我使用的是Ubuntuv14.04.3 LTS和星号13.3.2。当我试图从一个sipml5客户端调用我的分机,只播放一个演示-祝贺音频,我的电话立即中断。当我检查星号日志时,会得到以下错误
[2016-08-24 06:07:49] ERROR[31730][C-0000000c]: res_rtp_asterisk.c:2042 __rtp_recvfrom: DTLS failure occurred on RTP instance '0x7f547c013c68' due to reason 'sslv3 alert handshake failure', terminating
[2016-08-24 06:07:49] WARNING[31730][C-0000000c]: res_rtp_asterisk.c:3911 ast_rtcp_read: RTCP Read error: Unspecified. Hanging up.
[2016-08-24 06:07:49] WARNING[31730][C-0000000c]: app_playback.c:493 playback_exec: Playback failed on SIP/104600-00000007 for /var/www/html/fetch_prompt
[2016-08-24 06:07:49] ERROR[31730][C-0000000c]: utils.c:1402 ast_carefulwrite: write() returned error: Broken pipe我正在使用Chrome v54。
我认为这个错误与openssl有关,但是还没有得到一个正确和完整的答案来解决这个问题。有谁知道如何解决这个问题吗?
发布于 2016-08-30 10:41:44
通过升级openssl解决了这个问题。使用以下命令升级Ubuntu 14中的openssl
# echo 'deb http://us.archive.ubuntu.com/ubuntu/ xenial main restricted universe multiverse' > /etc/apt/sources.list.d/xenial.list
# aptitude update
# aptitude install -y openssl libssl-dev
# rm /etc/apt/sources.list.d/xenial.list
# aptitude update使用以下命令检查openssl版本
# ldd /usr/sbin/asterisk | grep libssl
libssl.so.1.0.0 => /lib/x86_64-linux-gnu/libssl.so.1.0.0 (0x00007f33ce117000)
# strings /lib/x86_64-linux-gnu/libssl.so.1.0.0 | grep 1.0.2
OPENSSL_1.0.2
OPENSSL_1.0.2g
SSLv3 part of OpenSSL 1.0.2g-fips 1 Mar 2016
TLSv1 part of OpenSSL 1.0.2g-fips 1 Mar 2016
DTLSv1 part of OpenSSL 1.0.2g-fips 1 Mar 2016
OpenSSL 1.0.2g-fips 1 Mar 2016
# openssl version
OpenSSL 1.0.2g-fips 1 Mar 2016在此之后,删除所有现有的星号键,并再次重新创建键。
# rm /etc/asterisk/keys/*
# cd /usr/src/astersik*/contrb/scripts
# sudo ./ast_tls_cert -C pbx.mycompany.com -O "My Super Company" -d /etc/asterisk/keys
# asterisk -rx "reload"https://serverfault.com/questions/798743
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