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A
Anonymous
November 9, 2010
Question

Two way audio calls?

  • November 9, 2010
  • 1 reply
  • 1014 views

I just tried to call from a Flash Client to a phone using a SIP account. It worked well!

"FMG enables two way audio calls between Flash clients and PSTN/Mobile  phones. All you needs is a SIP account with a SIP service provider or on  an in-premise SIP capable PBX."

Does that mean that there is also a possibility to call a Flash client from a phone (for example by calling the number of a SIP account from a PSTN phone)?

This topic has been closed for replies.

1 reply

A
Anonymous
November 9, 2010

Congratulations for your first test call using FMG !

Yes, FMG can be used to route incoming calls from SIP (trunks) to flash clients.With default settings, provided a sample flash phone with destination number is open, an incoming call from SIP side to flash phone number (say 1001, 1002 ... )will be routed to the Sample Flash Phone with respective number;

The term "two way" is used in the context of media flow being interactive i.e. both end point can send media.

-Pankaj

A
Anonymous
November 9, 2010

Dear Pankaj,

thanks for your quick response!

I can see in the log-files, that the FMG reacts if I´m calling the SIP number from the PSTN phone:

2010-11-09::14:42:19đ128    DEBUG    SIP    7424    Trying to find matching sip profile in Sip Database
2010-11-09::14:42:19đ128    DEBUG    SIP    7424    Trying to find matching sip profile through TO hdr for username : [my SIP number]
2010-11-09::14:42:19đ128    DEBUG    SIP    7424    Trying to find matching sip profile through FROM hdr for username : [my PSTN number]
2010-11-09::14:42:19đ128    DEBUG    SIP    7424    Trying to find matching sip profile through CONTACT hdr for username : [my PSTN number]

How can I tell the FMG to route these calls to the flash phone number? Creating a <Profile> with my PSTN number as <profileID>? How would it look like? I couldn´t find an example for that.

Thanks in advance!

P
Piyush-G
Participating Frequently
November 9, 2010

ProfileID could be any identifier. The username tag would contain [my SIP number]. If <doRegister> is 1, then username/password will be used by FMG also to register itself at SIP server. <remoteSipHost> contains the IP address (or FQDN) of the SIP server. The value of <context> tag is used to select appropriate context in the workflow.xml for call routing.
Inside context of same name (in workflow.xml), we can specify a rule to route the call as follows:
<Condition variable="destNum" value="[username i.e. SIP number]">
   <AppNode sequence="1" app="bridge" args="rtmp|<flash_phone_number say 1001>@profile_default" />
   <AppNode sequence="2" app="hangup" args="null"/>
</Condition>

-Piyush

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