Israel Gottlieb
2016-Aug-23 18:53 UTC
[asterisk-users] Dial and start music on hold after timeout
Maybe try progress() instead of answer () ?????? 23 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???:> Thank you, I just tried your suggestion. Strangely, the announcement is > played only if I try to dial a SIP peer which is not available (not > registered to be more precise). If the SIP peer is available, I only get > the ring tone, and never hear the announcement. Here is the dialplan (I had > to add an Answer() before the Dial, otherwise the announcement is never > played, even in the first case) : > > exten = 007,1,Answer() > same = n,Dial(SIP/foo&Local/s at playme,40) > > [playme] > exten = s,1,Ringing() > same = n,Wait(10) > same = n,Playback(/var/lib/asterisk/sounddir/announce,noanswer) > When it is working, I can see the following output in the CLI, which is > not there otherwise : > -- SIP/xxxxxxxxx requested media update control 26, passing it to > Local/s at playme-000005be;1 > > Otherwise, no error message, Asterisk tells he is playing the announcement > but I don't hear it. > > Best regards > > Jean Aunis > > Le 23/08/2016 ? 16:07, David Duffett a ?crit : > > How about: > > exten => s,1,Dial(SIP/alice&LOCAL/555 at delayed-announce,40) > > [delayed-announce] > exten => 555,1,Wait(20) > same => n,Playback(myannouncement,noanswer) > same => n,NoOP(Whatever else you want to do goes here) > > The 'noanswer' option on the Playback means that SIP/alice should continue > to ring for the remaining 20 of the 40 seconds, as the Playback will not > answer (terminate) the call. > > Don't forget AstriCon this year - www.astricon.net > > On 23 August 2016 at 12:52, Israel Gottlieb <isrlgb at gmail.com> wrote: > >> You could m and make a moh file that has ringing the first 30 sec and >> then the anouncment >> >> ?????? 22 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???: >> >> Thank you for the idea. The problem with RetryDial, is that it will >>> cancel the first call, play the announce and then dial the SIP peer once >>> again, so the telephone will display a missed call. I would prefer to do >>> everything in a single call. >>> >>> Le 22/08/2016 ? 17:57, John Kiniston a ?crit : >>> >>> You could try using RetryDial() instead of Dial, It supports playing an >>> announcement. >>> >>> >>> On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis <jean.aunis at prescom.fr> >>> wrote: >>> >>>> Sorry, I forgot to write that the SIP peer must keep ringing while the >>>> announcement is being played. >>>> >>>> Le 22/08/2016 ? 17:42, John Kiniston a ?crit : >>>> >>>> This seems like the obvious answer but maybe I'm misunderstanding the >>>> question. >>>> >>>> exten => s,1,Dial(SIP/alice,20) >>>> same => n,Playback(myannouncement) >>>> same => n,NoOP(Whatever else you want to do goes here) >>>> >>>> On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis <jean.aunis at prescom.fr> >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> I am searching a way to dial a SIP peer, and if it does not answer >>>>> within 20 seconds, play an announcement to the caller. This means that the >>>>> caller would hear a ring tone for 20 seconds, and only then hear the >>>>> announcement if the callee did not answer. >>>>> >>>>> I know it is possible to do this with ARI, but in this particular case >>>>> I do not want to use ARI. I would like to do this purely with dialplan and >>>>> AGI scripts, but I cannot find a way. I have read about the "m" option of >>>>> Dial application, but it starts the announcement immediately, whereas I >>>>> would like to start it after 20 seconds of timeout. >>>>> >>>>> Does anybody have an idea ? >>>>> >>>>> Best regards, >>>>> >>>>> Jean Aunis >>>>> >>>>> >>>>> -- >>>>> _____________________________________________________________________ >>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>>>> http://www.asterisk.org/hello >>>>> >>>>> asterisk-users mailing list >>>>> To UNSUBSCRIBE or update options visit: >>>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>>> >>>> >>>> >>>> >>>> -- >>>> A human being should be able to change a diaper, plan an invasion, >>>> butcher a hog, conn a ship, design a building, write a sonnet, balance >>>> accounts, build a wall, set a bone, comfort the dying, take orders, give >>>> orders, cooperate, act alone, solve equations, analyze a new problem, pitch >>>> manure, program a computer, cook a tasty meal, fight efficiently, die >>>> gallantly. Specialization is for insects. >>>> ---Heinlein >>>> >>>> >>>> >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>>> http://www.asterisk.org/hello >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> >>> >>> >>> -- >>> A human being should be able to change a diaper, plan an invasion, >>> butcher a hog, conn a ship, design