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 Заголовок сообщения: Re: call forward
СообщениеДобавлено: 09 сен 2014 21:37 

Зарегистрирован: 21 авг 2014 14:58
Сообщения: 40
Спасибо. Внес изменения в конфиг завтра с утра попробуем, когда пользователи будут на месте.


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 Заголовок сообщения: Re: call forward
СообщениеДобавлено: 10 сен 2014 16:26 

Зарегистрирован: 21 авг 2014 14:58
Сообщения: 40
Добрый день.
Потестил, телефоны в одном офисе. Пытаюсь сделать трансфер с GrandStream JXP-2000.
[+] 
Код:
2014-09-10 16:19:56.510418 [INFO] mod_dialplan_xml.c:558 Processing Popov2 Alex <193>->att_xfer in context features
Dialplan: sofia/internal/sip:110@109.197.55.18:1034 parsing [features->dx] continue=false
Dialplan: sofia/internal/sip:110@109.197.55.18:1034 Regex (FAIL) [dx] destination_number(att_xfer) =~ /^dx$/ break=on-false
Dialplan: sofia/internal/sip:110@109.197.55.18:1034 parsing [features->att_xfer] continue=false
Dialplan: sofia/internal/sip:110@109.197.55.18:1034 Regex (PASS) [att_xfer] destination_number(att_xfer) =~ /^att_xfer$/ break=on-false
Dialplan: sofia/internal/sip:110@109.197.55.18:1034 Action read(3 4 'tone_stream://%(10000,0,350,440)' digits 30000 #)
Dialplan: sofia/internal/sip:110@109.197.55.18:1034 Action set(origination_cancel_key=#)
Dialplan: sofia/internal/sip:110@109.197.55.18:1034 Action att_xfer(user/${digits})
2014-09-10 16:19:56.510418 [NOTICE] switch_core_session.c:2986 Execute read(3 4 'tone_stream://%(10000,0,350,440)' digits 30000 #)
EXECUTE sofia/internal/sip:110@109.197.55.18:1034 read(3 4 'tone_stream://%(10000,0,350,440)' digits 30000 #)
2014-09-10 16:19:56.510418 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@16000hz 1 channels 20ms
2014-09-10 16:19:56.710420 [DEBUG] switch_rtp.c:5831 RTP RECV DTMF 1:1120
2014-09-10 16:19:56.710420 [DEBUG] switch_channel.c:487 RECV DTMF 1:1120
2014-09-10 16:19:56.710420 [DEBUG] switch_ivr_play_say.c:1714 done playing file tone_stream://%(10000,0,350,440)
2014-09-10 16:19:57.130415 [DEBUG] switch_rtp.c:5831 RTP RECV DTMF 1:1280
2014-09-10 16:19:57.130415 [DEBUG] switch_channel.c:487 RECV DTMF 1:1280
2014-09-10 16:19:57.470407 [DEBUG] switch_rtp.c:5831 RTP RECV DTMF 4:1280
2014-09-10 16:19:57.470407 [DEBUG] switch_channel.c:487 RECV DTMF 4:1280
2014-09-10 16:19:58.330420 [DEBUG] switch_rtp.c:5831 RTP RECV DTMF #:1600
2014-09-10 16:19:58.330420 [DEBUG] switch_channel.c:487 RECV DTMF #:1600
2014-09-10 16:19:58.330420 [NOTICE] switch_core_session.c:2986 Execute set(origination_cancel_key=#)
EXECUTE sofia/internal/sip:110@109.197.55.18:1034 set(origination_cancel_key=#)
2014-09-10 16:19:58.330420 [DEBUG] mod_dptools.c:1435 sofia/internal/sip:110@109.197.55.18:1034 SET [origination_cancel_key]=[#]
2014-09-10 16:19:58.330420 [NOTICE] switch_core_session.c:2986 Execute att_xfer(user/${digits})
EXECUTE sofia/internal/sip:110@109.197.55.18:1034 att_xfer(user/114)
2014-09-10 16:19:58.330420 [DEBUG] switch_channel.c:1765 (sofia/internal/sip:110@109.197.55.18:1034) Callstate Change ACTIVE -> RING_WAIT
2014-09-10 16:19:58.330420 [DEBUG] switch_channel.c:1200 sofia/internal/sip:110@109.197.55.18:1034 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 10 Sep 2014 16:19:35 +0400] to event
2014-09-10 16:19:58.330420 [DEBUG] switch_channel.c:1200 sofia/internal/sip:110@109.197.55.18:1034 EXPORTING[export_vars] [dialed_extension]=[110] to event
2014-09-10 16:19:58.330420 [DEBUG] switch_ivr_originate.c:2079 Parsing global variables
2014-09-10 16:19:58.330420 [DEBUG] switch_channel.c:1200 sofia/internal/sip:110@109.197.55.18:1034 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 10 Sep 2014 16:19:35 +0400] to event
2014-09-10 16:19:58.330420 [DEBUG] switch_channel.c:1200 sofia/internal/sip:110@109.197.55.18:1034 EXPORTING[export_vars] [dialed_extension]=[110] to event
2014-09-10 16:19:58.330420 [DEBUG] switch_ivr_originate.c:2079 Parsing global variables
2014-09-10 16:19:58.330420 [DEBUG] switch_event.c:1688 Parsing variable [sip_invite_domain]=[31.184.192.155]
