[asterisk-users] 1.8 and prematuremedia problem

[Satish Patel]

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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>>>
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>>
>