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Астериск с реальным приоритетом в виртуальной машине

https://forum.asterisk.ru/viewtopic.php?t=2836

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Re: Астериск с реальным приоритетом в виртуальной машине

Добавлено: 16 апр 2013, 10:14
shaverdoff
awsswa писал(а):а "не дадут заказчики" - это по материальным основаниям ?
потому как тут уже был график работы на процессоре "интел атом" - при 15000 разговоров в месяц - себестоимость решения на уровне 7000 рублей

Проблем нет, когда машина не нагружена - мы когда включаем запись со всех камер , все 8 процов стоят со 100% нагрузкой и звук лагает по черному.
запись с камер? видеонаблюдение астериском? кхе.. ну вы сильны я вам скажу.. кроме шуток:)

реально возвращаясь к офтопной теме..
виртуальной машине астериск выделено 2 гига озу и 2 проца. астериск-прокси 4 гига озу и 4 процессора. и вся эта зараза в реальном приоритете работала и падал и квакала. и иакс каналы падали в момент разговоров. и по калфайлам тоже..

Re: Астериск с реальным приоритетом в виртуальной машине

Добавлено: 16 апр 2013, 10:19
shaverdoff
Sfinx писал(а):
shaverdoff писал(а):на физику не получится перетащить... не дадут заказчики
вопрос именно с реальным приоритетом внутри виртуальной среды. он способен помочь или только зря тратит время?
Есть такая хорошая украинская пословица - "до срацi карi очi". Вам верно заметили - пока нагрузки нет, как-то оно еще будет работать. Под ESXi, так понимаю, ведь несколько гостевых OS крутится ? Ставить aster на виртуалку - это сущее нищебродство, обычно в таких случаях никто и ничего не гарантирует.
крутится :) еще как. схд фсшная на ней 3 луна по 30 терабайт каждый и внутрях 300 разнообразных виртуалок.. разбиты 24 ноды в блейд шасси по ресурсным пулам. один пул выделен для астерисков

Есхи решение используется для возможности быстренько развернуть еще парочку, если не хватает ресурсов. для быстрой миграции и бэкапов. я конечно же понимаю что вы все это и так без меня знаете.. (честно говоря для каких целей засунули в эсхи астерики и что преследовали, мои заказчики не отвечают на данный вопрос)))

p/s/ можно смеяться над таким решением.. я уже неделю смеюсь.. но иногда просто в голос плачу..

не дадут перевести на физику - по соображениям что у них уже было решение на физики означенных проблем у них не было. но заказчики захотели другой проект. все объеденить так сказать. торпеду им в шасси ...

Re: Астериск с реальным приоритетом в виртуальной машине

Добавлено: 18 апр 2013, 09:10
shaverdoff
а пишутся разговоры куда? в дефолтовый каталог? 1000 вызовов +кал файлы.. это какой конфы у вас виртуальная машина?

Re: Астериск с реальным приоритетом в виртуальной машине

Добавлено: 24 апр 2013, 19:16
kostoprav
Возможно, не совсем по теме.
Имеется Asterisk 1.8 на свежеустановленном Debian под Vmware. Внутренние звонки работают следующим образом: с ноутбука(софтфон) на смартфон(софтфон) звонок проходит, между двумя смартфонами нет. При ответе на звонок он сразу сбрасывается, в консоли пишет 486 Busy. лог звонка

Код: Выделить всё

<--- SIP read from UDP:10.57.63.35:42484 --->
INVITE sip:103@10.57.63.100 SIP/2.0
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 5711 INVITE
From: "102" <sip:102@10.57.63.100>;tag=2467209167
To: <sip:103@10.57.63.100>
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bKbf722a4848c4b0d98e624397db50ed49353032;rport
Max-Forwards: 70
Contact: "102" <sip:102@10.57.63.35:42484;transport=udp>
Content-Type: application/sdp
Content-Length: 295

v=0
o=- 1366791436982 1366791436983 IN IP4 10.57.63.35
s=-
c=IN IP4 10.57.63.35
t=0 0
m=audio 48766 RTP/AVP 96 97 3 0 8 127
a=rtpmap:96 GSM-EFR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
<------------->
--- (10 headers 13 lines) ---
Sending to 10.57.63.35:42484 (no NAT)
Using INVITE request as basis request - 636a96c4b407799d0333674dd52eb4db@10.57.63.35
Found peer '102' for '102' from 10.57.63.35:42484

