hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd
2011/5/13 d tbsky <[email protected]>: > 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 <[email protected]>: >> 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 <[email protected]>: >>> 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: [email protected] >>>> To: [email protected]; [email protected] >>>> >>>> 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 <[email protected]>: >>>> > 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: [email protected] >>>> >> To: [email protected] >>>> >> Subject: Re: [asterisk-users] 1.8 and prematuremedia problem >>>> >> Date: Tue, 10 May 2011 07:43:47 -0400 >>>> >> CC: [email protected] >>>> >> >>>> >> 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 <[email protected]> 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 <[email protected]>: >>>> >> >> 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 <[email protected]>: >>>> >> >>> 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 <[email protected]> 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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