VIDEOCHAT  ::   FAQ  ::   Поиск  ::   Регистрация  ::   Вход

sip-tcp транк между Avaya 8800 и Астериском

Проблемы Asterisk без вэб-оболочек и их решения

Модераторы: april22, Zavr2008

Ответить
Иван89
Сообщения: 11
Зарегистрирован: 02 апр 2013, 16:22

sip-tcp транк между Avaya 8800 и Астериском

Сообщение Иван89 »

Здравствуйте господа специалисты! Подскажите пожалуйста в чем проблема.
На станции Авая обнаружил один свободный sip trunk : display capacity - SIP Trunks (included in 'Trunk ports'): 1

Пытаюсь тестово-подключить Астериск, но не получается. Вот конфиг.

[root]# cat /etc/asterisk/sip.conf
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
tcpenable=yes

[authentication]

[6000]
type=friend
…...........
…............

[avaya]
type=peer
host=192.168.0.1XX — ip-адрес AVAYA
dtmfmode=rfc2833
transport=tcp
context=avaya
disallow=all
allow=ulaw

[root]# cat /etc/asterisk/extensions.conf
[globals]

[general]
autofallthrough=yes

[default]
…..........
…..........
[avaya]
exten => _33XX,1,Dial(SIP/${EXTEN}@avaya,20,)

Подскажите пожалуйста в чем может быть проблема!

На сервере Астериск два сетевых интерфейса :
eth0 192.168.0.1YY – для Авая с ip-адресом 192.168.0.1XX
eth1 192.168.102.4 — к работающим SIP-телефонам.

ast-real*CLI> sip set debug on -------При звонке на номер Авая 3311

SIP Debugging enabled

<--- SIP read from UDP:192.168.4.3:11966 --->
INVITE sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-7c004c6c7989ac39-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:6004@192.168.4.3:11966>
To: <sip:3311@192.168.102.4>
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5 stamp 69607
Content-Length: 243

v=0
o=- 13009378186140040 1 IN IP4 192.168.2.22
s=X-Lite 4 release 4.5 stamp 69607
c=IN IP4 192.168.2.22
t=0 0
m=audio 55508 RTP/AVP 9 0 8 100 101
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 10 lines) ---
Sending to 192.168.4.3:11966 (NAT)
Using INVITE request as basis request - ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
Found peer '6004' for '6004' from 192.168.4.3:11966

<--- Reliably Transmitting (NAT) to 192.168.4.3:11966 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-7c004c6c7989ac39-1---d8754z-;received=192.168.4.3;rport=11966
From: <sip:6004@192.168.102.4>;tag=f4838ed2
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4788ee86"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.4.3:11966 --->
ACK sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-7c004c6c7989ac39-1---d8754z-;rport
Max-Forwards: 70
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.4.3:11966 --->
INVITE sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:6004@192.168.4.3:11966>
To: <sip:3311@192.168.102.4>
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5 stamp 69607
Authorization: Digest username="6004",realm="asterisk",nonce="4788ee86",uri="sip:3311@192.168.102.4",response="bca1a172f10773e6211c115b44a98ffc",algorithm=MD5
Content-Length: 243

v=0
o=- 13009378186140040 1 IN IP4 192.168.2.22
s=X-Lite 4 release 4.5 stamp 69607
c=IN IP4 192.168.2.22
t=0 0
m=audio 55508 RTP/AVP 9 0 8 100 101
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 10 lines) ---
Sending to 192.168.4.3:11966 (NAT)
Using INVITE request as basis request - ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
Found peer '6004' for '6004' from 192.168.4.3:11966
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format speex for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x20000100c (ulaw|alaw|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.22:55508
Looking for 3311 in demo1 (domain 192.168.102.4)

<--- Reliably Transmitting (NAT) to 192.168.4.3:11966 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;received=192.168.4.3;rport=11966
From: <sip:6004@192.168.102.4>;tag=f4838ed2
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU' in 6400 ms (Method: INVITE)
Retransmitting #1 (NAT) to 192.168.4.3:11966:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;received=192.168.4.3;rport=11966
From: <sip:6004@192.168.102.4>;tag=f4838ed2
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.4.3:11966 --->
ACK sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;rport
Max-Forwards: 70
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.4.3:11966 --->
ACK sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:11966;branch=z9hG4bK-d8754z-aa365141c465cea2-1---d8754z-;rport
Max-Forwards: 70
To: <sip:3311@192.168.102.4>;tag=as6e2c9d95
From: <sip:6004@192.168.102.4>;tag=f4838ed2
Call-ID: ZWMxN2ZiMDUwZWIwZWVkYTdkZjAyOWYzNWEwMzMxNWU
CSeq: 2 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.4.3:11966 --->