<--- Reliably Transmitting (NAT) to 10.57.63.35:42484 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bKbf722a4848c4b0d98e624397db50ed49353032;received=10.57.63.35;rport=42484
From: "102" <sip:102@10.57.63.100>;tag=2467209167
To: <sip:103@10.57.63.100>;tag=as7c5a4d43
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 5711 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4dfa13ff"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '636a96c4b407799d0333674dd52eb4db@10.57.63.35' in 49920 ms (Method: INVITE)

<--- SIP read from UDP:10.57.63.35:42484 --->
ACK sip:103@10.57.63.100 SIP/2.0
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
Max-Forwards: 70
From: "102" <sip:102@10.57.63.100>;tag=2467209167
To: <sip:103@10.57.63.100>;tag=as7c5a4d43
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bKbf722a4848c4b0d98e624397db50ed49353032;rport
CSeq: 5711 ACK
Content-Length: 0

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

<--- SIP read from UDP:10.57.63.35:42484 --->
INVITE sip:103@10.57.63.100:5060 SIP/2.0
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 5712 INVITE
From: "102" <sip:102@10.57.63.100>;tag=2467209167
To: <sip:103@10.57.63.100>
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bK27b1fc75fa93292265a2cdf26d43d8a9353032;rport
Max-Forwards: 70
Contact: "102" <sip:102@10.57.63.35:42484;transport=udp>
Content-Type: application/sdp
Authorization: Digest username="102",realm="asterisk",nonce="4dfa13ff",uri="sip:103@10.57.63.100:5060",response="df0cd5ad3602164b53678304122679aa",algorithm=MD5
Content-Length: 295

v=0
o=- 1366791436982 1366791436983 IN IP4 10.57.63.35
s=-
c=IN IP4 10.57.63.35
t=0 0
m=audio 48766 RTP/AVP 96 97 3 0 8 127
a=rtpmap:96 GSM-EFR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
<------------->
--- (11 headers 13 lines) ---
Sending to 10.57.63.35:42484 (no NAT)
Using INVITE request as basis request - 636a96c4b407799d0333674dd52eb4db@10.57.63.35
Found peer '102' for '102' from 10.57.63.35:42484

<--- Reliably Transmitting (NAT) to 10.57.63.35:42484 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bK27b1fc75fa93292265a2cdf26d43d8a9353032;received=10.57.63.35;rport=42484
From: "102" <sip:102@10.57.63.100>;tag=2467209167
To: <sip:103@10.57.63.100>;tag=as44c3477a
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 5712 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="70416a09"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '636a96c4b407799d0333674dd52eb4db@10.57.63.35' in 49920 ms (Method: INVITE)

<--- SIP read from UDP:10.57.63.35:42484 --->
ACK sip:103@10.57.63.100:5060 SIP/2.0
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
Max-Forwards: 70
From: "102" <sip:102@10.57.63.100>;tag=2467209167
To: <sip:103@10.57.63.100>;tag=as44c3477a
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bK27b1fc75fa93292265a2cdf26d43d8a9353032;rport
CSeq: 5712 ACK
Content-Length: 0

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

<--- SIP read from UDP:10.57.63.35:42484 --->
INVITE sip:103@10.57.63.100:5060 SIP/2.0
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 5713 INVITE
From: "102" <sip:102@10.57.63.100>;tag=2467209167
To: <sip:103@10.57.63.100>
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bK58af0ec097aace64c9c8a1f73e6f8f65353032;rport
Max-Forwards: 70
Contact: "102" <sip:102@10.57.63.35:42484;transport=udp>
Content-Type: application/sdp
Authorization: Digest username="102",realm="asterisk",nonce="70416a09",uri="sip:103@10.57.63.100:5060",response="071f229fd51300135bfde95e37e2b0c5",algorithm=MD5
Content-Length: 295

