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? 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