 192.168.14.4:5060 )
ACK sip:105@192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.4:5060;branch=z9hG4bKce4fa1f3a4514
From: <sip:90339@192.168.14.4>;tag=ce4fa1f3a4
To: <sip:105@192.168.14.55>;tag=d0008504a4
Call-ID: cebc704f-f21b-a184-8bf3-0002a40864fc@192.168.14.4
CSeq: 514 ACK
Content-Length: 0
Max-Forwards: 70


37      <SIP    5>      : ACK received
38      <SIP    5>      : Receive ACK Request
39      <SIP    5>      : Set Terminated Success for 514 INVITE
[1752.040] VP(0/1/0) GeneralEvent IBS gen end
40      <SIP    4>      : Set Terminated Success for 2 BYE
[1759.305] VM(0/1/0) vmOnHook
[1759.355] VM(0/1/0) vmTmoOnHook
[1759.405] VM(0/1/0) vmTmoOnHook
[1759.455] VM(0/1/0) vmTmoOnHook
[1759.505] VM(0/1/0) vmTmoOnHook
[1759.555] VM(0/1/0) vmTmoOnHook
[1759.605] VM(0/1/0) vmTmoOnHook
[1759.655] VM(0/1/0) vmTmoOnHook
[1759.705] VM(0/1/0) vmTmoOnHook
[1759.755] VM(0/1/0) vmTmoOnHook
[1759.805] VM(0/1/0) vmTmoOnHook
[1759.855] VM(0/1/0) vmTmoOnHook
[1759.905] VM(0/1/0) vmTmoOnHook
[1759.955] VM(0/1/0) vmTmoOnHook
[1760.005] VM(0/1/0) vmTmoOnHook
[1760.005] VM(0/1/0) Rx OnHook
[1760.005] VM(0/1/0) vopp idle
[1760.005] VP(0/1/0) close channel
[1760.005] VM(0/1/0) Tx DISCONN_CNF
41      <CEP    000100> : Disconnected(16) at Busy
42      <Call   5>      : Terminated from(100) this(Local:CallClear) before(NULL
) forced(0) time(1752)
43      <SIP    5>      : ReleaseWithBYE
44      <SIP    5>      : Send BYE Request

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
BYE sip:90339@192.168.14.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.55:5060;branch=z9hG4bKd0008504a43
From: <sip:105@192.168.14.55>;tag=d0008504a4
To: <sip:90339@192.168.14.4>;tag=ce4fa1f3a4
Call-ID: cebc704f-f21b-a184-8bf3-0002a40864fc@192.168.14.4
CSeq: 3 BYE
Date: Thu, 01 Jan 1970 00:29:12 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:105@192.168.14.55>
Content-Length: 0
Max-Forwards: 70


[1760.010] RTA(0/1/0) Rx RS_CLOSE_REQ callId=5 ssId=1 dir=all
[1760.010] RTA(0/1/0) close Media socket
[1760.010] RTA(0/1/0) close RTCP socket
45      <NetEP  5>      : Call FROM <90339> terminated reason(Local:CallClear)
46      <CEP    000100> : DisconnectCall at Idle
47      <SIP    5>      : Transaction Client  (3 BYE) Timeout (retry #1)
48      <SIP    5>      : Send BYE Request

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
BYE sip:90339@192.168.14.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.55:5060;branch=z9hG4bKd0008504a43
From: <sip:105@192.168.14.55>;tag=d0008504a4
To: <sip:90339@192.168.14.4>;tag=ce4fa1f3a4
Call-ID: cebc704f-f21b-a184-8bf3-0002a40864fc@192.168.14.4
CSeq: 3 BYE
Date: Thu, 01 Jan 1970 00:29:12 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:105@192.168.14.55>
Content-Length: 0
Max-Forwards: 70


49      <SIP    5>      : Transaction Client  (3 BYE) Timeout (retry #2)
50      <SIP    5>      : Send BYE Request

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
BYE sip:90339@192.168.14.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.55:5060;branch=z9hG4bKd0008504a43
From: <sip:105@192.168.14.55>;tag=d0008504a4
To: <sip:90339@192.168.14.4>;tag=ce4fa1f3a4
Call-ID: cebc704f-f21b-a184-8bf3-0002a40864fc@192.168.14.4
CSeq: 3 BYE
Date: Thu, 01 Jan 1970 00:29:12 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:105@192.168.14.55>
Content-Length: 0
Max-Forwards: 70



