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