hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help!
Regards, tbskyd 2011/5/11 d tbsky <tbs...@gmail.com>: > hi: > ok I will create a bug report. and I found I still need > "prematuremedia=no" in asterisk 1.6.2.18. > yesterday I was testing at home with zoiper softphone + iax. today I > test snom hardware sip phone and found that "prematuremedia=no" is > still necessary. > > Regards, > tbskyd > > > 2011/5/11 satish patel <satish...@hotmail.com>: >> I am sorry about that but its interesting it doesn't work with 1.8 SVN >> >> I would say please report this bug so that way you can track issue, And may >> be in future it help us :) >> >> -S >> >>> Date: Wed, 11 May 2011 01:31:34 +0800 >>> Subject: Re: [asterisk-users] 1.8 and prematuremedia problem >>> From: tbs...@gmail.com >>> To: asterisk-users@lists.digium.com; satish...@hotmail.com >>> >>> 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