[OpenSIPS-Users] Knowing when a call was diverted

[Jeff Pyle]

If all endpoints in your SIP network supports REFER and INVITE with
Replaces correctly, all you have to do with OpenSIPS is route the SIP
messages between the endpoints for transfer to work correctly.  It's up to
the endpoints to understand what to do with the dialogs, media sessions,
etc.  In my experience this level of compatibility is rare unless you have
a completely private SIP network and control every endpoint.  I mean
phones, gateways, soft clients, everything that handles a media stream,
really.

If that is the case for you, then I would think all you need to do is
account the REFERs within the acc module.  Granted that's an initial
reaction and I haven't thought through it completely.

If you connect to the PSTN through a SIP-speaking carrier, chances are they
do not support REFER.  In this case you'll need something heavier duty in
your network than OpenSIPS, some type of a B2BUA (i.e. FreeSWTICH) that can
isolate segments of your network and handle the REFERs on behalf of the
elements that cannot.  Some may call this implementation an "SBC", although
that term is ambiguous at best.

Hopefully this helps illustrate what is and is not possible with OpenSIPS.

Attended transfer is quite involved at the SIP level.  If you want to try
it with OpenSIPS, you're going to have to get your hands dirty, so to
speak.  You'll want to check out a SIP ladder diagram[1] of what's
happening between the endpoints during such a transfer.  The good news is
that all OpenSIPS has to do is route the packets to the right place.  In
networking OSI terms, think of it as a layer 5 switch.

  [1] http://www.vocal.com/sip-2/call-transferring/   (one such diagram)


- Jeff

On Fri, Nov 27, 2015 at 5:11 AM, Aqs Younas <aqsyounas at gmail.com> wrote:

> I don't thinks opensips can handle transfer. It just proxy invites and
> re-invites or may be someone more expert on mailing list can correct me.
> But you can achieve above by simply using freeswitch.
> On 26 Nov 2015 12:19, "Michele Pinassi" <michele.pinassi at unisi.it> wrote:
>
>> Hi all,
>>
>> i have a problem setting up a boss/secretary function with my OpenSIPS
>> router.
>>
>> Scenario:
>>
>> 5000 was the boss phone
>> 5002 was the secretary phone
>>
>> Only 5002 can call directly 5000 (the boss phone) and all call directed
>> to 5000 were diverted to 5002, that answer and decide if the call should
>> be transferred to the boss phone (5000) or not.
>>
>> Just imagine i'm 3000. And i call the 5000 (boss phone)
>>
>> 3000 ---> 5000 ---[DIVERT]---> 5002
>>
>> The secretary answer [5002] and call the boss ----> [5000] to ask
>> permission to forward the call. If the boss say YES, secretary TRASNFER
>> the call originated to 3000 (answered 5002) to 5000. And on OpenSIPS i
>> see the call originated from 3000 and directed to 5000, so DENIED.
>>
>> I try with brach flag, with AVP...no way, i cannot determine if a call
>> was firstly answered (and allowed) by 5002. Just for information, i'm
>> using SNOM 710/760 phones.
>>
>> Any hint/suggestions ?
>>
>> Thanks, Michele
>>
>> --
>> Michele Pinassi
>> Responsabile Telefonia di Ateneo
>> Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi
>> di Siena
>> tel: 0577.(23)5000 - centralino at unisi.it
>>
>> Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ
>> di Ateneo, http://www.faq.unisi.it
>>
>>
>>
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>
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