2014-09-10 16:19:58.330420 [DEBUG] switch_event.c:1688 Parsing variable [presence_id]=[114@31.184.192.155]
2014-09-10 16:19:58.330420 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/sip:114@109.197.55.18:1025 [c7c7ce3a-38e4-11e4-9e92-132074d1c88e]
2014-09-10 16:19:58.330420 [DEBUG] mod_sofia.c:4564 (sofia/internal/sip:114@109.197.55.18:1025) State Change CS_NEW -> CS_INIT
2014-09-10 16:19:58.330420 [DEBUG] switch_core_session.c:1387 Send signal sofia/internal/sip:114@109.197.55.18:1025 [BREAK]
2014-09-10 16:19:58.330420 [DEBUG] mod_sofia.c:4634 [zrtp_passthru] Setting a-leg inherit_codec=true
2014-09-10 16:19:58.330420 [DEBUG] mod_sofia.c:4637 [zrtp_passthru] Setting b-leg absolute_codec_string='G722@8000h@20i@64000b'
2014-09-10 16:19:58.330420 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:114@109.197.55.18:1025) Running State Change CS_INIT
2014-09-10 16:19:58.330420 [DEBUG] switch_core_state_machine.c:510 (sofia/internal/sip:114@109.197.55.18:1025) State INIT
2014-09-10 16:19:58.330420 [DEBUG] mod_sofia.c:87 sofia/internal/sip:114@109.197.55.18:1025 SOFIA INIT
2014-09-10 16:19:58.330420 [DEBUG] sofia_glue.c:1226 sofia/internal/sip:114@109.197.55.18:1025 sending invite version: 1.4.7  32bit
Local SDP:
v=0
o=FreeSWITCH 1410329192 1410329193 IN IP4 31.184.192.155
s=FreeSWITCH
c=IN IP4 31.184.192.155
t=0 0
m=audio 22406 RTP/AVP 9 101 13
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2014-09-10 16:19:58.330420 [DEBUG] switch_core_state_machine.c:40 sofia/internal/sip:114@109.197.55.18:1025 Standard INIT
2014-09-10 16:19:58.330420 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/sip:114@109.197.55.18:1025) State Change CS_INIT -> CS_ROUTING
2014-09-10 16:19:58.330420 [DEBUG] switch_core_session.c:1387 Send signal sofia/internal/sip:114@109.197.55.18:1025 [BREAK]
2014-09-10 16:19:58.330420 [DEBUG] switch_core_state_machine.c:510 (sofia/internal/sip:114@109.197.55.18:1025) State INIT going to sleep
2014-09-10 16:19:58.330420 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:114@109.197.55.18:1025) Running State Change CS_ROUTING
2014-09-10 16:19:58.330420 [DEBUG] switch_core_state_machine.c:526 (sofia/internal/sip:114@109.197.55.18:1025) State ROUTING
2014-09-10 16:19:58.330420 [DEBUG] mod_sofia.c:123 sofia/internal/sip:114@109.197.55.18:1025 SOFIA ROUTING
2014-09-10 16:19:58.330420 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:114@109.197.55.18:1025) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2014-09-10 16:19:58.330420 [DEBUG] switch_core_session.c:1387 Send signal sofia/internal/sip:114@109.197.55.18:1025 [BREAK]
2014-09-10 16:19:58.330420 [DEBUG] switch_core_state_machine.c:526 (sofia/internal/sip:114@109.197.55.18:1025) State ROUTING going to sleep
2014-09-10 16:19:58.330420 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:114@109.197.55.18:1025) Running State Change CS_CONSUME_MEDIA
2014-09-10 16:19:58.330420 [DEBUG] switch_core_state_machine.c:545 (sofia/internal/sip:114@109.197.55.18:1025) State CONSUME_MEDIA
2014-09-10 16:19:58.330420 [DEBUG] switch_core_state_machine.c:545 (sofia/internal/sip:114@109.197.55.18:1025) State CONSUME_MEDIA going to sleep
2014-09-10 16:19:58.330420 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/sip:114@109.197.55.18:1025 [BREAK]
2014-09-10 16:19:58.330420 [DEBUG] sofia.c:6373 Channel sofia/internal/sip:114@109.197.55.18:1025 entering state [calling][0]
2014-09-10 16:19:58.350978 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/sip:114@109.197.55.18:1025 [BREAK]
2014-09-10 16:19:58.350978 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/sip:114@109.197.55.18:1025 [BREAK]
2014-09-10 16:19:58.350978 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/sip:114@109.197.55.18:1025 [BREAK]
2014-09-10 16:19:58.350978 [DEBUG] sofia.c:6373 Channel sofia/internal/sip:114@109.197.55.18:1025 entering state [terminated][488]
2014-09-10 16:19:58.350978 [NOTICE] sofia.c:7256 Hangup sofia/internal/sip:114@109.197.55.18:1025 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]

