我正在使用 Ubuntu v14.04.3 LTS 和 Asterisk 13.3.2。当我尝试从 sipml5 客户端呼叫我的分机以播放演示祝贺音频时,我的呼叫立即断开。当我检查星号日志时,我收到以下错误
[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。
我认为这个错误是 openssl 造成的,但目前还没有得到正确完整的答案来解决这个问题。有人知道如何解决这个问题吗?
答案1
通过升级 openssl 解决了这个问题。使用以下命令在 Ubuntu 14 中升级 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
此后,删除所有现有的星号密钥并重新创建密钥
# 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"