it is just bypass to enable call transfers for users connected to sipx, when uplink ITSP doesn't support REFER. sipxbridge works as SBC in b2bua mode, it stays in path of all calls to and from Your uplink ITSP and when it receives REFER i just makes new call (using INVITE) to Your ITSP, and then bridges new call with proper leg of previous call and disconects other one. so inside sipx 'domain' you can use call transers using refer, and from Your ITSP side there are only invites byes and cancels. in ex. in case when someone from outside calls you inside sipx, and you forward it back outside - from ITSP point of view there will be 2 calls, one to and one from your sipx, and he would have no idea that those calls are related.

drawback is of course that rtp streams are going back and forth through your internet uplink and AFAIR deaf silence at some point of transfer, but at least it works with all lousy ITSP's.

You should check sipx documentation or sipx mailing list, this thread is getting off-topic.

and REFER is not only for blind transfers ;-)

Grzegorz Stanislawski


W dniu 2011-02-27 20:35, Youngjin Park pisze:

    Hi,

    Can you tell me what advantage on INVITE has against REFER?

    REFER is a kind of blind transfer and INVITE in sipXbridge is 3PCC?

    Thanks in advance.

    Youngjin


    On Sun, Feb 27, 2011 at 6:40 AM, Grzegorz Stanislawski
    <stang...@netitel.pl <mailto:stang...@netitel.pl>> wrote:

        Hi.
        We have proverb for this, i don't know english version but it
        goes like this:
        "When it isn't known what it's all about, it's about money"

        Your ITSP had troubles with proper handling second transfer for
        billing purposes so decided to disable it.
        Proxy doesnt participate in call transfer, but ITSP it must
        charge users properly: Alice should pay just for call to Bob,
        Bob for "his" call to Charlie and so on.
        If You are using sipX You should use sipXbridge, it replaces
        REFER with INVITE and bridges calls.

        Grzegorz Stanislawski



        W dniu 2011-02-25 11:40, niklas rehnberg pisze:

            Hi,
            Thank for the quick response.
            The issue occur only when the Alice is a PSTN client.
            My ITSP says that they only supporting one call transfer
            (very strange).
            They can not explain why etc...
            PSTN client:              Alice
            MGW/MGC(ITSP):     Cisco/SER
            Our sip server:           SIPX

            BR Niklas
            2011/2/25 Iñaki Baz Castillo <i...@aliax.net
            <mailto:i...@aliax.net> <mailto:i...@aliax.net
            <mailto:i...@aliax.net>>>

                2011/2/25 niklas rehnberg <niklas.rehnb...@gmail.com
            <mailto:niklas.rehnb...@gmail.com>
            <mailto:niklas.rehnb...@gmail.com
            <mailto:niklas.rehnb...@gmail.com>>>:
             > Hi,
             > Have following issue:
             >
             > Alice calling Bob.
             > Bob make call transfer to Charlie  (works fine)
             > Charlie transfer Alice to David.    (the call break)
             >
             > Why is not possible to transfer the call more than one time?
             > Is it any parameters?
             >
             > My ITSP use SER together with Cisco MGW.

                Niklas, nothing in SIP protocol neither in SER/Kamailio
            makes your
                scenario to fail. It must be a problem in your custom
            setup. Try
                identifying the problem capturing SIP traces.
                Also take into account that a proxy doesn't participate
            at all in the
                process of a "call transfer". It's just a transparent
            mechanism for a
                proxy.

                --
                Iñaki Baz Castillo
            <i...@aliax.net <mailto:i...@aliax.net> <mailto:i...@aliax.net
            <mailto:i...@aliax.net>>>




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