И опять почему-то INCOMPATIBLE_DESTINATION....друзья, помогите. где засада?


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 Заголовок сообщения: Re: call forward
СообщениеДобавлено: 10 сен 2014 16:43 
FreeSWITCH_GuRu

Зарегистрирован: 22 авг 2012 09:52
Сообщения: 1710
покажите dial-string в профилях directory

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ЛС: @rostel
Сообщество: @ru_freeswitch


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 Заголовок сообщения: Re: call forward
СообщениеДобавлено: 10 сен 2014 17:39 

Зарегистрирован: 21 авг 2014 14:58
Сообщения: 40
Честно, не совсем понял, о чем речь идет.
Типовой конфиг sip-пользователя:
<include>
<user id="110">
<params>
<param name="password" value="<пароль>"/>
<param name="vm-password" value="110"/>
</params>
<variables>
<variable name="toll_allow" value="domestic,international,local"/>
<variable name="accountcode" value="110"/>
<variable name="user_context" value="default"/>
<variable name="effective_caller_id_name" value="Extension 110"/>
<variable name="effective_caller_id_number" value="110"/>
<variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
<variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
</variables>
</user>
</include>

В default.xml
<domain name="$${domain}">
<params>
<param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/>
</params>

то есть все по дефолту.


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 Заголовок сообщения: Re: call forward
СообщениеДобавлено: 10 сен 2014 19:44 
FreeSWITCH_GuRu

Зарегистрирован: 22 авг 2012 09:52
Сообщения: 1710
вроде норма

снимайте дамп сигналки
смущает
Код:
2014-09-10 16:19:58.330420 [DEBUG] mod_sofia.c:4637 [zrtp_passthru] Setting b-leg absolute_codec_string='G722@8000h@20i@64000b'

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Сообщество: @ru_freeswitch


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 Заголовок сообщения: Re: call forward
СообщениеДобавлено: 10 сен 2014 21:32 

Зарегистрирован: 21 авг 2014 14:58
Сообщения: 40
Обычно этой командой смотрел sip-трафик:
sofia profile internal siptrace on, но боюсь в рабочее время может что-то замешаться ненужное...
нет ли возможности в FS как-то сузить снятие дампа до необходимого номера/пользователя?


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 Заголовок сообщения: Re: call forward
СообщениеДобавлено: 10 сен 2014 21:36 
FreeSWITCH_GuRu

Зарегистрирован: 22 авг 2012 09:52
Сообщения: 1710
tcpdump c фильтрами по IP

https://confluence.freeswitch.org/displ ... et+Capture

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 Заголовок сообщения: Re: call forward
СообщениеДобавлено: 10 сен 2014 23:59 

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так а желательно в pcap формате собрать для анализа wireshark. Спасибо.завтра с утра сделаю.


