Транк с TLS

[olnn]

В консоли вообще ничего нет

[ded]

Астериск за Натом

В чем может быть проблема (в SIP приложения входящий вызов есть)

В консоли нет, а что такое в SIP приложения входящий вызов есть?

[gland]

Наверное, имеется в виду софтфоны)
Интересно, что он всё это время делает?
фига, только сейчас заценил,что чел тут с 2011 года..

[whoim]

Код: Выделить всё

#install libsrtp
yum -y remove libsrtp
cd /usr/src/
wget http://srtp.sourceforge.net/srtp-1.4.2.tgz
gunzip srtp-1.4.2.tgz
rm -f srtp-1.4.2.tgz
cd /srtp
autoconf
./configure CFLAGS=-fPIC
make
make runtest
make uninstall
make install

Код: Выделить всё

#generate asterisk TLS certificates
export TSL_HOSTNAME=server.dns.name
export TLS_EMAIL=my@email.com
mkdir /etc/asterisk/ssl
cd /etc/asterisk/ssl
mkdir db
mkdir db/certs
mkdir db/newcerts
touch db/index.txt
echo "01" > db/serial
openssl req -new -newkey rsa:1024 -nodes -keyout ca.key -x509 -days 500 -subj /C=RU/ST=Russia/L=Runet/O=${TLS_HOSTNAME}/OU=asterisk/CN=${TLS_HOSTNAME}/emailAddress=${TLS_EMAIL} -out ca.crt
cp ca.crt asterisk.crt
cat ca.key >> asterisk.crt
#generate client cert
openssl req -new -newkey rsa:1024 -nodes -keyout client.key -subj /C=RU/ST=Russia/L=Runet/O=Inc/OU=SIP/CN=${TLS_HOSTNAME}/emailAddress=${TLS_EMAIL} -out client.csr
rm -f ca.config && wget http://vmclouds.ru/dnld/ca.config && chown asterisk:asterisk ca.config
openssl ca -config ca.config -in client.csr -out client.crt -batch
chown -R asterisk:asterisk /etc/asterisk/ssl

[whoim]

ессно, тут поле для творчества - юзать клиентский серт или нет, проверять серт сервера или нет. Это для примера. Когда нужно только шифрование без заморочек с копированием сертификатов тудасюда.

[olnn]

Астериск за натом

При входящем вызове в консоле сообщения
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

При отладке
A.A.A.A - адрес провайдера
B.B.B.B - мой внешний адрес астериска


[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: --- (17 headers 44 lines) ---
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Sending to A.A.A.A:5061 (NAT)
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Using INVITE request as basis request - 0da319ef0886f78966b8006002fc6559@XXX.111.140.234
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found peer '031712' for '031711' from A.A.A.A:5061
[Apr 23 11:13:46] VERBOSE[3216] netsock2.c: == Using SIP RTP TOS bits 184
[Apr 23 11:13:46] VERBOSE[3216] netsock2.c: == Using SIP RTP CoS mark 5
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 0
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 8
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 18
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 4
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 3
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 112
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 5
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 110
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 97
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 111
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 9
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 102
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 115
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 116
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 117
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 10
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 118
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 119
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 120
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found RTP audio format 101
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format PCMU for ID 0
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format PCMA for ID 8
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format G729 for ID 18
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format G723 for ID 4
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format GSM for ID 3
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format AAL2-G726-32 for ID 112
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format DVI4 for ID 5
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format speex for ID 110
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format iLBC for ID 97
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format G726-32 for ID 111
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format G722 for ID 9
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format G7221 for ID 102
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format G7221 for ID 115
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format G719 for ID 116
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format speex for ID 117
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format L16 for ID 10
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format L16 for ID 118
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found unknown media description format speex for ID 119
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found unknown media description format ip-mr_v2.5 for ID 120
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 23 11:13:46] WARNING[3216] chan_sip.c: Rejecting text media offer due to invalid or unsupported syntax: text 49972 RTP/SAVP 105 106
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c:
<--- Reliably Transmitting (NAT) to A.A.A.A:5061 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/TLS A.A.A.A:5061;branch=z9hG4bKf7d3.5409f086314b8329ead6ba37765c05d6.0;received=A.A.A.A;rport=5061
Via: SIP/2.0/UDP 10.12.0.1:6050;received=10.12.0.1;branch=z9hG4bK24e0e6f5;rport=6050
From: "031711" <sip:031711@XX.111.140.234>;tag=as1e255da2
To: <sip:031712@10.11.0.2:5060>;tag=as170cc3d2
Call-ID: 0da319ef0886f78966b8006002fc6559@XXX.111.140.234
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: Scheduling destruction of SIP dialog '0da319ef0886f78966b8006002fc6559@XXX.111.140.234' in 6400 ms (Method: INVITE)
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c:
<--- SIP read from TLS:A.A.A.A:5061 --->
ACK sip:031712@B.B.B.B:36096;transport=TLS SIP/2.0
Via: SIP/2.0/TLS A.A.A.A:5061;branch=z9hG4bKf7d3.5409f086314b8329ead6ba37765c05d6.0
Max-Forwards: 69
From: "031711" <sip:031711@XXX.111.140.234>;tag=as1e255da2
To: <sip:031712@10.11.0.2:5060>;tag=as170cc3d2
Call-ID: 0da319ef0886f78966b8006002fc6559@XXX.111.140.234
CSeq: 102 ACK
Content-Length: 0

<------------->
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 23 11:13:47] VERBOSE[3174] chan_sip.c: Really destroying SIP dialog '0da319ef0886f78966b8006002fc6559@XXX.111.140.234' Method: INVITE

[awsswa]

[Apr 23 11:13:46] WARNING[3216] chan_sip.c: Rejecting text media offer due to invalid or unsupported syntax: text 49972 RTP/SAVP 105 106

SIP/2.0 488 Not acceptable here

Что о я не улавливаю
Via: SIP/2.0/UDP 10.12.0.1:6050;received=10.12.0.1;branch=z9hG4bK24e0e6f5;rport=6050
From: "031711" <sip:031711@XX.111.140.234>;tag=as1e255da2
To: <sip:031712@10.11.0.2:5060>;tag=as170cc3d2

Рисуйте схему с IP адресами

[ded]

[Apr 23 11:13:46] WARNING[3216] chan_sip.c: Rejecting text media offer due to invalid or unsupported syntax: text 49972 RTP/SAVP 105 106
[Apr 23 11:13:46] VERBOSE[3216] chan_sip.c:
<--- Reliably Transmitting (NAT) to A.A.A.A:5061 --->
SIP/2.0 488 Not acceptable here

КАкое слово перевести?

[olnn]

Как я понимаю не правильных синтаксис INVITE

[olnn]

и возможно не хватает кодека