[asterisk-users] 1.8 and prematuremedia problem
d tbsky
tbskyd at gmail.com
Thu May 12 20:31:36 CDT 2011
hi:
sorry. the issue number is 19268. not 19628.
sorry about that!!
Regards,
tbskyd
2011/5/13 d tbsky <tbskyd at gmail.com>:
> 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 <tbskyd at gmail.com>:
>> 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!!
>>>> >> >>>>
>>>> >> >>>> --
>>>> >> >>>>
>>>> >> >>>> _____________________________________________________________________
>>>> >>
>>>> >>
>>>> >> >>>> -- 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
>>>> >
>>>
>>
>
More information about the asterisk-users
mailing list