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 Заголовок сообщения: Re: call forward
СообщениеДобавлено: 11 сен 2014 10:30 

Зарегистрирован: 21 авг 2014 14:58
Сообщения: 40
снял дамп sip с проблемного телефона с попыткой перевода звонка:
[+] 
Код:
10:07:10.850153 IP (tos 0x0, ttl 64, id 23890, offset 0, flags [none], proto UDP (17), length 1252)
    ip-ats.sip > ip-phone.activesync: [bad udp cksum 63d0!] UDP, length 1224
E...]R..@.......m.7....
....INVITE sip:110@ip-phone:1034;transport=udp SIP/2.0
Via: SIP/2.0/UDP ip-ats;rport;branch=z9hG4bKHN361Xvtpv1ea
Max-Forwards: 69
From: "Popov2 Alex" <sip:193@ip-ats>;tag=Sr25r27j8Fr0Q
To: <sip:110@ip-phone:1034;transport=udp>
Call-ID: b57f6f3a-b41c-1232-d78d-52540046df15
CSeq: 64871751 INVITE
Contact: <sip:mod_sofia@ip-ats:5060>
User-Agent: FreeSWITCH-mod_sofia/1.4.7~32bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 275
X-FS-Support: update_display,send_info
Remote-Party-ID: "Popov2 Alex" <sip:193@ip-ats>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1410395876 1410395877 IN IP4 ip-ats
s=FreeSWITCH
c=IN IP4 ip-ats
t=0 0
m=audio 19754 RTP/AVP 9 0 8 101 13
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

10:07:10.881618 IP (tos 0x0, ttl 249, id 3136, offset 0, flags [none], proto UDP (17), length 359)
    ip-phone.activesync > ip-ats.sip: [udp sum ok] UDP, length 331
E..g.@..../.m.7......
...S..SIP/2.0 100 Trying
Via: SIP/2.0/UDP ip-ats;rport;branch=z9hG4bKHN361Xvtpv1ea
From: "Popov2 Alex" <sip:193@ip-ats>;tag=Sr25r27j8Fr0Q
To: <sip:110@ip-phone:1034;transport=udp>
Call-ID: b57f6f3a-b41c-1232-d78d-52540046df15
CSeq: 64871751 INVITE
User-Agent: Grandstream GXP2000 1.2.5.3
Content-Length: 0


10:07:10.890301 IP (tos 0x0, ttl 249, id 3137, offset 0, flags [none], proto UDP (17), length 521)
    ip-phone.activesync > ip-ats.sip: [udp sum ok] UDP, length 493
E..     .A.....xm.7......
....    .SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ip-ats;rport;branch=z9hG4bKHN361Xvtpv1ea
From: "Popov2 Alex" <sip:193@ip-ats>;tag=Sr25r27j8Fr0Q
To: <sip:110@ip-phone:1034;transport=udp>;tag=790c6e2be65c976a
Call-ID: b57f6f3a-b41c-1232-d78d-52540046df15
CSeq: 64871751 INVITE
User-Agent: Grandstream GXP2000 1.2.5.3
Contact: <sip:110@ip-phone:1034;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


10:07:13.380728 IP (tos 0x0, ttl 249, id 3138, offset 0, flags [none], proto UDP (17), length 792)
    ip-phone.activesync > ip-ats.sip: [udp sum ok] UDP, length 764
E....B....-hm.7......
.....+SIP/2.0 200 OK
Via: SIP/2.0/UDP ip-ats;rport;branch=z9hG4bKHN361Xvtpv1ea
From: "Popov2 Alex" <sip:193@ip-ats>;tag=Sr25r27j8Fr0Q
To: <sip:110@ip-phone:1034;transport=udp>;tag=790c6e2be65c976a
Call-ID: b57f6f3a-b41c-1232-d78d-52540046df15
CSeq: 64871751 INVITE
User-Agent: Grandstream GXP2000 1.2.5.3
Contact: <sip:110@ip-phone:1034;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 215

v=0
o=110 8000 8000 IN IP4 ip-phone
s=SIP Call
c=IN IP4 ip-phone
t=0 0
m=audio 5036 RTP/AVP 9 101 13
a=sendrecv
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

10:07:13.382426 IP (tos 0x0, ttl 64, id 23891, offset 0, flags [none], proto UDP (17), length 434)
    ip-ats.sip > ip-phone.activesync: [bad udp cksum e19e!] UDP, length 406
E...]S..@.......m.7....
....ACK sip:110@ip-phone:1034;transport=udp SIP/2.0
Via: SIP/2.0/UDP ip-ats;rport;branch=z9hG4bKmgFH7eF5DQ46c
Max-Forwards: 70
From: "Popov2 Alex" <sip:193@ip-ats>;tag=Sr25r27j8Fr0Q
To: <sip:110@ip-phone:1034;transport=udp>;tag=790c6e2be65c976a
Call-ID: b57f6f3a-b41c-1232-d78d-52540046df15
CSeq: 64871751 ACK
Contact: <sip:mod_sofia@ip-ats:5060>
Content-Length: 0