v=0
o=- 1366791436982 1366791436983 IN IP4 10.57.63.35
s=-
c=IN IP4 10.57.63.35
t=0 0
m=audio 48766 RTP/AVP 96 97 3 0 8 127
a=rtpmap:96 GSM-EFR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
<------------->
--- (11 headers 13 lines) ---
Sending to 10.57.63.35:42484 (NAT)
Using INVITE request as basis request - 636a96c4b407799d0333674dd52eb4db@10.57.63.35
Found peer '102' for '102' from 10.57.63.35:42484
  == Using SIP RTP CoS mark 5
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 127
Found unknown media description format GSM-EFR for ID 96
Found unknown media description format AMR for ID 97
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 127
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.57.63.35:48766
Looking for 103 in internal (domain 10.57.63.100)
list_route: hop: <sip:102@10.57.63.35:42484;transport=udp>

<--- Transmitting (NAT) to 10.57.63.35:42484 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bK58af0ec097aace64c9c8a1f73e6f8f65353032;received=10.57.63.35;rport=42484
From: "102" <sip:102@10.57.63.100>;tag=2467209167
To: <sip:103@10.57.63.100>
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 5713 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:103@10.57.63.100:5060>
Content-Length: 0


<------------>
    -- Executing [103@internal:1] Dial("SIP/102-00000018", "SIP/103,20") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 14328
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.57.63.39:35316:
INVITE sip:103@10.57.63.39:35316;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK7985d98b;rport
Max-Forwards: 70
From: "device" <sip:102@10.57.63.100>;tag=as5baea761
To: <sip:103@10.57.63.39:35316;transport=udp>
Contact: <sip:102@10.57.63.100:5060>
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.3.0
Date: Wed, 24 Apr 2013 06:51:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 1949876031 1949876031 IN IP4 10.57.63.100
s=Asterisk PBX 11.3.0
c=IN IP4 10.57.63.100
t=0 0
m=audio 14328 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/103

<--- SIP read from UDP:10.57.63.39:35316 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK7985d98b;rport=5060;received=10.57.63.100
From: "device" <sip:102@10.57.63.100>;tag=as5baea761
To: <sip:103@10.57.63.39:35316;transport=udp>;tag=1160800380
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
list_route: no route
    -- SIP/103-00000019 is ringing

<--- Transmitting (NAT) to 10.57.63.35:42484 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bK58af0ec097aace64c9c8a1f73e6f8f65353032;received=10.57.63.35;rport=42484
From: "102" <sip:102@10.57.63.100>;tag=2467209167
To: <sip:103@10.57.63.100>;tag=as6219b8f4
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 5713 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:103@10.57.63.100:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:10.57.63.35:42484 --->
OPTIONS sip:10.57.63.100 SIP/2.0
Call-ID: a4179065c6063b246e77ffb84a9aaad5@10.57.63.35
CSeq: 8109 OPTIONS
From: "102" <sip:102@10.57.63.100>;tag=1979661311
To: "102" <sip:102@10.57.63.100>
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bK214ed3479541b288b29e9ff0b1954340353032;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Looking for s in public (domain 10.57.63.100)

<--- Transmitting (no NAT) to 10.57.63.35:42484 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bK214ed3479541b288b29e9ff0b1954340353032;rport;received=10.57.63.35
From: "102" <sip:102@10.57.63.100>;tag=1979661311
To: "102" <sip:102@10.57.63.100>;tag=as18a172cf
Call-ID: a4179065c6063b246e77ffb84a9aaad5@10.57.63.35
CSeq: 8109 OPTIONS
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:10.57.63.100:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a4179065c6063b246e77ffb84a9aaad5@10.57.63.35' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '42c78d2432e113a441cd579d8671b84f@10.57.63.39' Method: OPTIONS