        Received SIP PDU from ( 192.168.14.50:5060 )
OPTIONS sip:192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.50:5060;branch=z9hG4bK035309e8;rport
From: "Unknown" <sip:Unknown@192.168.14.50>;tag=as62997fb5
To: <sip:192.168.14.55>
Contact: <sip:Unknown@192.168.14.50>
Call-ID: 3f46a33b046446e22489f70d08b107c3@192.168.14.50
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 Mar 2012 09:59:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0



        Sending SIP PDU to ( 192.168.14.50:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.50:5060;branch=z9hG4bK035309e8;rport=5060
From: "Unknown" <sip:Unknown@192.168.14.50>;tag=as62997fb5
To: <sip:192.168.14.55>
Call-ID: 3f46a33b046446e22489f70d08b107c3@192.168.14.50
CSeq: 102 OPTIONS
User-Agent: AddPac SIP Gateway
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Length: 0


51      <SIP    5>      : Transaction Client  (3 BYE) Timeout (retry #3)
52      <SIP    5>      : Send BYE Request

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
BYE sip:90339@192.168.14.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.55:5060;branch=z9hG4bKd0008504a43
From: <sip:105@192.168.14.55>;tag=d0008504a4
To: <sip:90339@192.168.14.4>;tag=ce4fa1f3a4
Call-ID: cebc704f-f21b-a184-8bf3-0002a40864fc@192.168.14.4
CSeq: 3 BYE
Date: Thu, 01 Jan 1970 00:29:12 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:105@192.168.14.55>
Content-Length: 0
Max-Forwards: 70


53      <SIP    5>      : Transaction Client  (3 BYE) Timeout (retry #4)
54      <SIP    5>      : Send BYE Request

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
BYE sip:90339@192.168.14.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.55:5060;branch=z9hG4bKd0008504a43
From: <sip:105@192.168.14.55>;tag=d0008504a4
To: <sip:90339@192.168.14.4>;tag=ce4fa1f3a4
Call-ID: cebc704f-f21b-a184-8bf3-0002a40864fc@192.168.14.4
CSeq: 3 BYE
Date: Thu, 01 Jan 1970 00:29:12 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:105@192.168.14.55>
Content-Length: 0
Max-Forwards: 70


Дэбаг с AP2640:

Код:
1    <SIP    14671>  : Set Terminated Success for 150 BYE
2       <SIP    14670>  : Set Terminated Success for 9752 INVITE

        Received SIP PDU from ( 192.168.14.4:5060 )
INVITE sip:102@192.168.14.53 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.4:5060;branch=z9hG4bKb34f9bf0a4512
From: <sip:90339@192.168.14.4>;tag=b34f9bf0a4
To: <sip:102@192.168.14.53>
Call-ID: b3bc704f-fba2-9b98-8bf0-0002a40864fc@192.168.14.4
CSeq: 512 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Mon, 26 Mar 2012 19:00:03 GMT
User-Agent: AddPac AP2620 8.30W
Contact: <sip:90339@192.168.14.4>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 176
Max-Forwards: 70
Remote-Party-ID: <sip:90339@192.168.14.4>;screen=yes;party=calling

v=0
o=90339 1332788403 1332788403 IN IP4 192.168.14.4
s=AddPac Gateway SDP
c=IN IP4 192.168.14.4
t=1332788403 0
m=audio 23094 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.14.4:5060;branch=z9hG4bKb34f9bf0a4512
From: <sip:90339@192.168.14.4>;tag=b34f9bf0a4
To: <sip:102@192.168.14.53>
Call-ID: b3bc704f-fba2-9b98-8bf0-0002a40864fc@192.168.14.4
CSeq: 512 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


