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GrandStream GXP1160

https://forum.asterisk.ru/viewtopic.php?t=4612

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GrandStream GXP1160

Добавлено: 08 авг 2014, 08:35
johny87
Есть проблема с данным аппаратом, заметил что во время долгого разговора начинается голос собеседника слышиться как робот, при этом тебя слышат отлично. Но если переключить допустим на другой аппарат, то слышимость того же звонка становится опять нормальной. Происходит данная фигня регулярно и рандомно, чаще на долгих разговорах. Захватить можно, но из-за рандомности ситуации тяжело, да и анализировать я особо не умею через wireshark.

Может по опыту подскажете от каких настроек телефона может зависеть ?

Re: GrandStream GXP1160

Добавлено: 08 авг 2014, 09:12
awsswa
не от каких - если конечно у вас CNG и VAD не включен
все настройки с картинками можете посмотреть тут http://awsswa.livejournal.com/23840.html

Re: GrandStream GXP1160

Добавлено: 12 авг 2014, 10:15
johny87
Теперь я точно понял что это телефон.
Сегодня звонил и опять начался робот, захват не успел поставить, т.к. разговор важный и не до этого. Но дело в том, что разговор записывался asteriskом и на записи опять все чисто, голос собеседника отлично слышно, а в телефоне был робот..

VAD выключен, CNG не вижу в настройках.

Re: GrandStream GXP1160

Добавлено: 12 авг 2014, 12:19
SolarW
В одном из последних реализованных проектов применено более сотни таких телефонов.
За два месяца эксплуатации указанная проблема не зафиксирована.

Re: GrandStream GXP1160

Добавлено: 12 авг 2014, 12:24
Vlad1983
допустим GXP1160 висит на номере 100

Код: Выделить всё

CLI> sip set debug peer 100
звоним на 100
выхоп под spoiler

Re: GrandStream GXP1160

Добавлено: 12 авг 2014, 15:06
johny87
Там ничего интересного нет, все в норме. Если и смотреть то RTP трафик, зачем SIP ?
Проанализировал встроенным в wireshark анализатором трафик когда слышен робот - ничего, все красиво, потери внутри сети 0%, в интернете 0%.
Очень редко в RTP потоках встречается Incorrect Timestamp..

Re: GrandStream GXP1160

Добавлено: 12 авг 2014, 15:11
Vlad1983
ok
разбирайтесь сами

Re: GrandStream GXP1160

Добавлено: 12 авг 2014, 15:49
johny87
В любом случае разбираться мне. Вот не поленился и сделал дебаг sip :
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
<--- SIP read from UDP:192.168.100.10:5060 --->
INVITE sip:88003001800@192.168.100.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bK1999480653;rport
From: <sip:105@192.168.100.1>;tag=518780165
To: <sip:88003001800@192.168.100.1>
Call-ID: 1534371858-5060-8@BJC.BGI.BAA.BA
CSeq: 70 INVITE
Contact: <sip:105@192.168.100.10:5060>
Max-Forwards: 70
User-Agent: Grandstream GXP1160 1.0.6.11
Privacy: none
P-Preferred-Identity: <sip:105@192.168.100.1>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 405

v=0
o=105 8000 8000 IN IP4 192.168.100.10
s=SIP Call
c=IN IP4 192.168.100.10
t=0 0
m=audio 5004 RTP/AVP 8 0 4 18 9 97 2 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:10
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (16 headers 19 lines) ---
Sending to 192.168.100.10:5060 (NAT)
Using INVITE request as basis request - 1534371858-5060-8@BJC.BGI.BAA.BA
Found peer '105' for '105' from 192.168.100.10:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format iLBC for ID 97
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Capabilities: us - 0x110f (g723|gsm|ulaw|alaw|g729|g722), peer - audio=0x1d0d (g723|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x110d (g723|ulaw|alaw|g729|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.100.10:5004
Looking for 88003001800 in 105 (domain 192.168.100.1)
list_route: hop: <sip:105@192.168.100.10:5060>

<--- Transmitting (NAT) to 192.168.100.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bK1999480653;received=192.168.100.10;rport=5060
From: <sip:105@192.168.100.1>;tag=518780165
To: <sip:88003001800@192.168.100.1>
Call-ID: 1534371858-5060-8@BJC.BGI.BAA.BA
CSeq: 70 INVITE
Server: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:88003001800@192.168.100.1:5060>
Content-Length: 0