<------------->
Reliably Transmitting (NAT) to 192.168.4.3:11966:
OPTIONS sip:6004@192.168.4.3:11966;rinstance=9a4e689abdd52f2d SIP/2.0
Via: SIP/2.0/UDP 192.168.102.4:5060;branch=z9hG4bK360ab8a2;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.102.4>;tag=as4e846880
To: <sip:6004@192.168.4.3:11966;rinstance=9a4e689abdd52f2d>
Contact: <sip:asterisk@192.168.102.4:5060>
Call-ID: 19ec299a1df4913213651aa878c8ac05@192.168.102.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert1
Date: Tue, 02 Apr 2013 12:05:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.4.3:11966 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.102.4:5060;branch=z9hG4bK360ab8a2;rport=5060
Contact: <sip:192.168.4.3:11966>
To: <sip:6004@192.168.4.3:11966;rinstance=9a4e689abdd52f2d>;tag=4a25b087
From: "asterisk"<sip:asterisk@192.168.102.4>;tag=as4e846880
Call-ID: 19ec299a1df4913213651aa878c8ac05@192.168.102.4:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite release 4.5 stamp 69607
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '19ec299a1df4913213651aa878c8ac05@192.168.102.4:5060' Method: OPTIONS
ast-real*CLI>
Вернуться к началу
Аватара пользователя
zzuz
Сообщения: 1658
Зарегистрирован: 21 сен 2010, 13:33
Контактная информация:
Контактная информация пользователя zzuz

Re: sip-tcp транк между Avaya 8800 и Астериском

Сообщение zzuz »

Все пакеты по UDP . Настраиваете дальше.
Линия24 - Системы Массового Телефонного Обслуживания
Вернуться к началу
Иван89
Сообщения: 11
Зарегистрирован: 02 апр 2013, 16:22

Re: sip-tcp транк между Avaya 8800 и Астериском

Сообщение Иван89 »

И как быть? Подскажите пожалуйста?

--------------------------------------------------------------------------------------------------------------------
Really destroying SIP dialog '3e190ecd4f3d95e413a04499232279aa@192.168.0.127:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.4.3:35534 --->


<------------->
Reliably Transmitting (NAT) to 192.168.4.3:35534:
OPTIONS sip:6004@192.168.4.3:35534;rinstance=d1f03e58caecdcf1 SIP/2.0
Via: SIP/2.0/UDP 192.168.102.4:5060;branch=z9hG4bK7d1095fd;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.102.4>;tag=as04f54006
To: <sip:6004@192.168.4.3:35534;rinstance=d1f03e58caecdcf1>
Contact: <sip:asterisk@192.168.102.4:5060>
Call-ID: 45ffc32044bb40db362275ad20029773@192.168.102.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert1
Date: Wed, 03 Apr 2013 10:51:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.4.3:35534 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.102.4:5060;branch=z9hG4bK7d1095fd;rport=5060
Contact: <sip:192.168.4.3:35534>
To: <sip:6004@192.168.4.3:35534;rinstance=d1f03e58caecdcf1>;tag=94430dfd
From: "asterisk"<sip:asterisk@192.168.102.4>;tag=as04f54006
Call-ID: 45ffc32044bb40db362275ad20029773@192.168.102.4:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite release 4.5 stamp 69607
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '45ffc32044bb40db362275ad20029773@192.168.102.4:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.0.101:5060:
OPTIONS sip:192.168.0.101 SIP/2.0
Via: SIP/2.0/TCP 192.168.0.127:5060;branch=z9hG4bK20075636;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.127>;tag=as580015cd
To: <sip:192.168.0.101>
Contact: <sip:asterisk@192.168.0.127:5060;transport=TCP>
Call-ID: 37fcd48112142ebe5cd0cf676085a727@192.168.0.127:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert1
Date: Wed, 03 Apr 2013 10:51:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
Вернуться к началу
ded
Сообщения: 15619
Зарегистрирован: 26 авг 2010, 19:00

Re: sip-tcp транк между Avaya 8800 и Астериском

Сообщение ded »

С чем быть?
Вы привели сейчас два пакета OPTIONS по UDP, с ними как-то надо быть? Или казаться?
(с) Шекспир. Гамлет.

ВАМ НАДО ИСКАТЬ по ключевым словам Asterisk sip TCP.
Ибо инструкций как настроить по ТСР написано уже тонна.
Вернуться к началу
Иван89
Сообщения: 11
Зарегистрирован: 02 апр 2013, 16:22

Re: sip-tcp транк между Avaya 8800 и Астериском

Сообщение Иван89 »

Изменил на:
[root ~]# vi /etc/asterisk/sip.conf
[general]
tcpenable=yes
tcpbindaddr=192.168.0.127 - адрес eth1 Астериска.

