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AS5300 peer to peer problem

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  • #25894

    Dear all ! I’m a beginner on AS5300 !

    I have a problem on a pair of AS5300 point to point toll bypass configuration !

    My AS5300 is connected to our traditional ISDN PRI voice switch. And the other AS5300 send their calls via h.323 to our as5300, then routed out through our voice switch. (which is a toll-bypass configuration)

    The problem is the calling party can’t hear any progress tone / ringback tone / vacant number announcements … etc

    I’ve the statement “cptone US” under the “voice-port 2:D” config. Any other commands needed on AS5300 dial peers ??

    Looking forward to your kindly advice !!

    Teodor Georgiev

    I remember, there was a document of Cisco, describing exactly that problem – absence of ring-back tones when a toll-free number is dialed. Browse, to find it.


    You already spent so much money.
    There’s an open source solution out there called Asterisk ( you may want to look into that could have very easily handled that for you.

    Telephony Services:
    Voicemail System
    Password Protected
    Separate Away and Unavailable Messages
    Default or Custom Messages
    Multiple Mail Folders
    Web Interface for Voicemail Checking
    E-mail notification of Voicemail
    Voicemail Forwarding
    Visual Message Waiting Indicator
    Message Waiting Stutter Dialtone
    Auto Attendant
    Interactive Voice Response
    Overhead Paging
    Flexible Extension Logic
    Multiple Line Extensions
    Multi-Layered Access Control
    Direct Inward System Access
    Directory Listing
    Conference Bridging
    Unlimited Conference Rooms
    Access Control
    Call Queuing
    ADSI Menu System
    Support for Advanced Telephony Features
    PBX Driven Visual Menu Systems
    Visual Notification of Voicemail
    Call Detail Records
    Local Call Agents
    Remote Call Agents
    Protocol Bridging
    Provides seamless integration of technologies
    Offers a unified set of services to users regardless of connection type
    Allows interoperability of VoIP systems
    Call Features:
    Music on Hold
    Music on Transfer
    Flexible mp3 based system
    Volume Control
    Random Play
    Linear Play
    Call Waiting
    Caller ID
    Caller ID Blocking
    Caller ID on Call Waiting
    Call Forward on Busy
    Call Forward on No Answer
    Call Forward Variable
    Call Transfer
    Call Parking
    Call Retrieval
    Remote Call Pickup
    Do Not Disturb
    Allows Direct Connection of Asterisk PBX
    Offers Zero Latency
    Uses Commodity Ethernet Hardware
    Voice over IP
    Allows for Integration of Physically Separate Installations
    Uses commonly deployed data connections
    Allows a unified dialplan across multiple offices
    Voice over IP Interoperability:
    Asterisk provides transparent bridging between Voice over IP protocols and traditional telephony equipment. In addition, Asterisk can transfer calls from one system to another via the Inter Asterisk Exchange protocol.

    Inter-Asterisk Exchange (IAX)
    Session Initiation Protocol (SIP)
    Media Gateway Control Protocol (MGCP)
    Traditional Telephony Interoperability
    Robbed Bit Signaling Types
    FXS and FXO
    E&M Wink
    Feature Group D
    PRI Protocols
    Lucent 5E
    National ISDN2
    BRI (ISDN4Linux)
    Codec Support
    G.729 (available through purchase of commercial license(s))
    G.723.1 (pass through)
    MP3 (decode only)


    Actually, you can still use it in your current situation… do you have a spare PC laying around? How many simultaneous ports do you normally use during “peak” hours?

    Drop me a line or post here and I’ll see if I can help you out without having to drain your pockets.

Viewing 4 posts - 1 through 4 (of 4 total)
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