Debian 7 is currently in the `freeze' status with 1.8.13 - that means Debian 7 is very likely to release 1.8.13 and be carrying it for the next 2-3 years (typical lifetime of a Debian release) I run 1.8.8. TLS has a bug: it fails to receive BYE over the TLS connection from my Polycom phone. I tried 1.8.13, the version in Debian 7, and found a more severe bug: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=683956 The TLS clients can't connect at all, this looks like a really bad regression from 1.8.8 I've looked at 1.8.(14, 15, 16-rc1) and their changelogs don't mention any fix. Debian is very conservative about accepting updates during the `freeze' process - they will most likely want to see a 1.8.13.2 release with ONLY the most essential fixes a) is anyone else aware of these bugs? b) what essential changes should go into 1.8.13.2 for Debian?
On 12-08-10 04:47 PM, Daniel Pocock wrote:> > > Debian 7 is currently in the `freeze' status with 1.8.13 - that means > Debian 7 is very likely to release 1.8.13 and be carrying it for the > next 2-3 years (typical lifetime of a Debian release) > > I run 1.8.8. TLS has a bug: it fails to receive BYE over the TLS > connection from my Polycom phone. > > I tried 1.8.13, the version in Debian 7, and found a more severe bug: > http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=683956 > The TLS clients can't connect at all, this looks like a really bad > regression from 1.8.8 > > I've looked at 1.8.(14, 15, 16-rc1) and their changelogs don't mention > any fix. > > Debian is very conservative about accepting updates during the `freeze' > process - they will most likely want to see a 1.8.13.2 release with ONLY > the most essential fixes > > a) is anyone else aware of these bugs? > > b) what essential changes should go into 1.8.13.2 for Debian? >We don't need to release a 1.8.13.2 release of Asterisk. Once the issue has been fixed in the 1.8 release branch, it would just be back-ported into a Debian patch for the package. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
Hi Team, I use Asterix 1.6.2.9-2 what is running on debian squeeze. I completely statisfied this software. I did everything I want so far. I love it so much, but there is a point where I can not step through. 1) I have connected to my telephone provider as a SIP client, but my Asterisk only one call make to the world in same time. My provider does not limit the number of simultaneous calls. The only limit is the bandwidth of my local internet link. How can I configure my asterisk to create more than one simultaneous calls through my provider? 2) If I use an ATA, which has 2 SIP clients. These SIP clients is the same asterisk user, but asterisk register only the last one. May I got chance for registering ATA with the same users in the asterisk or every ATA must have two different asterisk user for working well? Thanks for any hints in advance! Best regards, Gabor Hatos
2012/8/11 Hatos Gabor <hatos at ggki.hu>> > Hi Team, > > I use Asterix 1.6.2.9-2 what is running on debian squeeze. I completely > statisfied this software. I did everything I want so far. I love it so > much, but there is a point where I can not step through. > > 1) > I have connected to my telephone provider as a SIP client, but my Asterisk > only one call make to the world in same time. My provider does not limit > the number of simultaneous calls. The only limit is the bandwidth of my > local internet link. How can I configure my asterisk to create more than > one simultaneous calls through my provider? >Asterisk has no limitation on the number of simultaneous calls. Just place another call while one call is already going...> > 2) > If I use an ATA, which has 2 SIP clients. These SIP clients is the same > asterisk user, but asterisk register only the last one. May I got chance > for registering ATA with the same users in the asterisk or every ATA must > have two different asterisk user for working well? >Ata I have found so far allows to set two distinct SIP account for each one of the FXS/FXO ports they have. Leandro> > Thanks for any hints in advance! > > Best regards, > Gabor Hatos > > > > > -- > ______________________________**______________________________**_________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120811/1f4da2bc/attachment.htm>
Dear Leonardo, Thank your for your answers! The problem was at the provider! Best Regards, Gabor Hatos Leandro Dardini wrote:> > > 2012/8/11 Hatos Gabor <hatos at ggki.hu <mailto:hatos at ggki.hu>> > > > Hi Team, > > I use Asterix 1.6.2.9-2 what is running on debian squeeze. I > completely statisfied this software. I did everything I want so > far. I love it so much, but there is a point where I can not step > through. > > 1) > I have connected to my telephone provider as a SIP client, but my > Asterisk only one call make to the world in same time. My provider > does not limit the number of simultaneous calls. The only limit is > the bandwidth of my local internet link. How can I configure my > asterisk to create more than one simultaneous calls through my > provider? > > > Asterisk has no limitation on the number of simultaneous calls. Just > place another call while one call is already going... > > > > 2) > If I use an ATA, which has 2 SIP clients. These SIP clients is the > same asterisk user, but asterisk register only the last one. May I > got chance for registering ATA with the same users in the asterisk > or every ATA must have two different asterisk user for working well? > > > > Ata I have found so far allows to set two distinct SIP account for > each one of the FXS/FXO ports they have. > > Leandro > > > > Thanks for any hints in advance! > > Best regards, > Gabor Hatos > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ------------------------------------------------------------------------ > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users