<--- SIP read from UDP:10.57.63.39:35316 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK7985d98b;rport=5060;received=10.57.63.100
From: "device" <sip:102@10.57.63.100>;tag=as5baea761
To: <sip:103@10.57.63.39:35316;transport=udp>;tag=1160800380
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
CSeq: 102 INVITE
Contact: "103" <sip:103@10.57.63.39:35316;transport=udp>
Content-Type: application/sdp
Content-Length: 192

v=0
o=- 317941338504 317941341956 IN IP4 10.57.63.39
s=-
c=IN IP4 10.57.63.39
t=0 0
m=audio 38252 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (9 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.57.63.39:38252
list_route: hop: <sip:103@10.57.63.39:35316;transport=udp>
set_destination: Parsing <sip:103@10.57.63.39:35316;transport=udp> for address/port to send to
set_destination: set destination to 10.57.63.39:35316
Transmitting (NAT) to 10.57.63.39:35316:
ACK sip:103@10.57.63.39:35316;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK12bbd5ec;rport
Max-Forwards: 70
From: "device" <sip:102@10.57.63.100>;tag=as5baea761
To: <sip:103@10.57.63.39:35316;transport=udp>;tag=1160800380
Contact: <sip:102@10.57.63.100:5060>
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.3.0
Content-Length: 0


---

<--- SIP read from UDP:10.57.63.39:35316 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK7985d98b;rport=5060;received=10.57.63.100
From: "device" <sip:102@10.57.63.100>;tag=as5baea761
To: <sip:103@10.57.63.39:35316;transport=udp>;tag=1160800380
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
CSeq: 102 INVITE
Contact: "103" <sip:103@10.57.63.39:35316;transport=udp>
Content-Type: application/sdp
Content-Length: 192

v=0
o=- 317941338504 317941341956 IN IP4 10.57.63.39
s=-
c=IN IP4 10.57.63.39
t=0 0
m=audio 38252 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (9 headers 9 lines) ---
set_destination: Parsing <sip:103@10.57.63.39:35316;transport=udp> for address/port to send to
set_destination: set destination to 10.57.63.39:35316
Transmitting (NAT) to 10.57.63.39:35316:
ACK sip:103@10.57.63.39:35316;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK6c83603c;rport
Max-Forwards: 70
From: "device" <sip:102@10.57.63.100>;tag=as5baea761
To: <sip:103@10.57.63.39:35316;transport=udp>;tag=1160800380
Contact: <sip:102@10.57.63.100:5060>
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.3.0
Content-Length: 0


---
    -- SIP/103-00000019 answered SIP/102-00000018
Audio is at 19108
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 10.57.63.35:42484 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bK58af0ec097aace64c9c8a1f73e6f8f65353032;received=10.57.63.35;rport=42484
From: "102" <sip:102@10.57.63.100>;tag=2467209167
To: <sip:103@10.57.63.100>;tag=as6219b8f4
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 5713 INVITE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:103@10.57.63.100:5060>
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 1707421567 1707421567 IN IP4 10.57.63.100
s=Asterisk PBX 11.3.0
c=IN IP4 10.57.63.100
t=0 0
m=audio 19108 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Remotely bridging SIP/102-00000018 and SIP/103-00000019
set_destination: Parsing <sip:103@10.57.63.39:35316;transport=udp> for address/port to send to
set_destination: set destination to 10.57.63.39:35316
Audio is at 14328
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.57.63.39:35316:
INVITE sip:103@10.57.63.39:35316;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK43131fef;rport
Max-Forwards: 70
From: "device" <sip:102@10.57.63.100>;tag=as5baea761
To: <sip:103@10.57.63.39:35316;transport=udp>;tag=1160800380
Contact: <sip:102@10.57.63.100:5060>
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1949876031 1949876032 IN IP4 10.57.63.35
s=Asterisk PBX 11.3.0
c=IN IP4 10.57.63.35
t=0 0
m=audio 48766 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:10.57.63.35:42484 --->
ACK sip:103@10.57.63.100:5060 SIP/2.0
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 5713 ACK
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bK9962e9d946dcca78ce5456e8ec91e438353032
From: "102" <sip:102@10.57.63.100>;tag=2467209167
To: <sip:103@10.57.63.100>;tag=as6219b8f4
Max-Forwards: 70
Authorization: Digest username="102",realm="asterisk",nonce="70416a09",uri="sip:103@10.57.63.100:5060",response="071f229fd51300135bfde95e37e2b0c5",algorithm=MD5
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:102@10.57.63.35:42484;transport=udp> for address/port to send to
set_destination: set destination to 10.57.63.35:42484
Audio is at 19108
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.57.63.35:42484:
INVITE sip:102@10.57.63.35:42484;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK330680b3;rport
Max-Forwards: 70
From: <sip:103@10.57.63.100>;tag=as6219b8f4
To: "102" <sip:102@10.57.63.100>;tag=2467209167
Contact: <sip:103@10.57.63.100:5060>
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1707421567 1707421568 IN IP4 10.57.63.39
s=Asterisk PBX 11.3.0
c=IN IP4 10.57.63.39
t=0 0
m=audio 38252 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:10.57.63.35:42484 --->
OPTIONS sip:103@10.57.63.100 SIP/2.0
Call-ID: 76a00198f14093d609caf84ace789091@10.57.63.35
CSeq: 5300 OPTIONS
From: "102" <sip:102@10.57.63.100>;tag=2692549839
To: <sip:103@10.57.63.100>
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bK0fc6516971475e9e3ad7ac6610feb48c353032;rport
Max-Forwards: 70
Contact: "102" <sip:102@10.57.63.35:42484;transport=udp>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Looking for 103 in public (domain 10.57.63.100)

