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
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On May 10, 2011, at 5:58 AM, d tbsky <[email protected]> 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 <[email protected]>:
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 <[email protected]>:
Apply this patch https://issues.asterisk.org/view.php?id=18868
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On May 9, 2011, at 9:57 PM, d tbsky <[email protected]> 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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