3       <Call   14672>  : ******************  Call Created status(InitiatedByNet
)  *******************
4       <SIP    14672>  : Receive INVITE Request
5       <NetCon 14672>  : Found inbound voip peer by dest-pattern id(90)
6       <Call   14672>  : From Net - calledParty(102) callingParty(90339)
7       <Call   14672>  : MatchedPerfect
8       <Call   14672>  : MatchAllProcess After Sorted
                          <0>  id(101) dest(102) prefer(0) selected(344)
9       <Call   14672>  : Initiate callee with dial-peer(102) status(CalleeDeter
minedAll) id(00000000-0000-0000-0000-000000000000)
10      <CEP    000100> : InitiateOutCall :  calledNum(), callingNum(90339), cal
lerPort(ffffffff) type(FXS)
[2875096.240] RTA(0/1/0) Rx CC_RING_REQ [80 1f 01 08 30 33 32 36 31 35 35 36 02
05 39 30 33 33 39 07 0c 31 39 32 2e 31 36 38 2e 31 34 2e 34 ] peerId(-1)
[2875096.240] VM(0/1/0) DaTime [L=8] 30 33 32 36 31 35 35 36
[2875096.245] VM(0/1/0) CgNumb [L=5] 39 30 33 33 39
[2875096.245] VM(0/1/0) CgName [L=12] 31 39 32 2e 31 36 38 2e 31 34 2e 34
[2875096.245] VM(0/1/0) Line Reverse
[2875096.245] VM(0/1/0) Start ring actv
[2875096.245] VM(0/1/0) SW to -72V
11      <CEP    000100> : Outbound call to CEP callId(00000000-0000-0000-0000-00
0000000000) callNum(14672)
[2875096.245] VM(0/1/0) set T38 disable
[2875096.245] VM(0/1/0) Fax rate  9600
12      <SIP    14672>  : SetAlerting

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.14.4:5060;branch=z9hG4bKb34f9bf0a4512
From: <sip:90339@192.168.14.4>;tag=b34f9bf0a4
To: <sip:102@192.168.14.53>;tag=924fbf3ca4
Call-ID: b3bc704f-fba2-9b98-8bf0-0002a40864fc@192.168.14.4
CSeq: 512 INVITE
User-Agent: AddPac SIP Gateway
Contact: sip:102@192.168.14.53
Content-Length: 0


[2875097.245] VM(0/1/0) Gen ring idle
[2875097.745] VM(0/1/0) Tx CID enable
[2875097.745] VM(0/1/0) vopp enable
[2875097.745] VM(0/1/0) play mute
[2875097.805] VM(0/1/0) Tx CID ON
[2875097.860] VM(0/1/0) Rx CID_ACK
[2875097.860] VM(0/1/0) Tx CID DATA [L=68] 80 01 1f 02 01 05 08 06 30 08 33 08 3
2 08 36 08 31 07 35 07 35 07 36 07 02 05 05 06 39 09 30 09 33 09 33 09 39 09 07
05 0c 06 31 0b 39 0b 32 0b 2e 0b 31 0b 36 0b 38 0b 2e 0b 31 0b 34 0b 2e 0b 34 0b
 00 0f
[2875098.860] VM(0/1/0) Tx CID fin
[2875098.860] VM(0/1/0) vopp idle
[2875098.995] VM(0/1/0) vmOffHook
[2875099.055] VM(0/1/0) vmTmoOffHook
[2875099.055] VM(0/1/0) SW to -48V
[2875099.055] VM(0/1/0) Line Forward
[2875099.115] VM(0/1/0) vmTmoOffHook
[2875099.115] VM(0/1/0) Rx OffHook
[2875099.115] VM(0/1/0) vopp enable
[2875099.115] VM(0/1/0) T38 Fax disabled
[2875099.115] VM(0/1/0) Tx CONNECT_CNF
13      <Call   14672>  : Connected from(100)
[2875099.115] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
   VAD_CTRL 0
[2875099.115] VM(0/1/0) VAD disable
[2875099.115] VM(0/1/0) SID enable by CCC
14      <SIP    14672>  : SetConnected
15      <SIP    14672>  : Add Local Audio MediaFormat : 8

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.4:5060;branch=z9hG4bKb34f9bf0a4512
From: <sip:90339@192.168.14.4>;tag=b34f9bf0a4
To: <sip:102@192.168.14.53>;tag=924fbf3ca4
Call-ID: b3bc704f-fba2-9b98-8bf0-0002a40864fc@192.168.14.4
CSeq: 512 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:102@192.168.14.53
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 178

v=0
o=102 1332777365 1332777365 IN IP4 192.168.14.53
s=AddPac Gateway SDP
c=IN IP4 192.168.14.53
t=1332777365 0
m=audio 23638 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1
a=ptime:20