10:07:19.015775 IP (tos 0x0, ttl 64, id 23892, offset 0, flags [none], proto UDP (17), length 1230)
    ip-ats.sip > ip-phone.activesync: [bad udp cksum bac0!] UDP, length 1202
E...]T..@.......m.7....
....INVITE sip:110@ip-phone:1034;transport=udp SIP/2.0
Via: SIP/2.0/UDP ip-ats;rport;branch=z9hG4bKt665HHmtvcaQj
Max-Forwards: 11
From: "+78123892392" <sip:+78123892392@ip-ats>;tag=vKegyKtXZaUra
To: <sip:110@ip-phone:1034;transport=udp>
Call-ID: ba5d6a90-b41c-1232-d78d-52540046df15
CSeq: 64871755 INVITE
Contact: <sip:mod_sofia@ip-ats:5060>
User-Agent: FreeSWITCH-mod_sofia/1.4.7~32bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 251
X-FS-Support: update_display,send_info
P-Asserted-Identity: "+78123892392" <sip:+78123892392@ip-ats>

v=0
o=FreeSWITCH 1410396159 1410396160 IN IP4 ip-ats
s=FreeSWITCH
c=IN IP4 ip-ats
t=0 0
m=audio 19480 RTP/AVP 8 0 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

10:07:19.271888 IP (tos 0x0, ttl 249, id 3140, offset 0, flags [none], proto UDP (17), length 369)
    ip-phone.activesync > ip-ats.sip: [udp sum ok] UDP, length 341
m.7......./
...].lSIP/2.0 100 Trying
Via: SIP/2.0/UDP ip-ats;rport;branch=z9hG4bKt665HHmtvcaQj
From: "+78123892392" <sip:+78123892392@ip-ats>;tag=vKegyKtXZaUra
To: <sip:110@ip-phone:1034;transport=udp>
Call-ID: ba5d6a90-b41c-1232-d78d-52540046df15
CSeq: 64871755 INVITE
User-Agent: Grandstream GXP2000 1.2.5.3
Content-Length: 0


10:07:19.341487 IP (tos 0x0, ttl 249, id 3141, offset 0, flags [none], proto UDP (17), length 531)
    ip-phone.activesync > ip-ats.sip: [udp sum ok] UDP, length 503
E....E.....jm.7......
.....sSIP/2.0 180 Ringing
Via: SIP/2.0/UDP ip-ats;rport;branch=z9hG4bKt665HHmtvcaQj
From: "+78123892392" <sip:+78123892392@ip-ats>;tag=vKegyKtXZaUra
To: <sip:110@ip-phone:1034;transport=udp>;tag=cff22c053365cf8e
Call-ID: ba5d6a90-b41c-1232-d78d-52540046df15
CSeq: 64871755 INVITE
User-Agent: Grandstream GXP2000 1.2.5.3
Contact: <sip:110@ip-phone:1034;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


10:07:29.014183 IP (tos 0x0, ttl 64, id 23893, offset 0, flags [none], proto UDP (17), length 424)
    ip-ats.sip > ip-phone.activesync: [bad udp cksum ecdc!] UDP, length 396
E...]U..@.......m.7....
....CANCEL sip:110@ip-phone:1034;transport=udp SIP/2.0
Via: SIP/2.0/UDP ip-ats;rport;branch=z9hG4bKt665HHmtvcaQj
Max-Forwards: 11
From: "+78123892392" <sip:+78123892392@ip-ats>;tag=vKegyKtXZaUra
To: <sip:110@ip-phone:1034;transport=udp>
Call-ID: ba5d6a90-b41c-1232-d78d-52540046df15
CSeq: 64871755 CANCEL
Reason: Q.850;cause=19;text="NO_ANSWER"
Content-Length: 0


