Sometimes, we need limit the call right of some extensions. For example, we want to limit only specific extensions can make out-group calls to outsides, others can only make calls between extensions.
In MSS, we use “call level” feature to do it. By default, we don’t assign “call level” to any called number prefix in “analyze called number” table. That means all extensions can have the same right. To limit extensions, we should indicatedifferent “call level” to the called number prefix and assign relative “call level” right to special extensions, then they will have the right to make such calls.
For example, the default out-group call prefix is “9”. Please click menu “Dial plan / Analyze called number” and edit or add a record whose prefix is “9” and route type is “external line”. In this configuration, we can select “call level 1” to this prefix “9”.
Then, please click menu “data / local users” and edit or add a local user. In the pop-up dialog, please click “Basic Call” tab and enable “Call level 1” to this extension.
After that, the extension has the “call level 1” right to make calls to outside by dialing “9xxxxx”. For others, since they don’t have “call level 1” right, their calls will be rejected when they dial “9xxxxx”.
Almost all of us will meet this problem when we deploy our first VoIP network. We are often confused: why I cannot hear peer guy but he can hear me? why we cannot hear each other?
The root reason is that there is private network and public network. If both sides are in different network, the problem will happen. Please look at below figure which desribe a very simple VoIP network:
In this simple network, we have two VoIP devices, one is SIP phone whose number is 100, another is SIP client whose number is 101.
SIP phone is in a private network and its private address is 192.168.1.100, and its router is connected to public network with address 18.104.22.168.
SIP client is installed in one PC which is in the public network with address 22.214.171.124.
So when SIP phone makes a call to the SIP client, what will happen?
SIP phone say: Hi, I am 100, my audio address is 192.168.1.100. Please send audio stream to me.
SIP client answers it: ok. I am 101, my audio address is 126.96.36.199. Please send your audio to me.
SIP phone sends audio stream to SIP client. Since “188.8.131.52 ” is a public address, it is no problem for SIP client to receive the audio stream from SIP phone. That means SIP client can hear SIP phone now.
SIP client sends its audio stream to SIP phone “192.168.1.100”. You can see it is a private address and cannot be reached by SIP client which is in public address. SIP client will fail to send its audio stream to SIP phone in fact.
So finally, SIP client can hear SIP phone, but SIP phone cannot hear SIP client. This is a very typical one-way audio problem.