[2875099.120] RTA(0/1/0) Rx RS_LISTEN_REQ callId=14672 ssId=1 G711A
        peer=192.168.14.4 mp=23638/23639 hp=23094/23095
[2875099.120] VM(0/1/0) vopp idle
[2875099.120] VM(0/1/0) start codec replace timer to G711A
[2875099.120] RTA(0/1/0) Rx RS_OPEN_REQ callId=14672 ssId=1 G711A
        peer=192.168.14.4 mp=23638/23639 hp=23094/23095
[2875099.120] VM(0/1/0) under codec replace to G711A
[2875099.120] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
   DTMF_CTRL 0
[2875099.120] VM(0/1/0) DTMF disable

        Received SIP PDU from ( 192.168.14.4:5060 )
ACK sip:102@192.168.14.53 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.4:5060;branch=z9hG4bKb34f9bf0a4512
From: <sip:90339@192.168.14.4>;tag=b34f9bf0a4
To: <sip:102@192.168.14.53>;tag=924fbf3ca4
Call-ID: b3bc704f-fba2-9b98-8bf0-0002a40864fc@192.168.14.4
CSeq: 512 ACK
Content-Length: 0
Max-Forwards: 70

16      <SIP    14672>  : ACK received
17      <SIP    14672>  : Receive ACK Request
18      <SIP    14672>  : Set Terminated Success for 512 INVITE
[2875099.180] VM(0/1/0) vopp enable
[2875099.180] VM(0/1/0) codec replaced to G711A
[2875099.180] VM(0/1/0) T38 Fax disabled
[2875099.180] VM(0/1/0) play mute
[2875110.355] VM(0/1/0) vmOnHook
[2875110.405] VM(0/1/0) vmTmoOnHook
[2875110.455] VM(0/1/0) vmTmoOnHook
[2875110.505] VM(0/1/0) vmTmoOnHook
[2875110.555] VM(0/1/0) vmTmoOnHook
[2875110.605] VM(0/1/0) vmTmoOnHook
[2875110.655] VM(0/1/0) vmTmoOnHook
[2875110.705] VM(0/1/0) vmTmoOnHook
[2875110.755] VM(0/1/0) vmTmoOnHook
[2875110.805] VM(0/1/0) vmTmoOnHook
[2875110.855] VM(0/1/0) vmTmoOnHook
[2875110.905] VM(0/1/0) vmTmoOnHook
[2875110.955] VM(0/1/0) vmTmoOnHook
[2875111.005] VM(0/1/0) vmTmoOnHook
[2875111.055] VM(0/1/0) vmTmoOnHook
[2875111.105] VM(0/1/0) vmTmoOnHook
[2875111.155] VM(0/1/0) vmTmoOnHook
[2875111.205] VM(0/1/0) vmOffHook
[2875111.265] VM(0/1/0) vmTmoOffHook
[2875111.265] VM(0/1/0) Rx OffHook
[2875111.265] VM(0/1/0) Tx FLASH_IND
19      <CEP    000100> : Hook Flashed
20      <Call   14672>  : Hold Request from Channel.
21      <SIP    14672>  : Re-INVITE send
22      <SIP    0>      : No authentication information available
23      <SIP    14672>  : Send INVITE Request

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
INVITE sip:90339@192.168.14.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bK924fbf3ca49753
From: <sip:102@192.168.14.53>;tag=924fbf3ca4
To: <sip:90339@192.168.14.4>;tag=b34f9bf0a4
Call-ID: b3bc704f-fba2-9b98-8bf0-0002a40864fc@192.168.14.4
CSeq: 9753 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Mon, 26 Mar 2012 15:56:17 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:102@192.168.14.53>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 184
Max-Forwards: 70

v=0
o=102 1332777365 1332777365 IN IP4 192.168.14.53
s=AddPac Gateway SDP
c=IN IP4 0.0.0.0
t=1332777365 0
a=sendonly
m=audio 23638 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1
a=ptime:20


