[Asterisk-Users] No ringback over IAX - LiveVoip

[Brian Dingman]

Found this info on their website:

http://www.livevoip.com/index.php?subject=2&content=networkStatus

LiveVoip Operations Staff

----

DTMF - Ringback Issues

Currently, Asterisk is using the timing of the input stream to
reproduce the output stream. This means that when no RTP streams are
being sent from the peer Endpoint Gateway, Asterisk is unable to
generate audio. This approach or limitation leads to "one way speech"
conditions. Plus - Some devices don't generate audio until the answer
supervision is received from the called. For all these scenarios, no
ringback can be presented to the calling party. In cases where the
endpoints are using silence compression, the audio from asterisk is
chopped. Its fine if your run Asterisk with a T-1 Card, if not then
you are going to experience issues.

What Can or Should be Done?

To get this solved, Asterisk should obtain its clocking from an
internal source in a way that an output stream can be generated
without getting any RTP input. The clocking should then be taken from
an internal timing mechanism that keeps track of the synchronization.
The solution should not require T1 connectivity [IE: no TDM hardware].
Such T1 connectivity would severely limit traffic on the LiveVoip
Global SIP network via IP. Developers should work to solve the no
alerting scenario's [when peer is set in RCV only mode] and all issues
related to the use of silence compression. A configuration option
should exist to choose the timing method for customers that want to
use Asterisk in calling card applications or any application where no
T-1 cards will ever be required.

Status:

LiveVoip engineers have developed a workaround for our internal switch
network. This will be tested and could take up to 14 days to install
in every LiveVoip Network Node location.


On Tue, 15 Mar 2005 17:07:53 -0500, Robert Webb <asterisk at ropeguru.com> wrote:
> 
> On Tue, 15 Mar 2005 14:50:38 -0700
>   Daniel Webb <lists at danielwebb.us> wrote:
> > On Fri, Mar 11, 2005 at 11:50:01PM +0000, Jay Milk
> >wrote:
> >
> >> Dude, where have you been?  This has been discussed here
> >>at length.
> >> Everyone agrees that it's on LiveVOIP's end, but they're
> >>shrugging their
> >> shoulders and pointing toward *.  Search the list.
> >
> > Could you point out the best way to "search the list"?
> >
> > Perhaps go to
> >http://lists.digium.com/pipermail/asterisk-users/, go to
> > each month one at a time, then click "threads", then do
> >a page search?
> > What a swell interface.
> 
> How about learning a few Google skills and in the search
> line type:
> 
> site:lists.digium.com <search criteria>
> 
> THe above site command will only search the url specified.
> In this case the Asterisk lists.
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