Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck
> From: satish...@hotmail.com > To: tbs...@gmail.com > Subject: Re: [asterisk-users] 1.8 and prematuremedia problem > Date: Tue, 10 May 2011 07:43:47 -0400 > CC: asterisk-users@lists.digium.com > > I have applied this patch in 1.8 svn branch and it works great for me. > > I have nothing special configuration just simple dial command for > outgoing call. > > Also check there are progress=yes option in chan_dahdi > > -- > Sent from my iPhone > > On May 10, 2011, at 5:58 AM, d tbsky <tbs...@gmail.com> wrote: > > > hi: > > I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not > > apply to 1.8.3.2 or 1.8.4-rc3). > > but the situation is the same. do I need to play with other options > > with the patch? or I need > > newer asterisk versions to solve the problem? > > thanks a lot for information!! > > > > 2011/5/10 d tbsky <tbs...@gmail.com>: > >> hi: > >> thanks a lot for your quick reply. I saw that patch and think that > >> it was already included in 1.8.3. > >> now I know it will be included in 1.8.5. > >> I will try it and thanks again for your kindly help!! > >> > >> 2011/5/10 Satish Patel <satish...@hotmail.com>: > >>> Apply this patch https://issues.asterisk.org/view.php?id=18868 > >>> > >>> -- > >>> Sent from my iPhone > >>> > >>> On May 9, 2011, at 9:57 PM, d tbsky <tbs...@gmail.com> wrote: > >>> > >>>> hi: > >>>> our current connection is below: > >>>> > >>>> sip phone<--->asterisk<---->alcatel PBX<---->PSTN > >>>> > >>>> asterisk and alcatel PBX is connected via E1 isdn-pri. > >>>> > >>>> when I use sip phone to dial outside PSTN world: > >>>> 1. with 1.4 it is fine. > >>>> 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or > >>>> sip > >>>> phone can not hear the ring and the beginning of the PSTN voice. > >>>> 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN > >>>> voice. I try to play options with "prematuremedia" and > >>>> "progressinband". but I can not find working settings. > >>>> > >>>> I don't know what other options I can try. > >>>> thank a lot for information!! > >>>> > >>>> -- > >>>> _____________________________________________________________________ > > > >>>> -- 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 > >>> > >> > >
-- _____________________________________________________________________ -- 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