chan_sip
- chan_sip 是什么？
- pjsip 之前的 sip 通道实现
- 功能满足 VoIP
- 不支持现代化通讯 - WebRTC, Websocket 等
caution
- chan_sip 已经废弃，尽量使用 chan_pjsip
modules.conf
autoload => no
load => chan_sip
# 自动加载
autoload => yes
# 排除 pjsip
noload => chan_pjsip
noload => res_pjsip
noload => res_pjsip_exten_state
noload => res_pjsip_log_forwarder
noload => res_pjsip_pubsub
noload => res_pjsip_session
Dial 语法
- SIP/devicename
- SIP/username@domain - SIP uri
SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]- 直接定义完整信息
- SIP/devicename/extension
- SIP/devicename/extension/IPorHost
- SIP/username@domain//IPorHost
- 添加
[![touser[@todomain]][![fromuser][@fromdomain]]]修改 To: 和 From:
- devicename - 定义的 peer 名字
- username@domain - 互联网上的 SIP 用户 - 建议开启 DNS SRV
- devicename/extension - 用于代理场景、Trunk 线路
SIP/proxyhostname/user和SIP/user@proxyhostname相同
命令
| cli | desc |
|---|---|
| sip notify | Send a notify packet to a SIP peer |
| `sip prune realtime [peer [< name > | all |
| sip qualify peer | Send an OPTIONS packet to a peer |
| sip reload | Reload SIP configuration |
| `sip set debug {on | off |
| `sip set history {on | off}` |
| `sip show {channels | subscriptions}` |
| sip show bridged | List all SIP bridges |
| sip show channelstats | List statistics for active SIP channels |
| sip show channel | Show detailed SIP channel info |
| sip show domains | List our local SIP domains |
| sip show history | Show SIP dialog history |
| sip show inuse | List all inuse/limits |
| sip show mwi | Show MWI subscriptions |
| sip show objects | List all SIP object allocations |
| sip show peers | List defined SIP peers |
| sip show peer | Show details on specific SIP peer |
| sip show registry | List SIP registration status |
| sip show sched | Present a report on the status of the scheduler queue |
| sip show settings | Show SIP global settings |
| sip show tcp | List TCP Connections |
| sip show users | List defined SIP users |
| sip show user | Show details on specific SIP user |
| sip unregister | Unregister (force expiration) a SIP peer from the registry |
sip.conf
设备配置
- type=peer
- 处理呼入呼出， ip/port 匹配
- host=dynamic
- type=user
- 处理呼入 - user 能呼叫 asterisk, asterisk 不能呼叫 user
- 通过注册信息匹配 - authname, secret
- 不依赖 IP 信息，不关心 host 设置
- type=friend
- 会创建 friend 和 user
- 一般本地电话使用
- remotesecret 呼出认证, secret 呼入认证
tip
- 推荐设备名字
[0-9a-z_] - 如果用到了 nat 确保
qualify=yes
通用配置
- 安全相关
- allowguest - 默认开启
- permit/deny/acl - IP address filters
- contactpermit/contactdeny/contactacl - IP address filters for registrations
- context - Which set of services you offer various users
[general]
; 默认 context - 默认为 default
context=public
; 是否允许匿名呼叫
;allowguest=no
; 尝试使用授权信息里的 username 匹配而不是 From 头
;match_auth_username=yes
; overlap dialing - use the Incomplete application to collect the needed digits from an ambiguous dialplan match.
; 默认 yes
; yes - RFC3578 overlap dialing
; dtmf - inband, RFC2833, SIP INFO
allowoverlap=no
; 禁止 transfer
;allowtransfer=no
; digest 授权 realm
; 默认 asterisk - 如果 asterisk.conf 设置了系统名则默认为系统名字 - 需要全局唯一
;realm=mydomain.tld
; 使用域名列表作为域
;domainsasrealm=no
; 处理 'Record: on' 头逻辑 - 默认 automon, 设置为 空 禁用
;recordonfeature=automixmon
; 处理 'Record: off' 头逻辑 - 默认 automon, 设置为 空 禁用
;recordofffeature=automixmon
udpbindaddr=0.0.0.0
;rtpbindaddr=172.16.42.1
; disallowed_methods = UPDATE
tcpenable=no
tcpbindaddr=0.0.0.0
tlsenable=no
tlsbindaddr=0.0.0.0
;tcpauthtimeout = 30
; unauthenticated sessions limit
;tcpauthlimit = 100
; 与 res_pjsip_transport_websockets 冲突
;websocket_enabled = true
; 单位 ms
;websocket_write_timeout = 100
transport=udp
; DNS SRV lookups on outbound
; 只会用 SRV 记录 - 禁用后外呼无法只使用域名
srvlookup=yes
;pedantic=yes
; https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service
; 全局 outbound 授权信息
[authentication]
; <user>:<secret>@<realm>
; <user>#<md5secret>@<realm>
;auth=mark:topsecret@digium.com
nat
| conf | desc |
|---|---|
| nat | 配置 NAT 处理逻辑 |
| qualify=yes | 保持 NAT 会话开启 |
Asterisk 不同版本 nat 的选项
| asterisk | nat |
|---|---|
>= 1.8 | no, force_rport, comedia, auto_force_rport, auto_comedia |
< 1.8 | no, yes, never, route |
chan_sip nat vs chan_pjsip
| chan_sip/nat | chan_pjsip/rtp_symmetric | chan_pjsip/force_rport | chan_pjsip/rewrite_contact |
|---|---|---|---|
| nat=yes | rtp_symmetric=yes | force_rport=yes | rewrite_contact=yes |
| nat=no | rtp_symmetric=no | force_rport=no | rewrite_contact=no |
| nat=never | rtp_symmetric=no | force_rport=no | rewrite_contact=no |
| nat=route | rtp_symmetric=no | force_rport=yes | rewrite_contact=yes |