        Received SIP PDU from ( 192.168.14.4:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bK924fbf3ca49753
From: <sip:102@192.168.14.53>;tag=924fbf3ca4
To: <sip:90339@192.168.14.4>;tag=b34f9bf0a4
Call-ID: b3bc704f-fba2-9b98-8bf0-0002a40864fc@192.168.14.4
CSeq: 9753 INVITE
User-Agent: AddPac AP2620 8.30W
Contact: sip:90339@192.168.14.4
Content-Type: application/sdp
Content-Length: 190

v=0
o=90339 1332788418 1332788418 IN IP4 192.168.14.4
s=AddPac Gateway SDP
c=IN IP4 192.168.14.4
t=1332788418 0
a=recvonly
m=audio 23094 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1
a=ptime:20
24      <SIP    14672>  : Receive 200 OK
25      <SIP    14672>  : Get SIP Audio MediaFormat : 8
26      <SIP    14672>  : SetRemoteSocketInfo : ip(192.168.14.4) port(23094)
27      <Call   14672>  : Connected from(fffffffe)
28      <NetEP  14672>  : Call with 192.168.14.4 established
29      <Call   14672>  : Session switch request from Network.
30      <CEP    000100> : Session Switch : current(14672) , hold(-1)
[2875111.290] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=2 rawDataLen=0
   SESS_HOLD
   SESS_NEW OFF_HOOK
[2875111.290] VM(0/1/0) vopp idle
[2875111.290] VM(0/1/0) vopp enable
[2875111.290] VM(0/1/0) play Dial tone
31      <CEP    000100> : Call Received
32      <CEP    000100> : Call Initiated : calledNumber() crv(0) total(1)
33      <Call   14673>  : ******************  Call Created status(InitiatedByFXS
)  *******************
34      <CEP    000100> : Calling number(102)
35      <CEP    000100> : Call id(a191704f-222d-16f5-813e-0002a4046af2) callNum(
14673)
36      <SIP    14672>  : Received INVITE OK response
37      <SIP    14672>  : Send ACK Request

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
ACK sip:90339@192.168.14.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bK924fbf3ca49753
From: <sip:102@192.168.14.53>;tag=924fbf3ca4
To: <sip:90339@192.168.14.4>;tag=b34f9bf0a4
Call-ID: b3bc704f-fba2-9b98-8bf0-0002a40864fc@192.168.14.4
CSeq: 9753 ACK
Content-Length: 0
Max-Forwards: 70


38      <SIP    14672>  : Check Event Relation
39      <SIP    14672>  : Set Terminated Success for 9753 INVITE
[2875112.355] VM(0/1/0) Tx DIGIT_IND '1'
[2875112.355] VM(0/1/0) play mute
40      <Call   14673>  : Digit(1) at InitiatedByFXS
41      <Call   14673>  : MatchedPartially

        Received SIP PDU from ( 192.168.14.50:5060 )
OPTIONS sip:192.168.14.53 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.50:5060;branch=z9hG4bK33f3c617;rport
From: "Unknown" <sip:Unknown@192.168.14.50>;tag=as23b8ba94
To: <sip:192.168.14.53>
Contact: <sip:Unknown@192.168.14.50>
Call-ID: 59f76b6b016bba2f22c3eea851156a0a@192.168.14.50
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 Mar 2012 09:58:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


        Sending SIP PDU to ( 192.168.14.50:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.50:5060;branch=z9hG4bK33f3c617;rport
From: "Unknown" <sip:Unknown@192.168.14.50>;tag=as23b8ba94
To: <sip:192.168.14.53>
Call-ID: 59f76b6b016bba2f22c3eea851156a0a@192.168.14.50
CSeq: 102 OPTIONS
User-Agent: AddPac SIP Gateway
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Length: 0


[2875112.905] VM(0/1/0) Tx DIGIT_IND '0'
42      <Call   14673>  : Digit(0) at CalleeUndetermined
43      <Call   14673>  : MatchedPartially
[2875113.365] VM(0/1/0) Tx DIGIT_IND '5'
44      <Call   14673>  : Digit(5) at CalleeUndetermined
45      <Call   14673>  : MatchedPerfect
46      <Call   14673>  : MatchAllProcess After Sorted
                          <0>  id(1105) dest(105) prefer(0) selected(206)
                          <1>  id(1100) dest(1..) prefer(1) selected(8)
47      <Call   14673>  : Initiate callee with dial-peer(105) status(CalleeDeter
minedAll) id(a191704f-222d-16f5-813e-0002a4046af2)
48      <NetEP  14673>  : InitiateOutCall: calledNum(105) callingNum(102) target
(192.168.14.55)
49      <NetEP  14673>  : DoCall: calledAddr(sip:105@192.168.14.55) callingAddr(
83102)
[2875113.365] VM(0/1/0) set T38 disable
[2875113.365] VM(0/1/0) Fax rate  9600
50      <SIP    0>      : No authentication information available
51      <SIP    14673>  : Send INVITE Request