<--- Transmitting (no NAT) to 10.57.63.35:42484 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.57.63.35:42484;branch=z9hG4bK0fc6516971475e9e3ad7ac6610feb48c353032;rport;received=10.57.63.35
From: "102" <sip:102@10.57.63.100>;tag=2692549839
To: <sip:103@10.57.63.100>;tag=as4a10b083
Call-ID: 76a00198f14093d609caf84ace789091@10.57.63.35
CSeq: 5300 OPTIONS
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '76a00198f14093d609caf84ace789091@10.57.63.35' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:10.57.63.39:35316 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK43131fef;rport=5060;received=10.57.63.100
From: "device" <sip:102@10.57.63.100>;tag=as5baea761
To: <sip:103@10.57.63.39:35316;transport=udp>;tag=1160800380
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
CSeq: 103 INVITE
Contact: "103" <sip:103@10.57.63.39:35316;transport=udp>
Content-Type: application/sdp
Content-Length: 192

v=0
o=- 317941338504 317941342849 IN IP4 10.57.63.39
s=-
c=IN IP4 10.57.63.39
t=0 0
m=audio 38252 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (9 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.57.63.39:38252
set_destination: Parsing <sip:103@10.57.63.39:35316;transport=udp> for address/port to send to
set_destination: set destination to 10.57.63.39:35316
Transmitting (NAT) to 10.57.63.39:35316:
ACK sip:103@10.57.63.39:35316;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK7ced693c;rport
Max-Forwards: 70
From: "device" <sip:102@10.57.63.100>;tag=as5baea761
To: <sip:103@10.57.63.39:35316;transport=udp>;tag=1160800380
Contact: <sip:102@10.57.63.100:5060>
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.3.0
Content-Length: 0


---

<--- SIP read from UDP:10.57.63.35:42484 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK330680b3;rport=5060;received=10.57.63.100
From: <sip:103@10.57.63.100>;tag=as6219b8f4
To: "102" <sip:102@10.57.63.100>
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:10.57.63.39:35316 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK43131fef;rport=5060;received=10.57.63.100
From: "device" <sip:102@10.57.63.100>;tag=as5baea761
To: <sip:103@10.57.63.39:35316;transport=udp>;tag=1160800380
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
CSeq: 103 INVITE
Contact: "103" <sip:103@10.57.63.39:35316;transport=udp>
Content-Type: application/sdp
Content-Length: 192

v=0
o=- 317941338504 317941342849 IN IP4 10.57.63.39
s=-
c=IN IP4 10.57.63.39
t=0 0
m=audio 38252 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (9 headers 9 lines) ---
set_destination: Parsing <sip:103@10.57.63.39:35316;transport=udp> for address/port to send to
set_destination: set destination to 10.57.63.39:35316
Transmitting (NAT) to 10.57.63.39:35316:
ACK sip:103@10.57.63.39:35316;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK4d38a265;rport
Max-Forwards: 70
From: "device" <sip:102@10.57.63.100>;tag=as5baea761
To: <sip:103@10.57.63.39:35316;transport=udp>;tag=1160800380
Contact: <sip:102@10.57.63.100:5060>
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.3.0
Content-Length: 0


