Call not releasing / disconnecting properly with analog devices in a SfB Enterprise Voice deployment !
Recently we deployed a SfB Enterprise in a Production site (We used AudioCodes Mediant 1000 PSTN Gateway + SfB SBA + AudioCodes Analog Gateway) in this site. The Analog gateway was to integrate the Analog devices with the SfB clients and Polycom VVX Phones.
Requirement: In a SfB Enterprise Voice deployment scenario, users from their SfB client, Polycom VVX phones should be able to dial Announcement system (by dialing a phone number) and make announcements successfully.
We deployed AudioCodes SBA for a Branch site. Since it is a manufacturing site it had a requirement to integrate the announcement systems. These Analog phones are required to meet the regulatory requirements of any Manufacturing sites).
So, we also installed an Analog Gateway (AudioCodes Media Pack MP124) and connected the analog devices like (analog phones and the announcement systems) to it.
In the SfB Environment, We had the SfB Dial plan created with all the normalized rules.
Like users dial only the last 4 digits and it gets converted to the +
Issue: A call made from a SfB client or VVX Polycom Phone to the announcement works but when the Call ends (either from SfB client you disconnect or when you disconnect the call from the Phone VVX phone), the phone line does not get disconnected or released.
Troubleshooting:
We connected an Analog Phone to the Media pack port and then test by dialing the number for example 8xxx.
The analog phone rang and when we hung up the phone it also disconnected we did not get the busy or fast busy tone this time. However, only when we connect the actual Announcement device (Which is an analog device) and when we call the announcement device it accepts the call and we can make the announcement through it. However, When we disconnect the call from our end the call does not get released instead we get a fast busy.
To fix the issue:
1) Created a new Coder (Gateway --> Configuration --> Coders and Profiles --> Coders created a new Proifle ID 1.
a) Enabled Polarity Reversal to Enable
When this feature is enabled, the analog port (FXS) interface polarity is reversed to indicate the start of a VoIP session and it is reversed back when the VoIP session ends.
b) Enable Current Disconnect to Enable
By enabling these two parameters the call is immediately disconnected after polarity reversal or current disconnect is detected on the Tel side (assuming the PBX / CO produces this signal).
Creating a new Tel Profile:
Once you create the new Tel Profile 1 then you need to assign it to the required phone number in my case it is Analog Announcement system like shown below.
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