[OpenSIPS-Users] OpenSIPS > announcement > pstn
Stefan Sayer
stefan.sayer at googlemail.com
Tue May 18 20:02:37 CEST 2010
David J. wrote:
> Stefan,
>
> I had a similar question, we don't play ads to the caller, but we do
> locate the callee while the caller is waiting or "on hold", so we could
> use the B2BUA functionality.
what does the desired call flow actually look like?
>
> Is there an example available of this functionality in the SEMS examples?
pre-call RBT is readily available (e.g. early_announce - early media
then final reply or early media then B2BUA) or a simple DSM script
(see e.g. early_media.dsm or test_b2b.dsm in doc/dsm/examples); also see
http://ftp.iptel.org/pub/sems/doc/current/AppDoc.html
Mixing destination RBT with announcement would need some hacking;
though in webconference application something similar is done, early
media from outgoing call mixed into conference, and you could add that
code to b2b_connect application. If you want to get destination RBT to
your media server, you need to send media server address in outgoing
INVITE - which means after the call is established you relay RTP
through your media server (at least until you do some reinvites do
drop out of the call, which can be a little complex and does not
necessarily work too well in all constellations).
>
> (Sorry to post a SEMS question on the OpenSIPs mailing list, I will
> gladly post to SEMS list if one exists.)
mailing lists are at http://lists.iptel.org/ (see http://iptel.org/sems).
Regards
Stefan
>
>
> Thanks.
>
>
>
> On 5/18/10 1:41 PM, Stefan Sayer wrote:
>> Hi Bogdan,
>>
>> Bogdan-Andrei Iancu wrote:
>>
>>> Hi Stefan,
>>>
>>> There is a built in functionality for this in OpenSIPS: see the
>>> minor_branch_flag()
>>> http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id271212
>>>
>>> This can be used when you parallely fork a branch to a media server to
>>> get media via 183 (like ring back tone), but you do not want the
>>> transaction engine to wait for the completion of that branch (if all
>>> other did end with negative answer).
>>> Again this is mainly for parallel forking scenarios.
>>>
>> thanks for the pointer, thats interesting for RBT. It understood
>> (possibly wrong) that the OP wanted to have his ads completed before
>> the call continues. not that I would personally like it much to listen
>> to long ads before the call, but if the ads are only played while its
>> connecting/ringing, thats probably ok (for a free service). If I were
>> the OP, I would send the call through SEMS B2BUA and mix the actual
>> RBT audio from destination with the ad from DB, that way the caller
>> knows what's happening with the call while listening to ad, and
>> listens probably with much more attention.
>>
>> Regards
>> Stefan
>>
>>
>>> Regards,
>>> Bogdan
>>>
>>> Stefan Sayer wrote:
>>>
>>>> Albert Paijmans wrote:
>>>>
>>>>
>>>>> Hi Andreas,
>>>>>
>>>>> Thanks for the reply. The reason we do not want to use Asterisk, but
>>>>> SEMS, is because SEMS offers the possibility to play a different
>>>>> announcement (could be from database) to every extension. This ofcourse
>>>>> makes it more attractive to our sponsors. We want to do both sponsor
>>>>> messages for outgoing calls and we will have some discreet advertisement
>>>>> on our website. We think we can offer free phonecalls to most
>>>>> international destinations thanks to Open Source and we are all
>>>>> volunteers :)
>>>>>
>>>>> So forwarding calls to Asterisk and using Asterisk as a media server for
>>>>> voicemail or busy tones I understand that part. But how could I send
>>>>> outgoing (pstn) calls to SEMS first and then to Asterisk? Is there
>>>>> something like a service route for this?
>>>>>
>>>>>
>>>> whether you are using SEMS or Asterisk for pre call/early media
>>>> announcement, you would first send the call to the media server of
>>>> your choice, have an announcement played with 183, then the media
>>>> server replies with negative final reply, which you catch in your
>>>> proxy and add as another branch the final destination (pstn/asterisk).
>>>>
>>>> alternatively, you can send the call to SEMS, have the announcement
>>>> played there in early media, and then continue the call in B2BUA mode
>>>> through SEMS (see ann_b2b application, you can modify that a little to
>>>> use 183 instead of 200; or use a simple DSM script and connectCallee
>>>> action).
>>>>
>>>> Regards
>>>> Stefan
>>>>
>>>>
>>>>
>>>>
>>>>> Thanks
>>>>>
>>>>> Albert
>>>>>
>>>>>
>>>>>
>>>>> On Sat, May 15, 2010 at 2:06 AM, Andreas Sikkema<h323 at ramdyne.nl
>>>>> <mailto:h323 at ramdyne.nl>> wrote:
>>>>>
>>>>> On May 14, 2010, at 11:13 PM, Albert Paijmans wrote:
>>>>>
>>>>> > Is it possible to add an extra announcement server in the call path?
>>>>> > So OpenSIPS acts as registrar/proxy, Asterisk does pstn,
>>>>> voicemail etc. But on certain destinations the call is relayed
>>>>> through an announcement server before continuing to Asterisk.
>>>>>
>>>>> I'd just use the existing Asterisk for it (providing it has a
>>>>> reliable timing source) and have it play a wav file during "ringing
>>>>> phase" and after the WAV file ends do the rest of the dialplan and
>>>>> have the outgoing call answer the incoming call.
>>>>>
>>>>> This sudden influx of "let's do add before the call" business plans
>>>>> of late really takes me back to my first VoIP operator job, they
>>>>> just stopped doing that (in the Netherlands and Germany) because
>>>>> there was no money around 2002 after the whole 9/11 thing when there
>>>>> was an economic crisis and advertisers stopped advertising ;-)
>>>>>
>>>>> I must be getting old....
>>>>>
>>>>> --
>>>>> Andreas
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org<mailto:Users at lists.opensips.org>
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>
>>>
>>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
--
Stefan Sayer
VoIP Services Consulting and Development
Warschauer Str. 24
10243 Berlin
tel:+491621366449
sip:sayer at iptel.org
email/xmpp:stefan.sayer at gmail.com
More information about the Users
mailing list