10:07:29.222407 IP (tos 0x0, ttl 249, id 3142, offset 0, flags [none], proto UDP (17), length 414)
    ip-phone.activesync > ip-ats.sip: [udp sum ok] UDP, length 386
E....F......m.7......
.....:SIP/2.0 200 OK
Via: SIP/2.0/UDP ip-ats;rport;branch=z9hG4bKt665HHmtvcaQj
From: "+78123892392" <sip:+78123892392@ip-ats>;tag=vKegyKtXZaUra
To: <sip:110@ip-phone:1034;transport=udp>;tag=cff22c053365cf8e
Call-ID: ba5d6a90-b41c-1232-d78d-52540046df15
CSeq: 64871755 CANCEL
User-Agent: Grandstream GXP2000 1.2.5.3
Supported: replaces, timer
Content-Length: 0


10:07:29.281850 IP (tos 0x0, ttl 249, id 3143, offset 0, flags [none], proto UDP (17), length 401)
    ip-phone.activesync > ip-ats.sip: [udp sum ok] UDP, length 373
E....G......m.7......
...}..SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP ip-ats;rport;branch=z9hG4bKt665HHmtvcaQj
From: "+78123892392" <sip:+78123892392@ip-ats>;tag=vKegyKtXZaUra
To: <sip:110@ip-phone:1034;transport=udp>;tag=cff22c053365cf8e
Call-ID: ba5d6a90-b41c-1232-d78d-52540046df15
CSeq: 64871755 INVITE
User-Agent: Grandstream GXP2000 1.2.5.3
Content-Length: 0


10:07:29.282079 IP (tos 0x0, ttl 64, id 23895, offset 0, flags [none], proto UDP (17), length 398)
    ip-ats.sip > ip-phone.activesync: [bad udp cksum 2597!] UDP, length 370
E...]W..@.......m.7....
.z..ACK sip:110@ip-phone:1034;transport=udp SIP/2.0
Via: SIP/2.0/UDP ip-ats;rport;branch=z9hG4bKt665HHmtvcaQj
Max-Forwards: 11
From: "+78123892392" <sip:+78123892392@ip-ats>;tag=vKegyKtXZaUra
To: <sip:110@ip-phone:1034;transport=udp>;tag=cff22c053365cf8e
Call-ID: ba5d6a90-b41c-1232-d78d-52540046df15
CSeq: 64871755 ACK
Content-Length: 0


10:07:45.803077 IP (tos 0x0, ttl 249, id 3144, offset 0, flags [none], proto UDP (17), length 558)
    ip-phone.activesync > ip-ats.sip: [udp sum ok] UDP, length 530
E....H.....Lm.7......
....|.BYE sip:mod_sofia@ip-ats:5060 SIP/2.0
Via: SIP/2.0/UDP ip-phone:1034;branch=z9hG4bK7d57c91a2aea90df
From: <sip:110@ip-phone:1034;transport=udp>;tag=790c6e2be65c976a
To: "Popov2 Alex" <sip:193@ip-ats>;tag=Sr25r27j8Fr0Q
Supported: path
Call-ID: b57f6f3a-b41c-1232-d78d-52540046df15
CSeq: 64848 BYE
User-Agent: Grandstream GXP2000 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Reason: SIP ;text="Onhook event"
Content-Length: 0


10:07:45.805595 IP (tos 0x0, ttl 64, id 23901, offset 0, flags [none], proto UDP (17), length 520)
    ip-ats.sip > ip-phone.activesync: [bad udp cksum 14f5!] UDP, length 492
E...]]..@..]....m.7....
...0SIP/2.0 200 OK
Via: SIP/2.0/UDP ip-phone:1034;branch=z9hG4bK7d57c91a2aea90df
From: <sip:110@ip-phone:1034;transport=udp>;tag=790c6e2be65c976a
To: "Popov2 Alex" <sip:193@ip-ats>;tag=Sr25r27j8Fr0Q
Call-ID: b57f6f3a-b41c-1232-d78d-52540046df15
CSeq: 64848 BYE
User-Agent: FreeSWITCH-mod_sofia/1.4.7~32bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Content-Length: 0



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 Заголовок сообщения: Re: call forward
СообщениеДобавлено: 11 сен 2014 10:41 
FreeSWITCH_GuRu

Зарегистрирован: 22 авг 2012 09:52
Сообщения: 1710
что тут можно разобрать?
Код:
tcpdump -nq -s 0 -A -vvv -i eth0 src or dst 109.197.x.x and port 1034 -w /tmp/saas.cap
файл в личку, если чего-то боитесь

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ЛС: @rostel
Сообщество: @ru_freeswitch


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