Sorry, the link to Asterisk 1.6.0 is: http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.0.tar.gz -stephan On Tue, Oct 5, 2010 at 9:46 AM, Stephan Ellis <stephan.ellis at gmail.com>wrote: > 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/20101005/0194a5be/attachment.htm
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