Chapter 3 – Basic Operation

Dialing method between H.323 and SIP Participants

The RMX system can work as a gateway to enable the H.323 or SIP participants inviting the SIP or H.323 participants when creating a conference.

If the GK and SIP server are not configured on the network, dialing is executed in the same way as for inviting the same IP type endpoint. For dialing strings please refer to the previous section.

If the RMX 1000 and the endpoint are registered with the GK or SIP server, the format of the dial-in string is as follows:

 

Dialing Direction

 

 

Dial-in String

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

<RMX 1000 E.164 prefix>[Conference ID][##Conference

 

 

 

 

 

 

 

 

 

 

password][##Chairperson password]<* SIP participant's

 

 

 

 

 

TEL URI>

 

 

 

 

 

Here, the character in the <> symbol is mandatory, and that in

 

 

H.323->SIP

 

[ ] is optional.

 

 

 

Note: For an H.323 conference creator to invite a SIP

 

 

 

 

 

 

 

 

 

 

participant, if the H.323 endpoint has been registered to the

 

 

 

 

 

same gatekeeper as RMX 1000, the SIP endpoint needs to

 

 

 

 

 

register the SIP server with a numeric URI and the RMX 1000

 

 

 

 

 

should also register the same SIP server.

 

 

 

 

 

 

 

 

 

 

 

[Conference ID][$$Conference password][$$Chairperson

 

 

 

 

 

password]<* H.323 participant's E.164 prefix>@<RMX 1000

 

 

 

 

 

static route domain name>

 

 

 

 

 

Here, the character in the <> symbol is mandatory, and that in

 

 

SIP->H.323

 

[ ] is optional.

 

 

 

 

 

Note: For an SIP conference creator to invite an H.323

 

 

 

 

 

participant, if the SIP endpoint has been registered to the same

 

 

 

 

 

SIP server as RMX 1000, both the H.323 endpoint and the

 

 

 

 

 

RMX 1000 need to register the same gatekeeper.

 

 

 

 

 

 

 

 

In the gateway scenario, the user can create a conference between an H.323 participant and a SIP participant as a point-to-point call through the procedures below:

1Select the Terminate the call when last participant remains option in the default profile and keep the default idle time value (0). For more information about conference profile, please see Defining a Conference Profile.

2Dial [RMX Prefix]*[another participant’s E.164 or SIP URI]

In that way, when one participant disconnects, the other participant will be disconnected automatically like in a phone call.

Conference ID is not mandatory. If the user only dial the RMX 1000 E.164/IP plus *participant’s E.164/SIP URI/IP, a conference will be created with a random NID.

The conference ID the user enters for creating a new conference must be unique - different from existing conference IDs.

If only one password is entered, it will be defined as a chairperson password. In this case, there won’t be a conference password and the user will receive the chairperson privilege.

The conference password and chairperson password must be different. If a user enters the same password for both, the call will be rejected.

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Image 37
Polycom 3150-16966-002, RMX 1000 manual Dialing method between H.323 and SIP Participants, Tel Uri