2020-7-23 · Configuration file for Asterisk SIP channels, for both inbound and outbound calls.. Starting with Asterisk v1.2.0: The global option “port” in 1.0.X that is used to set which port to bind to has been changed to “bindport” to be more consistent with the other channel drivers and to avoid confusion with the “port” option for users/peers.
5 使用Asterisk作为SIP中继服务器 6 双SIP NAT环境中的Android SIP 7 perl调用ruby脚本使用反引号返回任何内容 8 Asterisk无法在LTE（4G）网络上传送声音 9 SIP的配置NAT（Asterisk） 10 Asterisk只能在Wifi网络上听不到声音 loading Asterisk - sip_parse_nat_option: nat=yes is deprecated 2019-11-18 · nat=yes is working for asterisk version 10 or older. From asterisk 11 , nat=yes is depricated. They said nat=yes and nat=force_rport,comedia are same. But i think both are different. If we change to nat=force_rport,comedia the behavior seems to be fine, except for outside users behind NAT. Our server is also behind NAT. asterisk - NAT configuration for SIP - Stack Overflow 2020-2-1 · No NAT in the middle #9 is solved with nat=yes and qualify=xxx in sip.conf for the client in most cases. Some clients (X-lite) assist themselves by using STUN and sending UDP keep-alive packets. Qualify sends keep-alive packets from Asterisk to the client on the inside.
2018-11-4 · SIP-NAT機能を利用することでAsteriskをNAT背後で動作させることが可能です。 ヤマハのルータにはSIP-NAT機能を持つものがあり動作実績があります。 ただし、この方法についてはヤマハが保証しているわけではありませんので、ヤマハには問い合わせないで下さい。
May 05, 2014 · You can use SIP and NAT if your firewall has application level SIP inspection. Otherwise, even forwarding all traffic from a public IP to the server's private IP won't work. The problem is when your server sends a SIP invite to an external server, it will tell the server it is contacting what IP address it should send the audio to. From the JIRA issue: I have some devices in the following scenario: Asterisk server with public IP address Mobile devices (clients): When in internal network, no NAT between the client and the server When in "roaming" (i.e. a Hotel with WiFi), the client is behing a NAT When in 3G, operator transparent sip proxy so it looks as no NAT, but does
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Asterisk的配置详解 - 魔流剑 - 博客园 2016-3-23 · Asterisk中RTP使用大数字的无特权的端口（默认10000至20000） SIP不是第一个，也不是唯一一个我们使用的VOIP协议（其它包括H.323,MGCP,IAX等），但是目前SIP似乎是硬件厂商最大的动力。SIP协议的优点是普遍的被接受和结构灵活，简单。 下面是基本 Asterisk NAT - VoIP-Info.jp 2018-11-4 · SIP-NAT機能を利用することでAsteriskをNAT背後で動作させることが可能です。 ヤマハのルータにはSIP-NAT機能を持つものがあり動作実績があります。 ただし、この方法についてはヤマハが保証しているわけではありませんので、ヤマハには問い合わせないで下さい。 How to setup your Asterisk PBX if you are behind a NAT If your Asterisk PBX is behind a NAT firewall, i.e. the PBX has an IP such as 192.168.0.2 then you will need to perform additional configuration to allow Asterisk to route the SIP and RTP correctly. The NAT configuration can be found in the file /etc/asterisk/sip.conf, the relevant section that needs to be edited is reproduced below: asterisk 问题 - 单车博客园 - 博客园