Are you looking for something like this?
Note: This will continuously go between the two trunks until the caller
hangs up, can be fixed by adding loop counter.
;
; extensions.conf
;
[LOADBALANCE]
exten => _X.,1,NoOp(Connect to least used trunk)
; - show active count
exten => _X.,n,NoOp(Calls:
Always has cleared the entire line..
On 15 December 2013 16:25, Dotan Cohen wrote:
> On Sun, Dec 15, 2013 at 3:58 PM, Tiago Geada wrote:
>> I would guess you need to recompile ?
>>
>
> I was under the impression that the library was dynamically linked.
>
> I am using the Ubuntu binaries for Aste
Yup - its definitely doable in FS.
On 15 December 2013 21:18, Patrick Lists
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 eliminate transcoding when un
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 si
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 vot
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
On Sun, Dec 15, 2013 at 9:32 AM, jg 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 you are the maker of rul
On Sun, Dec 15, 2013 at 3:58 PM, Tiago Geada wrote:
> I would guess you need to recompile ?
>
I was under the impression that the library was dynamically linked.
I am using the Ubuntu binaries for Asterisk, so if someone could
confirm that their Asterisk build does in fact kill (delete) a single
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 phone
On Sun, Dec 15, 2013 at 7:20 AM, jg 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 change the SIP config
I would guess you need to recompile ?
On 12 December 2013 20:07, Dotan Cohen wrote:
> On Wed, Dec 11, 2013 at 10:20 PM, Tzafrir Cohen
> wrote:
> > You need libedit-dev, not libeditline-dev.
> >
>
> Thank you Tzafrir. However, even after installing libedit and
> libedit-dev, Ctrl-W still kills
On Sun, Dec 15, 2013 at 5:07 AM, jg 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 used. I want
No - but this is a new setup so I can't say it worked before...it just isn't
working from the start.
I've found the call setup works and once bridged there is one way audio (to the
ATA, none from the ATA). And the the connection drops after 30 secs approx
because something on the path (or endp
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 engi
Hello
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]
disallow=
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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