[Freeswitch-users] Force SIP UA to pick up call during ringing?

Anthony Minessale anthony.minessale at gmail.com
Wed Jun 17 06:36:35 PDT 2009 

<clippy>Looks like you are trying to build a call center</clippy>

have you seen mod_fifo?
It's designed to let people on headsets sit idle and you can send calls to
them at will.


On Tue, Jun 16, 2009 at 3:11 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Thanks Michael,
>
> I have disabled it now.
>
> I finally got it to work, (sip_h_Call-Info=<sip:$${domain}>;answer-after=0)
> but the behaviour was not as desired, as I didn't manage the phone to
> pick up the call on the headset. It will only have the speaker enabled.
> So I will have to go a different way with parking the call and then
> forward it.
>
> Best regards
> Peter
>
>
> Michael Jerris schrieb:
> >   uuid_setvar <unique_id> sip_invite_params intercom=true should be
> > unnecessary.
> >
> > Mike
> >
> > On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote:
> >
> >
> >> It mainly works now by uuid_transfer the following way via event
> >> socket.
> >>      uuid_setvar <unique_id> sip_invite_params intercom=true
> >>      uuid_setvar <unique_id> sip_auto_answer true
> >>      uuid_transfer <unique_id> 1000 XML default
> >> so the call is transferred from 1000 to 1000.
> >>
> >> What happens:
> >> 1) If I disable intercom on the Snom phone, the phone rings, stops
> >> ringing and rings again (ok)
> >> 1) If I enable intercom on the Snom phone, the phone rings, stops
> >> ringing and hangs up (not ok)
> >>
> >> So I do not get the Snom to pick up the call in intercom mode.
> >>
> >> The last invite is:
> >>    INVITE sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib
> >> SIP/2.0
> >>    Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF
> >>    Route: <sip:1000 at 217.24.11.189:2752>;transport=tls;line=er6kxnib
> >>    Max-Forwards: 68
> >>    From: "Peter FS" <sip:723323 at 217.xx.xx.xxx>;tag=9eQ8rjQy533HF
> >>    To:
> >> <sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true
> >>
> >>    Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
> >>    CSeq: 116467629 INVITE
> >>    Contact: <sip:mod_sofia at 217.xx.xx.xxx:5061;transport=tls>
> >>    Call-Info: <sip:217.xx.xx.xxx>;answer-after=0
> >> The intercom part is there and the Call-Info line with answer-after
> >> also.
> >>
> >> The phone answers with
> >>    SIP/2.0 401 Unauthorized
> >>    Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF
> >>    From: "Peter FS" <sip:723323 at 217.xx.xx.xxx>;tag=9eQ8rjQy533HF
> >>    To:
> >> <sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true
> >>
> >>> ;tag=71rskygkr2
> >>>
> >>    Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
> >>    CSeq: 116467629 INVITE
> >>    Contact:
> >> <sip:1000 at 192.168.178.50:2752;transport=tls;line=er6kxnib>;reg-id=1
> >>    WWW-Authenticate: Digest realm="sip2.mycompany.de",
> >> nonce="2ee26efe6ab27f88", algorithm=MD5
> >>    Content-Length: 0
> >> and hangs up.
> >>
> >> Anybody know how to solve this Snom intercom issue?
> >>
> >> Best regards
> >> Peter
> >>
> >>
> >> Michael Jerris schrieb:
> >>
> >>> The transfer should work but it sounds like offhook agents is what
> >>> your really trying to accomplish so I would go with brian's
> >>> suggestion.
> >>>
> >>>
> >>>
> >>> On Jun 16, 2009, at 7:38 AM, Peter P GMX <Prometheus001 at gmx.net>
> >>> wrote:
> >>>
> >>>
> >>>
> >>>> Hello Michael,
> >>>>
> >>>> I want the phone be ringing, just for acoustical feedback reasons.
> >>>>
> >>>> But what if I
> >>>>
> >>>>   * transfer it to the same user destination again (now with
> >>>> intercom
> >>>>     enabled), will this work?
> >>>>   * transfer it to park and then transfer it to the same destination
> >>>>     again (now with intercom enabled)
> >>>>
> >>>> Best regards
> >>>> Peter
> >>>>
> >>>> Michael Jerris schrieb:
> >>>>
> >>>>
> >>>>> The only way I can think to do this today would be to cancel the
> >>>>> call
> >>>>> and re send with the intercom headers for a phone that supports it.
> >>>>> It may be possible to send a reinvite with autoanswer headers but I
> >>>>> doubt that would work, all you could do is try making code to do it
> >>>>> it
> >>>>> a sipp or sipsak scenario and test it.  A better aproach might be
> >>>>> to
> >>>>> answer the call normally and detect that to start your web workflow
> >>>>> or
> >>>>> not really ring the phone, just the web app and deliver the call
> >>>>> with
> >>>>> autoanswer when the button is hit in the web ui.
> >>>>>
> >>>>> Mike
> >>>>>
> >>>>> On Jun 16, 2009, at 4:24 AM, Peter P GMX <Prometheus001 at gmx.net>
> >>>>> wrote:
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>> Hello Brian,
> >>>>>>
> >>>>>> this is too easy :-).
> >>>>>>
> >>>>>> This is for a small callcenter app and I only want the user to
> >>>>>> pickup
> >>>>>> the call once (to accept the call in X-Lite (or a Snom phone)
> >>>>>> and to
> >>>>>> start the workflow on the web application). I do not want him to
> >>>>>> accept
> >>>>>> the call on the phone and then on the Web app.
> >>>>>>
> >>>>>> Best regards
> >>>>>> Peter
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> Brian West schrieb:
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>> click on the AA button?  :)
> >>>>>>>
> >>>>>>> /b
> >>>>>>>
> >>>>>>> On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote:
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>> What is the best way to have this done? Move the call to park
> >>>>>>>> and
> >>>>>>>> then
> >>>>>>>> retransfer again with intercom, or is there a better solution?
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>> _______________________________________________
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> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
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> >>>>>
> >>>>>
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-- 
Anthony Minessale II

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