<------------>
-- Executing [88003001800@105:1] Set("SIP/105-00000e3c", "fname=1407842734.4114.wav") in new stack
-- Executing [88003001800@105:2] MixMonitor("SIP/105-00000e3c", "/home/voip/records/1407842734.4114.wav,W(0)b") in new stack
-- Executing [88003001800@105:3] Goto("SIP/105-00000e3c", "allow,88003001800,1") in new stack
-- Goto (allow,88003001800,1)
== Begin MixMonitor Recording SIP/105-00000e3c
-- Executing [88003001800@allow:1] Dial("SIP/105-00000e3c", "SIP/neofon/8003001800") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/neofon/8003001800
-- SIP/neofon-00000e3d is making progress passing it to SIP/105-00000e3c
Audio is at 14716
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 192.168.100.10:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bK1999480653;received=192.168.100.10;rport=5060
From: <sip:105@192.168.100.1>;tag=518780165
To: <sip:88003001800@192.168.100.1>;tag=as0336c01e
Call-ID: 1534371858-5060-8@BJC.BGI.BAA.BA
CSeq: 70 INVITE
Server: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:88003001800@192.168.100.1:5060>
Content-Type: application/sdp
Content-Length: 392

v=0
o=root 1444376752 1444376752 IN IP4 192.168.100.1
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.100.1
t=0 0
m=audio 14716 RTP/AVP 8 0 9 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
-- SIP/neofon-00000e3d answered SIP/105-00000e3c
Audio is at 14716
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bK1999480653;received=192.168.100.10;rport=5060
From: <sip:105@192.168.100.1>;tag=518780165
To: <sip:88003001800@192.168.100.1>;tag=as0336c01e
Call-ID: 1534371858-5060-8@BJC.BGI.BAA.BA
CSeq: 70 INVITE
Server: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:88003001800@192.168.100.1:5060>
ontent-Type: application/sdp
Content-Length: 392

v=0
o=root 1444376752 1444376753 IN IP4 192.168.100.1
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.100.1
t=0 0
m=audio 14716 RTP/AVP 8 0 9 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
-- fixed jitterbuffer created on channel SIP/neofon-00000e3d
-- fixed jitterbuffer created on channel SIP/105-00000e3c

