No relevant resource is found in the selected language.

This site uses cookies. By continuing to browse the site you are agreeing to our use of cookies. Read our privacy policy>

Reminder

To have a better experience, please upgrade your IE browser.

upgrade

connect PBX via VC system to make the audio site join in the conference.

Publication Date:  2013-07-26 Views:  68 Downloads:  2
Issue Description
How to config the system to make the audio site join in our video system.
Alarm Information
no
Handling Process
What is PBX?
Private Branch Exchange (PBX) is a customer premise switch used to connect customer telephones to LEC central office lines (trunks) as well as switch internal calls within the customers' premises, usually with software-controlled features such as call forwarding and call pickup.


2. We search the details in the Guide, but find little details in it.
Configuring the SIP Trunk
Currently, the MCU supports the SIP trunk configuration with simple rules. After the SIP trunk is configured, the MCU uses trunk addresses to call SIP sites defined with site numbers. Figure 20-6 shows SIP trunk settings.
SIP trunk settings


3. But we find that it mainly show the following steps:
a. First open the sip service。We can enter the command as follows
system-view sip-config-----------0: disable; 1:enable
b. And configure the settings as follows。
① SIP trunk enable:1-----------------open the sip trunk channel
② SIP trunk prefix/suffix------------configure the prefix/suffix of dial number.
③ SIP trunk ip----------------------- set the IP addr, of the other side.
④ SIP trunk connection type--------choose the ways of connection(communicate)
⑤ SIP trunk port----------------------configure the port(default)
Here ③④⑤ should be compatible with the other side,any issue of communication we can capture packets to check。
And MCU can configure many sip channels and it should be corresponding with the settings in the PBX.
c. In the other PBX side, we should give MCU one/more numbers to make the number reachable。
d. After we configure the settings as blow, the main thing finish。We can use a telephone call MCU number directly , and check whether you can hear IVR `welcome to huawei videoconference system` .
e. Then we can test it in the SMC2.0 software。
Add 1 sip site and check whether it can join in our conference.
      If everything goes well , the sip site will join in the conference, and we can use 7#/8# to mute/unmute the site。Of course the sip site can call MCU number, and we will hear the IVR voice and also will join in our conference.
Root Cause
no
Suggestions
After the configuration , we can call the sip site and they can join in easily. But we still met some issues about hanging up by MCU/SMC2.0, I will write some other cases.
We suggest you be familiar with the normal settings.

END