[Asterisk-Users] 183 Session in Progress

[Stewart Nelson]

Hi Charles,

Perhaps I misunderstood your situation.

I thought that your original problem was: Calls from a SIP phone
through Asterisk to an ATA or a 5300 did not provide audible
ringing to the caller when the called party was being alerted.
I also assumed that some calls made through the 5300 were destined
for the PSTN.

If the above is true, then converting 183->180 is fine for the ATAs,
but has problems on PSTN calls.  That's because there are many
situations where there is no answer supervision, but there is audio
information that the caller should hear.  Common examples include
"number is not in service", "number has been changed", "the customer
you're trying to reach is out of range", and similar announcements.

If such announcements are now being heard on the calling SIP phones,
then the 5300 must be sending 200 (Connect) in those cases.  That
may also be a problem, because the caller would be billed for calls
that did not really complete.

Also note that in many situations, if the called PSTN party is busy,
only an audio indication is available.  Try some busy numbers via
your various carriers to see if it works properly.

Even if the 180 is sent only when the called phone is really ringing,
there still might be some confusion on international calls.  When I
call e.g. France, I expect to hear a European ring tone.  If an
American tone plays instead, I would assume that the call was
misrouted, hang up, and try again.

Sorry that I don't know why the suggested fixes didn't work.  It
should not be difficult to use Ethereal to see where the audio
is getting lost.

Regards,

Stewart

----- Original Message ----- 
From: <charles at fmctel.com>
To: <asterisk-users at lists.digium.com>
Cc: "Stewart Nelson" <sn at scgroup.com>
Sent: Monday, June 21, 2004 11:13 PM
Subject: Re: [Asterisk-Users] 183 Session in Progress


> Hi Stewart, I've tryed all the options for connectionmode, also the
> connection progress in cisco, but it didn't work. The way that I solved it
> was changing the asterisk source to instead of sending Session Progress
> when the call is in progress to send Ringing. Because for the users "In
> progress" means "Ringing". I've tested it and it works well. This will be
> temporary until we take out the ciscos and do Radius and T1 in our
> asterisk.
> 
> thank you
> Charles Rauber Gomes
> 
> 
> > Hi Charles,
> >
> > Blocking the 183 is undesirable, because messages
> > from the PSTN indicating that e.g. a number has been changed,
> > will be lost.  Instead, do what's necessary to get audio
> > back to the caller.  On the ATA, set bit 19 of ConnectMode
> > (see table 5-8 of manual).  On the 5300, see
> > http://www.ciscopress.com/content/images/1587050757/tips/troubleshooting_tips.doc
> > and look for "No Ringback on an IP Phone When Calling the PSTN" .
> >
> > --Stewart
> >
> >> Date: Fri, 18 Jun 2004 16:30:41 -0400 (EDT)
> >> Subject: Re: [Asterisk-Users] 183 Session in Progress
> >> From: charles at fmctel.com
> >> To: asterisk-users at lists.digium.com
> >> Reply-To: asterisk-users at lists.digium.com
> >>
> >> I've the same problem with the Cisco ATA's  and Cisco 5300. The cisco
> >> sends the: "SIP/SDP Status: 183 Session Progress  , with session
> >> description", asterisk forwards is to the phone: "SIP/SDP Status: 183
> >> Session Progress, with session description" after that the SIP Phone
> >> stops
> >> ringing.
> >> People complains because that Dead Signal while they wait the call to be
> >> completed, but I don't know what to do, if it's possible to stop
> >> asterisk
> >> forwarding this, or stops cisco sending this.
> >>
> >>
> >> Thank you
> >
> >
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>