        Sending SIP PDU to ( 192.168.14.55:5060 ) from 5060
INVITE sip:105@192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49754
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9754 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Mon, 26 Mar 2012 15:56:19 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:83102@192.168.14.53>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 234
Max-Forwards: 70

v=0
o=83102 1332777379 1332777379 IN IP4 192.168.14.53
s=AddPac Gateway SDP
c=IN IP4 192.168.14.53
t=1332777379 0
m=audio 23640 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

[2875113.370] RTA(0/1/0) Rx RS_LISTEN_REQ callId=14673 ssId=1 G711U
        peer=0.0.0.0 mp=23640/23641 hp=0/0
[2875113.370] VM(0/1/0) codec replace later to G711U

        Received SIP PDU from ( 192.168.14.55:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49754
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9754 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0

52      <SIP    14673>  : Receive 100 Trying
53      <SIP    14673>  : Transaction (9754 INVITE) proceeding

        Received SIP PDU from ( 192.168.14.55:5060 )
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49754
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>;tag=c600b703a4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9754 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:105@192.168.14.55
Content-Length: 0

54      <SIP    14673>  : Receive 180 Ringing
55      <SIP    14673>  : Transaction (9754 INVITE) proceeding
56      <Call   14673>  : Alert from(fffffffe) pseudo(0) inband(0) status(Callee
Initiated)
[2875113.400] RTA(0/1/0) Rx CC_ALERT_RSP peerId(0/0/0)
[2875113.400] VM(0/1/0) play RingBack tone
[2875113.535] VM(0/1/0) codec same G711U

        Received SIP PDU from ( 192.168.14.55:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49754
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>;tag=c600b703a4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9754 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:105@192.168.14.55
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 215

v=0
o=105 1738 1738 IN IP4 192.168.14.55
s=AddPac Gateway SDP
c=IN IP4 192.168.14.55
t=0 0
m=audio 23014 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
57      <SIP    14673>  : Receive 200 OK
58      <SIP    14673>  : Get SIP Audio MediaFormat : 8
59      <SIP    14673>  : SetRemoteSocketInfo : ip(192.168.14.55) port(23014)
[2875116.810] RTA(0/1/0) Rx RS_OPEN_REQ callId=14673 ssId=1 G711A
        peer=192.168.14.55 mp=23640/23641 hp=23014/23015
[2875116.810] VM(0/1/0) vopp idle
[2875116.810] VM(0/1/0) start codec replace timer to G711A
[2875116.810] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
   DTMF_CTRL 1
[2875116.810] VM(0/1/0) DTMF_RTP_RFC2833 enable
[2875116.810] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
   Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65
60      <Call   14673>  : Connected from(fffffffe)
[2875116.810] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
   VAD_CTRL 0
[2875116.810] VM(0/1/0) VAD disable
[2875116.810] VM(0/1/0) SID enable by CCC
[2875116.810] RTA(0/1/0) Rx CC_CONNECT_RSP peerId(0/0/0)
[2875116.810] VM(0/1/0) T38 Fax disabled
61      <NetEP  14673>  : Call with sip:105@192.168.14.55 established
62      <SIP    14673>  : Received INVITE OK response
63      <SIP    14673>  : Send ACK Request
[2875116.815] VM(0/1/0) discard voice under codec replace

        Sending SIP PDU to ( 192.168.14.55:5060 ) from 5060
ACK sip:105@192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49754
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>;tag=c600b703a4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9754 ACK
Content-Length: 0
Max-Forwards: 70


