Hi Stefan and Bogdan,<div><br></div><div>Thanks for all the positive feedback, yes I thougt that 183 response and 404 not found on early announce or bye for announce application in SEMS would work. But it could also be used for service wide announcements and combined with callerid patch and database with language option... </div>
<div><br></div><div>Our concept is to make telecom free, so for internal calls, +883, ENUM and other voip networks there will not be an announcement We know that the promo announcement must be short. :) There are so many people willing to help out by donating a server. So we have 0 cost and only pay the actual minutes at increadible rates. If we could we'd let you dial everywhere in the world for free.</div>
<div><br></div><div>Just another question out of curiousity, in the OpenSIPS routing script, you can set sip messages "40X forbidden" etc. Could one also make this "Gandalf says: 40X forbidden"?<br> or would that really screw up any Linksys, Snom, Asterisk or other voip equipment?</div>
<div><br></div><div>Albert</div><div><br></div><div><br><div class="gmail_quote">On Tue, May 18, 2010 at 7:41 PM, Stefan Sayer <span dir="ltr"><<a href="mailto:stefan.sayer@googlemail.com">stefan.sayer@googlemail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi Bogdan,<br>
<div class="im"><br>
Bogdan-Andrei Iancu wrote:<br>
> Hi Stefan,<br>
><br>
> There is a built in functionality for this in OpenSIPS: see the<br>
> minor_branch_flag()<br>
> <a href="http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id271212" target="_blank">http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id271212</a><br>
><br>
> This can be used when you parallely fork a branch to a media server to<br>
> get media via 183 (like ring back tone), but you do not want the<br>
> transaction engine to wait for the completion of that branch (if all<br>
> other did end with negative answer).<br>
> Again this is mainly for parallel forking scenarios.<br>
</div>thanks for the pointer, thats interesting for RBT. It understood<br>
(possibly wrong) that the OP wanted to have his ads completed before<br>
the call continues. not that I would personally like it much to listen<br>
to long ads before the call, but if the ads are only played while its<br>
connecting/ringing, thats probably ok (for a free service). If I were<br>
the OP, I would send the call through SEMS B2BUA and mix the actual<br>
RBT audio from destination with the ad from DB, that way the caller<br>
knows what's happening with the call while listening to ad, and<br>
listens probably with much more attention.<br>
<br>
Regards<br>
<font color="#888888">Stefan<br>
</font><div><div></div><div class="h5"><br>
><br>
> Regards,<br>
> Bogdan<br>
><br>
> Stefan Sayer wrote:<br>
>> Albert Paijmans wrote:<br>
>><br>
>>> Hi Andreas,<br>
>>><br>
>>> Thanks for the reply. The reason we do not want to use Asterisk, but<br>
>>> SEMS, is because SEMS offers the possibility to play a different<br>
>>> announcement (could be from database) to every extension. This ofcourse<br>
>>> makes it more attractive to our sponsors. We want to do both sponsor<br>
>>> messages for outgoing calls and we will have some discreet advertisement<br>
>>> on our website. We think we can offer free phonecalls to most<br>
>>> international destinations thanks to Open Source and we are all<br>
>>> volunteers :)<br>
>>><br>
>>> So forwarding calls to Asterisk and using Asterisk as a media server for<br>
>>> voicemail or busy tones I understand that part. But how could I send<br>
>>> outgoing (pstn) calls to SEMS first and then to Asterisk? Is there<br>
>>> something like a service route for this?<br>
>>><br>
>> whether you are using SEMS or Asterisk for pre call/early media<br>
>> announcement, you would first send the call to the media server of<br>
>> your choice, have an announcement played with 183, then the media<br>
>> server replies with negative final reply, which you catch in your<br>
>> proxy and add as another branch the final destination (pstn/asterisk).<br>
>><br>
>> alternatively, you can send the call to SEMS, have the announcement<br>
>> played there in early media, and then continue the call in B2BUA mode<br>
>> through SEMS (see ann_b2b application, you can modify that a little to<br>
>> use 183 instead of 200; or use a simple DSM script and connectCallee<br>
>> action).<br>
>><br>
>> Regards<br>
>> Stefan<br>
>><br>
>><br>
>><br>
>>> Thanks<br>
>>><br>
>>> Albert<br>
>>><br>
>>><br>
>>><br>
>>> On Sat, May 15, 2010 at 2:06 AM, Andreas Sikkema <<a href="mailto:h323@ramdyne.nl">h323@ramdyne.nl</a><br>
>>> <mailto:<a href="mailto:h323@ramdyne.nl">h323@ramdyne.nl</a>>> wrote:<br>
>>><br>
>>> On May 14, 2010, at 11:13 PM, Albert Paijmans wrote:<br>
>>><br>
>>> > Is it possible to add an extra announcement server in the call path?<br>
>>> > So OpenSIPS acts as registrar/proxy, Asterisk does pstn,<br>
>>> voicemail etc. But on certain destinations the call is relayed<br>
>>> through an announcement server before continuing to Asterisk.<br>
>>><br>
>>> I'd just use the existing Asterisk for it (providing it has a<br>
>>> reliable timing source) and have it play a wav file during "ringing<br>
>>> phase" and after the WAV file ends do the rest of the dialplan and<br>
>>> have the outgoing call answer the incoming call.<br>
>>><br>
>>> This sudden influx of "let's do add before the call" business plans<br>
>>> of late really takes me back to my first VoIP operator job, they<br>
>>> just stopped doing that (in the Netherlands and Germany) because<br>
>>> there was no money around 2002 after the whole 9/11 thing when there<br>
>>> was an economic crisis and advertisers stopped advertising ;-)<br>
>>><br>
>>> I must be getting old....<br>
>>><br>
>>> --<br>
>>> Andreas<br>
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>>><br>
>>><br>
>>><br>
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>>><br>
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>>><br>
>><br>
>><br>
><br>
><br>
<br>
<br>
--<br>
</div></div><div class="im">Stefan Sayer<br>
VoIP Services Consulting and Development<br>
<br>
Warschauer Str. 24<br>
10243 Berlin<br>
<br>
tel:+491621366449<br>
<a href="mailto:sip%3Asayer@iptel.org">sip:sayer@iptel.org</a><br>
email/<a href="mailto:xmpp%3Astefan.sayer@gmail.com">xmpp:stefan.sayer@gmail.com</a><br>
<br>
<br>
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