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_lx at 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: tbskyd at gmail.com
>> To: asterisk-users at lists.digium.com; satish_lx at 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_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!!
>> >> >>>>
>> >> >>>> --
>> >> >>>>
>> >> >>>>
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