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  • SIP trunks via AudioCodes Mediant 800

    Hi All,

    I am wondering if there is anyone out there who may have setup SIP trunks on a shoretel system via an audiocodes mediant 800.

    Specifically using a carrier which rather than sending the DID called in the Request URI sends the GDN or registration ID, only sending the DID called in the TO header. This would mean you would have needed to modify the sip messages on the shoretel to copy the to header to the Request URI field or ShoreTel will assume you are calling the main DID/Registration ID.

    Also I am only using a single interface on the audiocodes meaning I have only one LAN IF active and in use and a single SIP interface.

    I am having issues where I can manipulate the Request URI to present the called number from the TO header which means incoming calls will now route correctly on the shoretel side, but I am having an issue where calls disconnect after approx 30 seconds. It seems as though this is because no response is given to the shoretel's 200 OK SDP messages.

    I am also finding that having the manipulation in the audiocodes to replace the incoming Request URI breaks the ability to dial out.

    Has anyone had any issues like this that may be able to help?