I think the order or elements is relevant:
[100]
disallow=all
allow=ulaw
allow=g722
or
[100]
allow=!all,ulaw,g722
should work.
jg
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Le 15/12/2013 11:07, jg a écrit :
I think the order or elements is relevant:
[100]
disallow=all
allow=ulaw
allow=g722
or
[100]
allow=!all,ulaw,g722
should work.
[...]
Yes, but what about if 100 have g722 as prefered codec? Eg:
[100]
disallow=all
allow=g722
allow=ulaw
[101]
I see, you do want something like picking g722 provided there is no transcoding. Because
Asterisk is a B2BUA it can transcode, so it would choose g722 where the other party is doing g711.
For known parties, maybe one could change the SIP configuration on the fly using the Asterisk
realtime
On Sun, Dec 15, 2013 at 5:07 AM, jg webaccou...@jgoettgens.de wrote:
I think the order or elements is relevant:
[100]
disallow=all
allow=ulaw
allow=g722
or
[100]
allow=!all,ulaw,g722
should work.
jg
If I choose that order and the phone supports both ulaw and g722 only ulaw
will be
On Sun, Dec 15, 2013 at 7:20 AM, jg webaccou...@jgoettgens.de wrote:
I see, you do want something like picking g722 provided there is no
transcoding. Because Asterisk is a B2BUA it can transcode, so it would
choose g722 where the other party is doing g711.
For known parties, maybe one could
You are correct. Your idea of the prioritize_matching_codecs option is what I am looking for.
Yes Asterisk can transcode, but why transcode when you don't need to. If the phone is
advertising both formats it should support them. If the phone only supports local MOH in one
format then the
On Sun, Dec 15, 2013 at 9:32 AM, jg webaccou...@jgoettgens.de wrote:
Is it possible to let the Sangoma card work only on the most demanding
codecs? This requires some analysis to estimate the benefits. Another
question is whether the user phones are provisioned or not. If provisioned,
then
I still don't have a way to enable the higher quality g722 codec for internal use without
making a transcoding mess. Maybe Asterisk 12 with pjsip will have a better solution.
Currently, I am no longer using g722 anymore for production setups. I had a some SIP-Phone
combinations (not Polycom, not
I have had the issue for years. The problem is that Asterisk
developers are removed from the business. We desperately need simple
way to eliminate transcoding when unnecessary. Transcoding brings a
server to its knees. It is a very simple new setting in sip.conf
prioritize_matching_codecs=yes
I
On 12/15/2013 09:55 PM, CDR wrote:
I have had the issue for years. The problem is that Asterisk
developers are removed from the business. We desperately need simple
way to eliminate transcoding when unnecessary. Transcoding brings a
server to its knees. It is a very simple new setting in
Yup - its definitely doable in FS.
On 15 December 2013 21:18, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 12/15/2013 09:55 PM, CDR wrote:
I have had the issue for years. The problem is that Asterisk
developers are removed from the business. We desperately need simple
way to
Let's say I have two devices configured and the follow call scenarios occur.
[100]
disallow=all
allow=g722ulaw
Polycom phone with g722,ulaw,alaw,g729
[101]
disallow=all
allow=ulaw
Polycom phone with g722,ulaw,alaw,g729
101 dials 100 - ulaw to ulaw is chosen
100 dials 101 - g722 to ulaw is
On Sat, Dec 14, 2013 at 10:31 PM, Ryan Wagoner rswago...@gmail.com wrote:
Let's say I have two devices configured and the follow call scenarios
occur.
[100]
disallow=all
allow=g722ulaw
Polycom phone with g722,ulaw,alaw,g729
[101]
disallow=all
allow=ulaw
Polycom phone with
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