[OpenSIPS-Users] OpenSIPS > announcement > pstn

Stefan Sayer stefan.sayer at googlemail.com
Tue May 18 19:41:33 CEST 2010


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
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
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>>> Users at lists.opensips.org
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>>>     
>>
>>   
> 
> 


-- 
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





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