64      <SIP    14673>  : Check Event Relation
65      <SIP    14673>  : Set Terminated Success for 9754 INVITE
[2875116.825] VM(0/1/0) discard voice under codec replace
[2875116.870] VM(0/1/0) vopp enable
[2875116.870] VM(0/1/0) codec replaced to G711A
[2875116.870] VM(0/1/0) T38 Fax disabled
[2875116.870] VM(0/1/0) play mute
[2875116.890] VM(0/1/0) codec same G711A
[2875116.890] VM(0/1/0) Rx RTP replace codec to G711A
[2875121.815] VM(0/1/0) vmOnHook
[2875121.865] VM(0/1/0) vmTmoOnHook
[2875121.915] VM(0/1/0) vmTmoOnHook
[2875121.965] VM(0/1/0) vmTmoOnHook
[2875122.015] VM(0/1/0) vmTmoOnHook
[2875122.065] VM(0/1/0) vmTmoOnHook
[2875122.115] VM(0/1/0) vmTmoOnHook
[2875122.165] VM(0/1/0) vmTmoOnHook
[2875122.215] VM(0/1/0) vmTmoOnHook
[2875122.265] VM(0/1/0) vmTmoOnHook
[2875122.315] VM(0/1/0) vmTmoOnHook
[2875122.365] VM(0/1/0) vmTmoOnHook
[2875122.415] VM(0/1/0) vmTmoOnHook
[2875122.465] VM(0/1/0) vmTmoOnHook
[2875122.515] VM(0/1/0) vmTmoOnHook
[2875122.565] VM(0/1/0) vmTmoOnHook
[2875122.615] VM(0/1/0) vmTmoOnHook
[2875122.665] VM(0/1/0) vmTmoOnHook
[2875122.715] VM(0/1/0) vmTmoOnHook
[2875122.715] VM(0/1/0) Rx OnHook
[2875122.715] VM(0/1/0) vopp idle
[2875122.715] VM(0/1/0) Tx DISCONN_CNF
66      <CEP    000100> : Disconnected(16) at Busy crv(0)
67      <Call   14673>  : Hold Request from Channel.
68      <SIP    14673>  : Re-INVITE send
69      <SIP    0>      : No authentication information available
70      <SIP    14673>  : Send INVITE Request

        Sending SIP PDU to ( 192.168.14.55:5060 ) from 5060
INVITE sip:105@192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49755
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>;tag=c600b703a4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9755 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Mon, 26 Mar 2012 15:56:29 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:83102@192.168.14.53>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 244
Max-Forwards: 70

v=0
o=83102 1332777379 1332777379 IN IP4 192.168.14.53
s=AddPac Gateway SDP
c=IN IP4 0.0.0.0
t=1332777379 0
a=sendonly
m=audio 23640 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=ptime:20

[2875122.720] RTA(0/1/0) Rx RS_CLOSE_REQ callId=14673 ssId=1 dir=reve
[2875122.720] RTA(0/1/0) Rx RS_CLOSE_REQ callId=14673 ssId=1 dir=forw
[2875122.720] RTA(0/1/0) close Media socket
[2875122.720] RTA(0/1/0) close RTCP socket
[2875122.720] RTA(0/1/0) Rx RS_CLOSE_REQ callId=14672 ssId=1 dir=reve
[2875122.720] RTA(0/1/0) Rx RS_CLOSE_REQ callId=14672 ssId=1 dir=forw
[2875122.720] RTA(0/1/0) close Media socket
[2875122.720] RTA(0/1/0) close RTCP socket

        Received SIP PDU from ( 192.168.14.55:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49755
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>;tag=c600b703a4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9755 INVITE
User-Agent: AddPac SIP Gateway
Contact: sip:105@192.168.14.55
Content-Type: application/sdp
Content-Length: 227

v=0
o=105 1744 1744 IN IP4 192.168.14.55
s=AddPac Gateway SDP
c=IN IP4 192.168.14.55
t=0 0
a=recvonly
m=audio 23014 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
71      <SIP    14673>  : Receive 200 OK
72      <Call   14673>  : Connected from(fffffffe)
73      <NetEP  14673>  : Call with sip:105@192.168.14.55 established
74      <Call   14673>  : Session switch request from Network.
75      <SIP    14673>  : Received INVITE OK response
76      <SIP    14673>  : Send ACK Request

        Sending SIP PDU to ( 192.168.14.55:5060 ) from 5060
ACK sip:105@192.168.14.55 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bKa34f143fa49755
From: "Polina_Gomzina" <sip:83102@192.168.14.53>;tag=a34f143fa4
To: <sip:105@192.168.14.55>;tag=c600b703a4
Call-ID: a391704f-b833-14bc-813f-0002a4046af2@192.168.14.53
CSeq: 9755 ACK
Content-Length: 0
Max-Forwards: 70


