[SR-Users] Can only tranfer (refer) a call one time

Youngjin Park gostyler at gmail.com
Sun Feb 27 20:35:18 CET 2011


> 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 <
> stangrze at 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 <ibc at aliax.net <mailto:ibc at aliax.net>>
>>>
>>>    2011/2/25 niklas rehnberg <niklas.rehnberg at gmail.com
>>>    <mailto:niklas.rehnberg at 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
>>>    <ibc at aliax.net <mailto:ibc at aliax.net>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
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