---
Retransmitting #1 (NAT) to 10.57.63.35:42484:
INVITE sip:102@10.57.63.35:42484;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK330680b3;rport
Max-Forwards: 70
From: <sip:103@10.57.63.100>;tag=as6219b8f4
To: "102" <sip:102@10.57.63.100>;tag=2467209167
Contact: <sip:103@10.57.63.100:5060>
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1707421567 1707421568 IN IP4 10.57.63.39
s=Asterisk PBX 11.3.0
c=IN IP4 10.57.63.39
t=0 0
m=audio 38252 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:10.57.63.35:42484 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK330680b3;rport=5060;received=10.57.63.100
From: <sip:103@10.57.63.100>;tag=as6219b8f4
To: "102" <sip:102@10.57.63.100>
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '2e68fc628731b8ea80bd26d7e595d35f@10.57.63.39' Method: OPTIONS
Reliably Transmitting (NAT) to 10.57.63.34:20544:
OPTIONS sip:101@10.57.63.34:20544;rinstance=d5557e92a597c323 SIP/2.0
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK249da631;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.57.63.100>;tag=as252681df
To: <sip:101@10.57.63.34:20544;rinstance=d5557e92a597c323>
Contact: <sip:asterisk@10.57.63.100:5060>
Call-ID: 4c76a45f718a17173e7ee1d3197ce7af@10.57.63.100:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.3.0
Date: Wed, 24 Apr 2013 06:51:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.57.63.34:20544 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK249da631;rport=5060
Contact: <sip:10.57.63.34:20544>
To: <sip:101@10.57.63.34:20544;rinstance=d5557e92a597c323>;tag=8b475f6a
From: "asterisk"<sip:asterisk@10.57.63.100>;tag=as252681df
Call-ID: 4c76a45f718a17173e7ee1d3197ce7af@10.57.63.100:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1102u stamp 52345
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '4c76a45f718a17173e7ee1d3197ce7af@10.57.63.100:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.57.63.10:36654:
OPTIONS sip:100@10.57.63.10:36654;rinstance=39cfcbf698d712a0 SIP/2.0
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK1d2fc61d;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.57.63.100>;tag=as2b4b4426
To: <sip:100@10.57.63.10:36654;rinstance=39cfcbf698d712a0>
Contact: <sip:asterisk@10.57.63.100:5060>
Call-ID: 67205d1625e9574b09fad04362f79230@10.57.63.100:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.3.0
Date: Wed, 24 Apr 2013 06:51:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.57.63.10:36654 --->


