[asterisk-dev] Bounty

[Geoffrey Mina]

Actually, you wouldn't stop using asyterisk, you would simply insert
the OpenSER/Kamailio server in the middle of the chain.

Asterisk --> kamailio --> itsp

Basically asterisk would still do the transcoding.  Kamailio would be
responsible for the least cost routing and automated failover when you
don't get a 1XX response.

As I said before, let asterisk do what it's good at: transcoding,
media processing, bridging, etc.  Let kamailio do what it's good at:
sip routing.

Or you can continue fighting with 1.6 bugs :)

On 2/26/10, CDR <venefax at gmail.com> wrote:
> Asterisk is unique in the sense that you get a G723 call and you may connect
> it out with Ulaw provider, a G729 carrier, and the reverse is also true.
> There are many technologies that deal only with signaling, but I have not
> found a clear alternative to Asterisk when you consider the codecs and
> transcoding issues. Also I get one call in, and start dialing all the
> carriers on the order of price, from cheapest to the most expensive. So I
> need a complete separation of the inbound channels and the outbound channel.
> Maybe somebody can recommend how to do these tricks away from Asterisk. I
> don't think it is possible.
> Version 1.6X with a new feature found in Trunk (Q850 relay on the BYE) make
> Asterisk into a Class 5. If we can solve this bug that I reported several
> months ago.
>
> Yours
>
> Federico
>
> On Thu, Feb 25, 2010 at 9:44 PM, Geoffrey Mina
> <geoffreymina at gmail.com>wrote:
>
>> Why not route your calls through an OpenSIPS or Kamailio server?
>> These both have the ability to fail over very quickly if no
>> provisional 1XX response is received... Plus you wouldn't have to
>> build failover into asterisk.  Which is a good thing as SIP failover
>> has no place in the dialplan.
>>
>> On 2/25/10, Olle E. Johansson <oej at edvina.net> wrote:
>> >
>> > 25 feb 2010 kl. 17.46 skrev Kevin P. Fleming:
>> >
>> >> Tim Ringenbach wrote:
>> >>> If "first ring" is what you want, you would probably not want to count
>> >>> 100's, only 183 or 180. For example. OpenSips automatically sends back
>> a
>> >>> 100 trying when you sent it a call, before it even passes the packet
>> >>> on
>> >>> to it's destination. It then absorbs the 100 Trying it gets from the
>> far
>> >>> end.
>> >>
>> >> All SIP UAS endpoints must send 100 Trying in response to an INVITE as
>> >> quickly as they can do so (and for other sorts of requests as well).
>> >> Asterisk does this now.
>> >
>> >
>> > All proxys has to send a 100 trying to say "I've got it and I'm trying
>> > to
>> > proxy this message through". At that point, we stop retransmitting
>> > stuff,
>> > since we've got someone acting on our behalf. That's why the proxy send
>> 100
>> > trying by itself and swallow the one received from the other side, if
>> > received.
>> >
>> > /O
>> >
>> >
>> >
>> >
>> > --
>> > _____________________________________________________________________
>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> >
>> > asterisk-dev mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >    http://lists.digium.com/mailman/listinfo/asterisk-dev
>> >
>>
>> --
>> Sent from my mobile device
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-dev mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
>

-- 
Sent from my mobile device