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
|