<--- SIP read from UDP:192.168.100.10:5060 --->
ACK sip:88003001800@192.168.100.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bK504414255;rport
From: <sip:105@192.168.100.1>;tag=518780165
To: <sip:88003001800@192.168.100.1>;tag=as0336c01e
Call-ID: 1534371858-5060-8@BJC.BGI.BAA.BA
CSeq: 70 ACK
Contact: <sip:105@192.168.100.10:5060>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK0cae1355;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as7e809be8
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 12d16c3271344bfe239d7b1c69aaab1c@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:26:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK0cae1355;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as7e809be8
To: <sip:105@192.168.100.10:5060>;tag=233480971
Call-ID: 12d16c3271344bfe239d7b1c69aaab1c@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '12d16c3271344bfe239d7b1c69aaab1c@192.168.100.1:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK32c0c7a4;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as08cfee67
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 27d79f3079a8e2e2266c213663f145af@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:27:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK32c0c7a4;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as08cfee67
To: <sip:105@192.168.100.10:5060>;tag=356475194
Call-ID: 27d79f3079a8e2e2266c213663f145af@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '27d79f3079a8e2e2266c213663f145af@192.168.100.1:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK05cce805;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as738741f8
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 7e650a973685738a06e0449c3093fd90@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:28:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK05cce805;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as738741f8
To: <sip:105@192.168.100.10:5060>;tag=1569310626
Call-ID: 7e650a973685738a06e0449c3093fd90@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7e650a973685738a06e0449c3093fd90@192.168.100.1:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK10cf67e2;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as19a0a1a3
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 7ffee3a6646de71e116a257876f9dc92@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:29:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK10cf67e2;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as19a0a1a3
To: <sip:105@192.168.100.10:5060>;tag=662078373
Call-ID: 7ffee3a6646de71e116a257876f9dc92@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7ffee3a6646de71e116a257876f9dc92@192.168.100.1:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK5e0006a5;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as7e89e766
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 3cab83497438e6a71d47780b783d2c7d@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:30:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK5e0006a5;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as7e89e766
To: <sip:105@192.168.100.10:5060>;tag=86485106
Call-ID: 3cab83497438e6a71d47780b783d2c7d@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3cab83497438e6a71d47780b783d2c7d@192.168.100.1:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK6f4ad9f1;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as37f8d8ed
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 1c5b7db20bc2a5f93643fdd05019995d@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:31:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK6f4ad9f1;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as37f8d8ed
To: <sip:105@192.168.100.10:5060>;tag=1659936418
Call-ID: 1c5b7db20bc2a5f93643fdd05019995d@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1c5b7db20bc2a5f93643fdd05019995d@192.168.100.1:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK298f0d12;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as01a83e7c
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 24b92fc7346c8df85f458d8613444486@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:32:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK298f0d12;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as01a83e7c
To: <sip:105@192.168.100.10:5060>;tag=1712162506
Call-ID: 24b92fc7346c8df85f458d8613444486@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '24b92fc7346c8df85f458d8613444486@192.168.100.1:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK774ee848;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as118cf097
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 2608b1d2578b9f6c70523b555871fc33@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:33:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK774ee848;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as118cf097
To: <sip:105@192.168.100.10:5060>;tag=1252417995
Call-ID: 2608b1d2578b9f6c70523b555871fc33@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK1bb7f1c1;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as49844cac
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 2964853c26a8ec5747d1d5ab4e4602af@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:34:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK1bb7f1c1;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as49844cac
To: <sip:105@192.168.100.10:5060>;tag=1014057279
Call-ID: 2964853c26a8ec5747d1d5ab4e4602af@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2964853c26a8ec5747d1d5ab4e4602af@192.168.100.1:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK79ab7208;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as0982c784
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 2911b02e64044be06bb9bcfa13d19b8a@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:35:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK79ab7208;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as0982c784
To: <sip:105@192.168.100.10:5060>;tag=1327065415
Call-ID: 2911b02e64044be06bb9bcfa13d19b8a@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2911b02e64044be06bb9bcfa13d19b8a@192.168.100.1:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK4a45ae59;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as222a5de2
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 539ca53e576b4eea791fb7bd2302dd6e@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:36:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK4a45ae59;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as222a5de2
To: <sip:105@192.168.100.10:5060>;tag=1870792876
Call-ID: 539ca53e576b4eea791fb7bd2302dd6e@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '539ca53e576b4eea791fb7bd2302dd6e@192.168.100.1:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK676aae01;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as4b111948
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 0692d9134fe5187271b2deed4420f087@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:37:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK676aae01;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as4b111948
To: <sip:105@192.168.100.10:5060>;tag=1248307314
Call-ID: 0692d9134fe5187271b2deed4420f087@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0692d9134fe5187271b2deed4420f087@192.168.100.1:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK4756e386;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as23391db5
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 0d276c2b6cbe87863a224d5374b4c798@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:38:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK4756e386;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as23391db5
To: <sip:105@192.168.100.10:5060>;tag=184416271
Call-ID: 0d276c2b6cbe87863a224d5374b4c798@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0d276c2b6cbe87863a224d5374b4c798@192.168.100.1:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK312351c2;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as0b7e0312
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 2f56dc153cf0e75271a5c6401308aed3@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:39:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK312351c2;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as0b7e0312
To: <sip:105@192.168.100.10:5060>;tag=230858597
Call-ID: 2f56dc153cf0e75271a5c6401308aed3@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2f56dc153cf0e75271a5c6401308aed3@192.168.100.1:5060' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK6be4ae91;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as537144a6
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 4c7dda566dc5b92268d61dba28caf7f8@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:40:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK6be4ae91;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as537144a6
To: <sip:105@192.168.100.10:5060>;tag=1012608474
Call-ID: 4c7dda566dc5b92268d61dba28caf7f8@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4c7dda566dc5b92268d61dba28caf7f8@192.168.100.1:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.100.10:5060 --->
BYE sip:88003001800@192.168.100.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bK1303486562;rport
From: <sip:105@192.168.100.1>;tag=518780165
To: <sip:88003001800@192.168.100.1>;tag=as0336c01e
Call-ID: 1534371858-5060-8@BJC.BGI.BAA.BA
CSeq: 71 BYE
Contact: <sip:105@192.168.100.10:5060>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.100.10:5060 (NAT)
Scheduling destruction of SIP dialog '1534371858-5060-8@BJC.BGI.BAA.BA' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bK1303486562;received=192.168.100.10;rport=5060
From: <sip:105@192.168.100.1>;tag=518780165
To: <sip:88003001800@192.168.100.1>;tag=as0336c01e
Call-ID: 1534371858-5060-8@BJC.BGI.BAA.BA
CSeq: 71 BYE
Server: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
-- fixed jitterbuffer destroyed on channel SIP/neofon-00000e3d
== Spawn extension (allow, 88003001800, 1) exited non-zero on 'SIP/105-00000e3c'
-- fixed jitterbuffer destroyed on channel SIP/105-00000e3c
== MixMonitor close filestream
== End MixMonitor Recording SIP/105-00000e3c
Reliably Transmitting (NAT) to 192.168.100.10:5060:
OPTIONS sip:105@192.168.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK362b49ff;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as18488c7f
To: <sip:105@192.168.100.10:5060>
Contact: <sip:asterisk@192.168.100.1:5060>
Call-ID: 58de775931df70dd14b448fd07a534fc@192.168.100.1:5060
CSeq: 102 OPTIONS
User-Agent: asterisk
Date: Tue, 12 Aug 2014 11:41:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK362b49ff;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as18488c7f
To: <sip:105@192.168.100.10:5060>;tag=674056550
Call-ID: 58de775931df70dd14b448fd07a534fc@192.168.100.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1160 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '58de775931df70dd14b448fd07a534fc@192.168.100.1:5060' Method: OPTIONS
Really destroying SIP dialog '1534371858-5060-8@BJC.BGI.BAA.BA' Method: BYE

Re: GrandStream GXP1160

Добавлено: 12 авг 2014, 16:05
johny87
Такой же робот бывает и через iax2, я так понял от канала вообще не зависит. Рандомно появляется и до переадресации или окончания звонка не исчезает.

Re: GrandStream GXP1160

Добавлено: 12 авг 2014, 16:08
Vlad1983
читаем ещё раз как нужно было снять дебаг
Часовой пояс: UTC+03:00
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