[asterisk-ss7] No Audio on SS7 calls to Remote PRIs

[Stephan Ellis]

Switching to 1.6.0 did the trick.  I tried to run 1.6.0.28 but I had the no
audio issue.  I'm not sure what you mean by there is no 1.6.0. I found it
here:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-addons-1.6.0.tar.gz

You're welcome to get with me out of band to see my specific setup.  I am
willing to post the traffic of the failing call with my asterisk 1.6.2 stack
if anyone is interested.

-stephan

On Tue, Oct 5, 2010 at 9:20 AM, Kevin P. Fleming <kpfleming at digium.com>wrote:

> On 10/05/2010 09:02 AM, Stephan Ellis wrote:
> > Any specific point version of 1.6.0 i should use? Or just 1.6.0?
>
> If the underlying problem is, as your switch technician suggests, lack
> of response to a particular SS7 message, then going back to an older
> version of Asterisk is not going to help. I know the author of the
> previous reply was trying to be helpful, but he posts the identical
> response to every thread where people are having issues with Asterisk
> and SS7... that does not mean he's actually analyzed the problem and
> knows that it will be resolved by using 1.6.0.x (there is no "1.6.0").
>
> Since you've determined that the problem only occurs when your Asterisk
> box is placing calls to specific remote destinations through your SS7
> switch, have you tried any other SS7 clients off that switch calling the
> same destinations? I know you've mentioned that you have an additional
> Asterisk box using ISDN PRI to that switch, but since that's a different
> protocol it's not really going to help (except to verify that your SS7
> switch does have a functional audio path between it and the remote
> destination).
>
> Most likely the message(s) involved here are related to some sort of
> SS7/ISDN (or SS7/PSTN) interworking, and chan_dahdi/libss7 just haven't
> taken those into account yet. In order to be able to debug the issue,
> you're going to have to post a debug log that shows the SS7 traffic
> being sent and received for this failing call, so that people who
> understand the protocol can try to figure out what is going wrong.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kfleming at digium.com
> Check us out at www.digium.com & www.asterisk.org
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20101005/bceb10c9/attachment-0001.htm