[asterisk-users] 1.8 and prematuremedia problem

[d tbsky]

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_lx at 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_lx at hotmail.com
>> To: tbskyd at gmail.com
>> Subject: Re: [asterisk-users] 1.8 and prematuremedia problem
>> Date: Tue, 10 May 2011 07:43:47 -0400
>> CC: asterisk-users at 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 <tbskyd at 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 <tbskyd at 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_lx at 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 <tbskyd at 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
>