[authentication]

[6000]
.......
[avaya]
type=peer
host=192.168.0.1хх - Адрес Авая
transport=tcp
context=avaya
disallow=all
allow=ulaw
dtmfmode=rfc2833
qualify=yes

Всё равно не коннектиться! Что ему опять не так? Подскажите кто может? :(

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '19875c1d154df76b289f6c4d0b528c3b@192.168.102.4:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.0.1хх:5060:
OPTIONS sip:192.168.0.1хх SIP/2.0
Via: SIP/2.0/TCP 192.168.0.127:5060;branch=z9hG4bK2e26b263;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.127>;tag=as221579ad
To: <sip:192.168.0.1хх>
Contact: <sip:asterisk@192.168.0.127:5060;transport=TCP>
Call-ID: 59a685df4bc5a65e256eb75931d62f5a@192.168.0.127:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert1
Date: Thu, 04 Apr 2013 10:52:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from TCP:192.168.0.1хх:5060 --->
SIP/2.0 503 Service Unavailable(Signaling Resources Unavailable)
From: "asterisk" <sip:asterisk@192.168.0.127>;tag=as221579ad
To: <sip:192.168.0.1хх>;tag=0a2a38e4cafe2179c5151cd800
Call-ID: 59a685df4bc5a65e256eb75931d62f5a@192.168.0.127:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/TCP 192.168.0.127:5060;branch=z9hG4bK2e26b263;rport
Server: Avaya CM/R015x.02.1.016.4
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '59a685df4bc5a65e256eb75931d62f5a@192.168.0.127:5060' Method: OPTIONS
Вернуться к началу
ded
Сообщения: 15619
Зарегистрирован: 26 авг 2010, 19:00

Re: sip-tcp транк между Avaya 8800 и Астериском

Сообщение ded »

<--- SIP read from TCP:192.168.0.1хх:5060 --->
О! скрываем внутренние ИП адреса? Паранойя?
Вам не надо qualify=yes, не надо бомбардировать Авайю пакетами OPTIONS.
SIP/2.0 503 Service Unavailable(Signaling Resources Unavailable)
j о чём она и сообщает.
Сделайте
nmap 192.168.0.1хх -p 5060
посмотрите - открыт на Авайе ТСР порт 5060?
Вернуться к началу
Иван89
Сообщения: 11
Зарегистрирован: 02 апр 2013, 16:22

Re: sip-tcp транк между Avaya 8800 и Астериском

Сообщение Иван89 »

Почему-то лезет в др. подсеть 192.168.102.0 - сеть sip-телефонов eth1, а не eth0 192.168.0.127.

<--- SIP read from UDP:192.168.4.3:16922 --->
INVITE sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:16922;branch=z9hG4bK-d8754z-f2158eba5598bc6f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:6004@192.168.4.3:16922>
To: <sip:3311@192.168.102.4>
From: <sip:6004@192.168.102.4>;tag=b7e585f5
Call-ID: NjJhMzM5MzhkMDEwNzgyYWRiYjU0MDY2ODE5MDkzYjA
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5 stamp 69607
Content-Length: 272

v=0
o=- 13009547351444116 1 IN IP4 192.168.2.22
s=X-Lite 4 release 4.5 stamp 69607
c=IN IP4 192.168.2.22
t=0 0
m=audio 54570 RTP/AVP 107 9 0 8 100 101
a=rtpmap:107 BV32/16000
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 11 lines) ---
Sending to 192.168.4.3:16922 (NAT)
Using INVITE request as basis request - NjJhMzM5MzhkMDEwNzgyYWRiYjU0MDY2ODE5MDkzYjA
Found peer '6004' for '6004' from 192.168.4.3:16922

<--- Reliably Transmitting (NAT) to 192.168.4.3:16922 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.22:16922;branch=z9hG4bK-d8754z-f2158eba5598bc6f-1---d8754z-;received=192.168.4.3;rport=16922
From: <sip:6004@192.168.102.4>;tag=b7e585f5
To: <sip:3311@192.168.102.4>;tag=as401816fb
Call-ID: NjJhMzM5MzhkMDEwNzgyYWRiYjU0MDY2ODE5MDkzYjA
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="279ef063"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NjJhMzM5MzhkMDEwNzgyYWRiYjU0MDY2ODE5MDkzYjA' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.4.3:16922 --->
ACK sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:16922;branch=z9hG4bK-d8754z-f2158eba5598bc6f-1---d8754z-;rport
Max-Forwards: 70
To: <sip:3311@192.168.102.4>;tag=as401816fb
From: <sip:6004@192.168.102.4>;tag=b7e585f5
Call-ID: NjJhMzM5MzhkMDEwNzgyYWRiYjU0MDY2ODE5MDkzYjA
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.4.3:16922 --->
INVITE sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:16922;branch=z9hG4bK-d8754z-84e3f344f746c365-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:6004@192.168.4.3:16922>
To: <sip:3311@192.168.102.4>
From: <sip:6004@192.168.102.4>;tag=b7e585f5
Call-ID: NjJhMzM5MzhkMDEwNzgyYWRiYjU0MDY2ODE5MDkzYjA
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5 stamp 69607
Authorization: Digest username="6004",realm="asterisk",nonce="279ef063",uri="sip:3311@192.168.102.4",response="bff296740e9ec693dc560be0652c7381",algorithm=MD5
Content-Length: 272

