hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868)
but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set "prematuremedia" with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel <satish...@hotmail.com>: > 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 > -- _____________________________________________________________________ -- 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