<------------->

<--- SIP read from UDP:10.57.63.10:36654 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK1d2fc61d;rport=5060
Contact: <sip:10.57.63.10:36654>
To: <sip:100@10.57.63.10:36654;rinstance=39cfcbf698d712a0>;tag=0a218513
From: "asterisk"<sip:asterisk@10.57.63.100>;tag=as2b4b4426
Call-ID: 67205d1625e9574b09fad04362f79230@10.57.63.100:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1102u stamp 52345
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '67205d1625e9574b09fad04362f79230@10.57.63.100:5060' Method: OPTIONS
Retransmitting #2 (NAT) to 10.57.63.35:42484:
INVITE sip:102@10.57.63.35:42484;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK330680b3;rport
Max-Forwards: 70
From: <sip:103@10.57.63.100>;tag=as6219b8f4
To: "102" <sip:102@10.57.63.100>;tag=2467209167
Contact: <sip:103@10.57.63.100:5060>
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1707421567 1707421568 IN IP4 10.57.63.39
s=Asterisk PBX 11.3.0
c=IN IP4 10.57.63.39
t=0 0
m=audio 38252 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:10.57.63.35:42484 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK330680b3;rport=5060;received=10.57.63.100
From: <sip:103@10.57.63.100>;tag=as6219b8f4
To: "102" <sip:102@10.57.63.100>
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Retransmitting #3 (NAT) to 10.57.63.35:42484:
INVITE sip:102@10.57.63.35:42484;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK330680b3;rport
Max-Forwards: 70
From: <sip:103@10.57.63.100>;tag=as6219b8f4
To: "102" <sip:102@10.57.63.100>;tag=2467209167
Contact: <sip:103@10.57.63.100:5060>
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1707421567 1707421568 IN IP4 10.57.63.39
s=Asterisk PBX 11.3.0
c=IN IP4 10.57.63.39
t=0 0
m=audio 38252 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:10.57.63.35:42484 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.57.63.100:5060;branch=z9hG4bK330680b3;rport=5060;received=10.57.63.100
From: <sip:103@10.57.63.100>;tag=as6219b8f4
To: "102" <sip:102@10.57.63.100>
Call-ID: 636a96c4b407799d0333674dd52eb4db@10.57.63.35
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:10.57.63.39:35316 --->
BYE sip:102@10.57.63.100:5060 SIP/2.0
Via: SIP/2.0/UDP 10.57.63.39:35316;branch=z9hG4bK198d987f41e615486a8d5a9692588012363838
CSeq: 1 BYE
From: <sip:103@10.57.63.39:35316;transport=udp>;tag=1160800380
To: "device" <sip:102@10.57.63.100>;tag=as5baea761
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Sending to 10.57.63.39:35316 (NAT)
Scheduling destruction of SIP dialog '4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060' in 23488 ms (Method: BYE)

<--- Transmitting (NAT) to 10.57.63.39:35316 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.57.63.39:35316;branch=z9hG4bK198d987f41e615486a8d5a9692588012363838;received=10.57.63.39;rport=35316
From: <sip:103@10.57.63.39:35316;transport=udp>;tag=1160800380
To: "device" <sip:102@10.57.63.100>;tag=as5baea761
Call-ID: 4f9b7b8f37e3bebc74b483842bfc0ae0@10.57.63.100:5060
CSeq: 1 BYE
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (internal, 103, 1) exited non-zero on 'SIP/102-00000018'
Scheduling destruction of SIP dialog '636a96c4b407799d0333674dd52eb4db@10.57.63.35' in 49920 ms (Method: ACK)

<--- SIP read from UDP:10.57.63.34:20544 --->
Ставлю Elastix проблема исчезает.

Re: Астериск с реальным приоритетом в виртуальной машине

Добавлено: 25 апр 2013, 13:08
SolarW
shaverdoff писал(а):и вся эта зараза в реальном приоритете работала и падал и квакала. и иакс каналы падали в момент разговоров. и по калфайлам тоже..
Может внешний источник синхронизации попробовать?
Прокинуть средствами ESXi виртуалке вот такой UBS-свисток

Изображение

поставить dahdi с его поддержкой...
Вроде недорого такое решение стоит.

Когда лопатил вопрос виртуализации астериска под OpenVZ попалось на каком-то англоязычном форуме сообщение от гражданина пишущего что вставил в ноду такой свисток и проблем с полутора десятками виртуальных машин с астерисками как не бывало...

Re: Астериск с реальным приоритетом в виртуальной машине

Добавлено: 25 апр 2013, 14:07
SolarW
switch писал(а):Если честно, то я не понимаю как оно работает
Магия? :-)
Судя по всему стоит устройство недорого (навскидку увидел цену 65$ в каком-то импортном интернет-магазине), заинтересованные лица могут проэкспериментировать и нам тут рассказать что получилось.

Re: Астериск с реальным приоритетом в виртуальной машине

Добавлено: 25 апр 2013, 14:17
SolarW
http://igorg.ru/2011/04/04/sangoma-usb-voicetime/ - Ага, вот эта статья.

P.S. Жалко что Игорь с работой/семейными заботами забил на свой блог (больше года ни одной записи), было очень интересно его читать.
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