一,为什么我的电话没有语音或只是单向的?
I am getting one-way or no audio on my calls. Why is that?
这儿问题可能是与防火墙或NAT有关,需要配置一下sip.conf
These problems are normally related to firewall/NAT issues. If your GOautodial server is behind a firewall, edit sip.conf:
#nano /etc/asterisk/sip.conf
将
;externip = 192.168.1.1
替换成
externip = 192.168.1.1
Where 192.168.1.1 is your public IP address. Reload Asterisk after the changes.
然后重新启动Asterisk
#asterisk -rx "reload"
二,为什么我的语音质量很差?
Why am I getting "choppy" calls? Why are most of my calls of poor quality? Are you inside a tunnel??
There are a lot of factors affecting the quality of calls. They are mainly:
原因是多方面的
1,Asterisk codec being used by the server
与Asterisk的编码方式有关;
2,Agent workstation
与客户端有关
3,Bandwidth consumption
网络带宽
4,Overloaded workstation
机器负载
5,Softphone (try to other softphones like zoiper, xlite and eyebeam)
与软电话
6,Poor quality headset (USB headsets are highly recommended)
耳机问题
如果你的网络带宽有限制,建议Asterisk的编码方式是GSM或G729,这两个编码比较高效.
另外检查一下你的网络,是否还能连上网,另外是否有其他应用已经将网络资源已经占用了.
If you have limited bandwidth, the codec used by your GOautodial server (to your SIP gateway) should either be GSM or G729. These are bandwidth efficient codecs.
You might also need to check the agents workstations to see if they're not overloaded. Meaning they're just running the necessary applications for dialing (Firefox, softphone, notepads or etc). Softphones eat CPU resources. If the workstation is overloaded then call quality can suffer.
Lastly, check your bandwidth consumption. You might be eating all your bandwidth. Make sure your internet connectivity is just being used for dialing purposes. Browsing social media sites like Facebook, Google+, Youtube and others will eat up most of your bandwidth.