77      <SIP    14673>  : Check Event Relation
78      <Call   14673>  : Start Attended Transfer.
79      <SIP    14672>  : REFER send
80      <SIP    14672>  : Send REFER Request

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
REFER sip:90339@192.168.14.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bK924fbf3ca49756
From: <sip:102@192.168.14.53>;tag=924fbf3ca4
To: <sip:90339@192.168.14.4>;tag=b34f9bf0a4
Call-ID: b3bc704f-fba2-9b98-8bf0-0002a40864fc@192.168.14.4
CSeq: 9756 REFER
Referred-By: <sip:102@192.168.14.53>
Date: Mon, 26 Mar 2012 15:56:29 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:102@192.168.14.53>
Refer-To: <sip:105@192.168.14.55?Replaces=a391704f-b833-14bc-813f-0002a4046af2@1
92.168.14.53;to-tag=c600b703a4;from-tag=a34f143fa4>
Expires: 180
Content-Length: 0
Max-Forwards: 70


81      <SIP    14673>  : Set Terminated Success for 9755 INVITE

        Received SIP PDU from ( 192.168.14.4:5060 )
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.14.53:5060;branch=z9hG4bK924fbf3ca49756
From: <sip:102@192.168.14.53>;tag=924fbf3ca4
To: <sip:90339@192.168.14.4>;tag=b34f9bf0a4
Call-ID: b3bc704f-fba2-9b98-8bf0-0002a40864fc@192.168.14.4
CSeq: 9756 REFER
User-Agent: AddPac AP2620 8.30W
Content-Length: 0

82      <SIP    14672>  : Receive 202 Accepted
83      <SIP    14672>  : Transaction (9756 REFER) completed

        Received SIP PDU from ( 192.168.14.4:5060 )
NOTIFY sip:102@192.168.14.53 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.4:5060;branch=z9hG4bKb34f9bf0a4513
From: <sip:90339@192.168.14.4>;tag=b34f9bf0a4
To: <sip:102@192.168.14.53>;tag=924fbf3ca4
Call-ID: b3bc704f-fba2-9b98-8bf0-0002a40864fc@192.168.14.4
CSeq: 513 NOTIFY
Date: Mon, 26 Mar 2012 19:00:30 GMT
User-Agent: AddPac AP2620 8.30W
Contact: <sip:90339@192.168.14.4>
Subscription-State: active;expires=180
Event: refer
Content-Type: message/sipfrag
Content-Length: 22
Max-Forwards: 70

SIP/2.0 100 Trying

84      <SIP    14672>  : NOTIFY received

        Sending SIP PDU to ( 192.168.14.4:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.4:5060;branch=z9hG4bKb34f9bf0a4513
From: <sip:90339@192.168.14.4>;tag=b34f9bf0a4
To: <sip:102@192.168.14.53>;tag=924fbf3ca4
Call-ID: b3bc704f-fba2-9b98-8bf0-0002a40864fc@192.168.14.4
CSeq: 513 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0




Заранее прошу прощенья, что прикладываю дебаг в том виде, в котором получил(без удаления строк, не касающихся наших абонентов и их звонков),т.к. боюсь удалить лишнее. Дэбаг с АР2620 выкладывать не стал по причине того, что аппарат активно используется и вычленить необходимые строки я не могу.
Прошу оказать помощь в исправлении данной ошибки: самостоятельно разобраться не в силах из-за незнания RFC.

Заранее спасибо!
Автор:  АдминАдм [ 06 апр 2012, 10:23 ]
Заголовок сообщения:  Re: Нет перевода звонка(tranfer) в связке AP2620-AP2640-APIP
Как выяснилось в процессе разбора полетов, все беда была из-за AddPac2620, который "подвисал" при переводе звонка. Т.е. после получения на 2620 "100 Trying" от IP100 с шлюзом рвалась связь, на нем обнулялся Running time, падал Е1 между ним и Nortel. Менее чем через 20 сек. работоспособность шлюза восстанавливалась.

Решеили проблему путем перехода на предыдущую версию прошивки:
с ap2620rom_v8_30W.BIN на ap2620rom_v8_30U.BIN
Спасибо представителю AddPac в оказании помощи!
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