2014-09-26 3:34 GMT+08:00 Eric Wieling :
> You will find not transcoding much less useful that one might imagine.
>
hi:
can you give some more hint about the topic?
in my testing, if the sip phone use G.722 and the sip trunk use G.711,
I can hear the quality is not as good as both side use G.7
-Commercial Discussion
Subject: Re: [asterisk-users] Change codec when dial from SIP to DAHDI
2014-09-25 20:46 GMT+08:00 Matthew Jordan :
>> https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
>
> That article is in the development section of the wiki. While that
> d
2014-09-25 20:46 GMT+08:00 Matthew Jordan :
>> https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
>
> That article is in the development section of the wiki. While that
> doesn't mean any of the information there is necessarily wrong, its
> purpose was to coordinate development eff
On Wed, Sep 24, 2014 at 10:20 PM, d tbsky wrote:
> hi:
> I Have tried asterisk 1.6.2, 1.8, 11, 12, 13. all versions behave
> the same => transcode in the middle even two legs use the same code.
>
> but I found an article which seems to solve this kind of problem:
>
> https://wiki.asterisk
hi:
forgot to mention. not only dialout DAHDI, even I dialout SIP
TRUNK, the situation is the same:
asterisk transcode in the middle even two legs use the same code.
2014-09-25 11:20 GMT+08:00 d tbsky :
> hi:
> I Have tried asterisk 1.6.2, 1.8, 11, 12, 13. all versions behave
> the
hi:
I Have tried asterisk 1.6.2, 1.8, 11, 12, 13. all versions behave
the same => transcode in the middle even two legs use the same code.
but I found an article which seems to solve this kind of problem:
https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
but I tried ver
Hi:
I am useing asterisk 11.12.
I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI
use alaw. G722 is great when ip-phone talks to each other. but when
ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to
transcode to alaw.
so I try to change the codec
hello,
I would like to know if it's possible to change the codec of a call in
the middle of the call.
I have an asterisk without g729 codecs and I recieve an incoming call.
The codec is negotiated in ulaw althought who is calling have a g729
codec. My * plays and announcements and call and extens
NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Arun Kumar
*Sent:* Tuesday, May 01, 2007 9:24 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Change Codec
Hi
I've inst
NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Arun Kumar
*Sent:* Tuesday, May 01, 2007 9:24 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Change Codec
Hi
I've inst
PROTECTED] On Behalf Of Arun Kumar
Sent: Tuesday, May 01, 2007 9:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Change Codec
Hi
I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've
allowed ulaw and g729. I want to change the
Hi
I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've
allowed ulaw and g729. I want to change the codec for outbond calls. Please
help not able to find anything using search.
thanks
arun
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trixter http://www.0xdecafbad.com wrote:
On Mon, 2005-09-26 at 10:33 -0700, Michael D Schelin wrote:
This can be done by modifying the source code.
how helpful. If I modify it enough it will be 100% identical to windows
xp, anything can be done by modifying any code. That however d
On Mon, 2005-09-26 at 10:33 -0700, Michael D Schelin wrote:
> This can be done by modifying the source code.
>
how helpful. If I modify it enough it will be 100% identical to windows
xp, anything can be done by modifying any code. That however doesnt
answer my question with anything that isnt
This can be done by modifying the source code.
trixter http://www.0xdecafbad.com wrote:
I have been asked if asterisk can change codecs dynamically based on the
calling party's caller id. I couldnt find anything, and dont know that
this is something that asterisk can do, but it occurs to
I have been asked if asterisk can change codecs dynamically based on the
calling party's caller id. I couldnt find anything, and dont know that
this is something that asterisk can do, but it occurs to me that
possibly with a reinvite it can be done, however I dont think you can
issue those from th
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