v=0
o=- 13009547351444116 1 IN IP4 192.168.2.22
s=X-Lite 4 release 4.5 stamp 69607
c=IN IP4 192.168.2.22
t=0 0
m=audio 54570 RTP/AVP 107 9 0 8 100 101
a=rtpmap:107 BV32/16000
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 192.168.4.3:16922 (NAT)
Using INVITE request as basis request - NjJhMzM5MzhkMDEwNzgyYWRiYjU0MDY2ODE5MDkzYjA
Found peer '6004' for '6004' from 192.168.4.3:16922
== Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found unknown media description format BV32 for ID 107
Found audio description format speex for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x20000100c (ulaw|alaw|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.22:54570
Looking for 3311 in demo1 (domain 192.168.102.4)

<--- Reliably Transmitting (NAT) to 192.168.4.3:16922 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.22:16922;branch=z9hG4bK-d8754z-84e3f344f746c365-1---d8754z-;received=192.168.4.3;rport=16922
From: <sip:6004@192.168.102.4>;tag=b7e585f5
To: <sip:3311@192.168.102.4>;tag=as401816fb
Call-ID: NjJhMzM5MzhkMDEwNzgyYWRiYjU0MDY2ODE5MDkzYjA
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NjJhMzM5MzhkMDEwNzgyYWRiYjU0MDY2ODE5MDkzYjA' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.4.3:16922 --->
ACK sip:3311@192.168.102.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:16922;branch=z9hG4bK-d8754z-84e3f344f746c365-1---d8754z-;rport
Max-Forwards: 70
To: <sip:3311@192.168.102.4>;tag=as401816fb
From: <sip:6004@192.168.102.4>;tag=b7e585f5
Call-ID: NjJhMzM5MzhkMDEwNzgyYWRiYjU0MDY2ODE5MDkzYjA
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'NjJhMzM5MzhkMDEwNzgyYWRiYjU0MDY2ODE5MDkzYjA' Method: ACK
Вернуться к началу
Иван89
Сообщения: 11
Зарегистрирован: 02 апр 2013, 16:22

Re: sip-tcp транк между Avaya 8800 и Астериском

Сообщение Иван89 »

Спасибо! Сейчас проверю!
Вернуться к началу
ded
Сообщения: 15619
Зарегистрирован: 26 авг 2010, 19:00

Re: sip-tcp транк между Avaya 8800 и Астериском

Сообщение ded »

Как-то многовато у Вас там сетевых нелепиц. Телефон Х-лайт на 192.168.4.3, - вообще не из той подсети, и почему-то Астериску сказано, что он за НАТ, а он и открывает медиа на 192.168.2.22 - что это за адреса?
Peer audio RTP is at port 192.168.2.22:54570

Looking for 3311 in demo1 (domain 192.168.102.4) отсюда приходит ответ - Not Found и передаётся на Х-лайт
<--- Reliably Transmitting (NAT) to 192.168.4.3:16922 ---> он за НАТом? Почему? Это же внутренние ИП адреса?
SIP/2.0 404 Not Found
Вернуться к началу
Иван89
Сообщения: 11
Зарегистрирован: 02 апр 2013, 16:22

Re: sip-tcp транк между Avaya 8800 и Астериском

Сообщение Иван89 »

Телефон Х-лайт на 192.168.4.3, - вообще не из той подсети - да, всё верно. Он за NAT
192.168.2.22 ПК на котором Х-лайт.
<--- Reliably Transmitting (NAT) to 192.168.4.3:16922 ---> он за НАТом? Почему? Это же внутренние ИП адреса? Да внутренние.
А Авая и Астериск в DMZ зоне.
Вернуться к началу
Ответить

Вернуться в «Конфигурация и настройка Asterisk»

© 2008 — 2024 Asterisk.ru
Digium, Asterisk and AsteriskNOW are registered trademarks of Digium, Inc.
Design and development by PostMet-Netzwerk GmbH