Asterisk Letting SIP clients connect directly
2010-11-22 13:44
190 查看
Asterisk by default connects all media streams through asterisk to be able to connect various protocols and media to each other.
If you have two SIP phones, the media path can be connected directly between the phones without going through Asterisk. Asterisk in this case only handles signalling. It requires that both extensions are using SIP and support the same codecs.
canreinvite=yes
in the configuration of the SIP extension. This is the default behaviour.
secret=thesweet43
type=friend
host=dynamic
context=sipexts
mailbox=1050
callerid="morgan@yourdomain.com" <1050>
dmtfmode=rfc2833
canreinvite=yes
See Asterisk sip canreinvite for more information.
If you have two SIP phones, the media path can be connected directly between the phones without going through Asterisk. Asterisk in this case only handles signalling. It requires that both extensions are using SIP and support the same codecs.
Disable transfer
If you want to transfer calls by pressing the # key during a call, Asterisk will stay in the media stream to be able to listen for # signals. Remove the "tT" from the dial() command to disable this.Configuration
This is done in sip.conf by usingcanreinvite=yes
in the configuration of the SIP extension. This is the default behaviour.
Example
[morgan]secret=thesweet43
type=friend
host=dynamic
context=sipexts
mailbox=1050
callerid="morgan@yourdomain.com" <1050>
dmtfmode=rfc2833
canreinvite=yes
Please note
There are SIP clients that do not work well with these settings, like the Cisco ATA 186.See Asterisk sip canreinvite for more information.
相关文章推荐
- Asterisk Letting SIP clients connect directly
- 如何实现在自己编写的asterisk用户平台实现添加Extensions , sip ,user!
- Asterisk Sip.conf 配置说明
- Asterisk sip canreinvite
- Asterisk 1.8 sip 协议栈分析 2
- Asterisk的SIP type和身份认证
- Asterisk配置SIP服务器
- asterisk的sip.conf中nat选项说明
- asterisk配置SIP服务器
- qutecom注册到asterisk上,sip信令使用rc4加密方法
- Asterisk SIP user vs peer
- Asterisk 1.8 sip 协议栈分析(1)
- Android’s HTTP Clients (httpClient 和 httpURLConnect 区别)
- Asterisk配置SIP服务器
- Asterisk SIP media path
- Asterisk 1.8 sip 协议栈分析(2)
- Asterisk配置SIP服务器
- Asterisk realtime 之SIP用户动态写入mysql 数据库(2)
- Connect two clients behind NAT
- Asterisk SIP user vs peer