a building, write a sonnet, balance >>> accounts, build a wall, set a bone, comfort the dying, take orders, give >>> orders, cooperate, act alone, solve equations, analyze a new problem, pitch >>> manure, program a computer, cook a tasty meal, fight efficiently, die >>> gallantly. Specialization is for insects. >>> ---Heinlein >>> >>> >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 >> http://www.asterisk.org/community/astricon-user-conference >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > [image: Digium logo] > *David Duffett* > Digium, Inc. ? Director, Worldwide Asterisk Community > 6 Landscape Close ? Weston on the Green ? Bicester ? Oxfordshire OX25 3SX > ? UK > direct/fax: +1 256 428 6119 ? mobile: +44 7722 442236 > twitter: dduffett ? linkedin: www.linkedin.com/in/davidduffett > Check us out at: http://digium.com ? http://asterisk.org > <http://www.asterisk.org/> > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 > http://www.asterisk.org/community/astricon-user-conference > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160823/f969ff9e/attachment.html>
Jean Aunis
2016-Aug-24 07:27 UTC
[asterisk-users] Dial and start music on hold after timeout
Using Progress didn't solve the problem. If I cannot find another way, I will use your solution of recording the ring tone. Le 23/08/2016 ? 20:53, Israel Gottlieb a ?crit :> > Maybe try progress() instead of answer () > > > ?????? 23 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr > <mailto:jean.aunis at prescom.fr>> ???: > > Thank you, I just tried your suggestion. Strangely, the > announcement is played only if I try to dial a SIP peer which is > not available (not registered to be more precise). If the SIP peer > is available, I only get the ring tone, and never hear the > announcement. Here is the dialplan (I had to add an Answer() > before the Dial, otherwise the announcement is never played, even > in the first case) : > > exten = 007,1,Answer() > same = n,Dial(SIP/foo&Local/s at playme,40) > > [playme] > exten = s,1,Ringing() > same = n,Wait(10) > same = n,Playback(/var/lib/asterisk/sounddir/announce,noanswer) > > When it is working, I can see the following output in the CLI, > which is not there otherwise : > -- SIP/xxxxxxxxx requested media update control 26, passing it to > Local/s at playme-000005be;1 > > Otherwise, no error message, Asterisk tells he is playing the > announcement but I don't hear it. > > Best regards > > Jean Aunis >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160824/e8568cca/attachment.html>
David Duffett
2016-Aug-24 07:52 UTC
[asterisk-users] Dial and start music on hold after timeout
Yes, sorry, my idea was too simplistic, as it did not take into account that the caller would already be hearing the ringing and therefore the announcement would need to be somehow mixed with that ringing... On 24 August 2016 at 08:27, Jean Aunis <jean.aunis at prescom.fr> wrote:> Using Progress didn't solve the problem. If I cannot find another way, I > will use your solution of recording the ring tone. > > Le 23/08/2016 ? 20:53, Israel Gottlieb a ?crit : > > Maybe try progress() instead of answer () > > ?????? 23 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???: > >> Thank you, I just tried your suggestion. Strangely, the announcement is >> played only if I try to dial a SIP peer which is not available (not >> registered to be more precise). If the SIP peer is available, I only get >> the ring tone, and never hear the announcement. Here is the dialplan (I had >> to add an Answer() before the Dial, otherwise the announcement is never >> played, even in the first case) : >> >> exten = 007,1,Answer() >> same = n,Dial(SIP/foo&Local/s at playme,40) >> >> [playme] >> exten = s,1,Ringing() >> same = n,Wait(10) >> same = n,Playback(/var/lib/asterisk/sounddir/announce,noanswer) >> When it is working, I can see the following output in the CLI, which is >> not there otherwise : >> -- SIP/xxxxxxxxx requested media update control 26, passing it to >> Local/s at playme-000005be;1 >> >> Otherwise, no error message, Asterisk tells he is playing the >> announcement but I don't hear it. >> >> Best regards >> >> Jean Aunis >> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 > http://www.asterisk.org/community/astricon-user-conference > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- [image: Digium logo] *David Duffett* Digium, Inc. ? Director, Worldwide Asterisk Community 6 Landscape Close ? Weston on the Green ? Bicester ? Oxfordshire OX25 3SX ? UK direct/fax: +1 256 428 6119 ? mobile: +44 7722 442236 twitter: dduffett ? linkedin: www.linkedin.com/in/davidduffett Check us out at: http://digium.com ? http://asterisk.org <http://www.asterisk.org/> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160824/90599ce6/attachment-0001.html>