<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 <prometheus...@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:1...@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:1...@217.24.11.189:2752>;transport=tls;line=er6kxnib > >> Max-Forwards: 68 > >> From: "Peter FS" <sip:723...@217.xx.xx.xxx>;tag=9eQ8rjQy533HF > >> To: > >> <sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true > >> > >> Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e > >> CSeq: 116467629 INVITE > >> Contact: <sip:mod_so...@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:723...@217.xx.xx.xxx>;tag=9eQ8rjQy533HF > >> To: > >> <sip:1...@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:1...@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 <prometheus...@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 <prometheus...@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? > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>> _______________________________________________ > >>>>>>> Freeswitch-users mailing list > >>>>>>> Freeswitch-users@lists.freeswitch.org > >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>>>> http://www.freeswitch.org > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>> _______________________________________________ > >>>>>> Freeswitch-users mailing list > >>>>>> Freeswitch-users@lists.freeswitch.org > >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>>> http://www.freeswitch.org > >>>>>> > >>>>>> > >>>>>> > >>>>> _______________________________________________ > >>>>> Freeswitch-users mailing list > >>>>> Freeswitch-users@lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> > >>>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users@lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users@lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users@lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users@lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com <msn%3aanthony_miness...@hotmail.com> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org> iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org> pstn:213-799-1400
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