You did read the documentation I presume?
It clearly states that you need to have separate G729 licenses to have *
talk G729, without them * can only pass thru G729, not transcode it (Zap <->
G729).
Since both your sip phones do G729 * is just performing pass thru.
Kind regards,
E. Ver
If you’re using
G.711 make sure you’ve got silence suppression turned OF, seems that the
phone only receives rtp while sending it.
(or, do as we did, throw
the OptiPoint out of the window J)
Van:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens richard Coco
Verzonden: m
Qui, mais je ne parle pas français ;)
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Patrick
Verzonden: donderdag 23 december 2004 13:28
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: RE: [Asterisk-Users] Asterisk billing solution
is busy, after that extension hangs up it
should trigger the dial to the originating extension and on pickup it should
dial the other extension J
Kind regards,
E. Versaevel
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Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] Queueueueuueue position
On Thu, 16 Dec 2004 15:18:10 +0100, E. Versaevel <[EMAIL PROTECTED]> wrote:
> When I call in (with an agent logged in) I get to hear the MOH on the
client
> side, hover no matter how high th
Discussion
Onderwerp: Re: [Asterisk-Users] Queueueueuueue position
On Thu, 16 Dec 2004 15:18:10 +0100, E. Versaevel <[EMAIL PROTECTED]> wrote:
> When I call in (with an agent logged in) I get to hear the MOH on the
client
> side, hover no matter how high the hold time is, I NEVER get an
hear the MOH on the client
side, hover no matter how high the hold time is, I NEVER get an announcement
over my queue position or my estimated wait time?
Both the incoming call and the agent are on SIP channels.
What is wrong ?
Kind regards,
E. Versaevel
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting
Erik
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Ian Chilton
Verzonden: dinsdag 14 december 2004 11:33
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] Dial Plan Problems
Hi,
I a
- Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] High(er) availability
On Tue, 2004-07-12 at 10:54 +0100, E. Versaevel wrote:
> Hello,
>
> If one would like to build a redundant Asterisk setup, would it be
possible
> to exchange the locationdb for the SIP users betw
cial Discussion
Onderwerp: Re: [Asterisk-Users] High(er) availability
On Tue, 2004-12-07 at 15:47 +0100, Stefan de Konink wrote:
> E. Versaevel wrote:
> >>Which app do you use for monitoring the primary box and if it fails
> >>taking over the IP address by the backup one? I haven&#
That would lead more to keepalived I think
Would be an option, but I would have to use fixed IP addresses for the IP
Phones (that should not be a problem)
Erik
E. Versaevel wrote:
>>Which app do you use for monitoring the primary box and if it fails
>>taking over the IP address b
>On Tue, 2004-12-07 at 10:54 +0100, E. Versaevel wrote:
>> Hello,
>>
>> If one would like to build a redundant Asterisk setup, would it be
>possible
>> to exchange the locationdb for the SIP users between then?
>
>Basically I would start with building redund
until they have
reregistered themselves at the backup asterisk. Is there a SER like t_relay
kinda thingy to let the backup know the locations of the Sip Phones?
Kind regards,
E. Versaevel
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[EMAIL PROTECTED]
http
A DigitTimeout(3) will do wonders to (and fix the non existing priorities).
Kind regards,
E. Versaevel
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens [EMAIL PROTECTED]
Verzonden: vrijdag 3 december 2004 21:52
Aan: Asterisk Users Mailing List - Non
Have you checked if ‘nice’
allso exists?
It tries to move the
soxmix to the background
Van:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Craig Waddington
Verzonden: woensdag 1 december
2004 15:56
Aan:
[EMAIL PROTECTED]
Onderwerp: [Asterisk-Users]
Asterisk Call Monitor
Before I start looking in the wrong places, does asterisk support the
indications.conf on SIP channels?
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens E. Versaevel
Verzonden: donderdag 25 november 2004 9:51
Aan: 'Asterisk Users Mailing
playback, but no busy tone.
I tried to enter the values directly into playtones,
but that didn’t work either.
Am I missing something?
Kind regards,
E. Versaevel
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[EMAIL PROTECTED]
http://lists.digium.com
tries to connect the incoming sip call
to 101, hence the loop :)
Kind regards,
E. Versaevel
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Adnan Ahmed
Verzonden: maandag 22 november 2004 21:34
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] SIP
Which is not for
commercial use…
Van:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Namens Vinicius Viana
Verzonden: dinsdag 23 november
2004 14:57
Aan: Asterisk
Users Mailing List - Non-Commercial Discussion
Onderwerp: RES: [Asterisk-Users]
Commercial g723.1 license for asterisk
sip message is not relayed back to my sip phone, it just sits and waits for a
timeout.
Is it possible to relay the returned SIP (error)
message to the softphone?
Kind reagards,
E. Versaevel
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[EMAIL PROTECTED]
Hmm, a bit closer, Asterisk seems to do the asterisk@ part only with non
numeric usernames, ie [EMAIL PROTECTED] stays [EMAIL PROTECTED] but
[EMAIL PROTECTED] turns into [EMAIL PROTECTED]
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens E. Versaevel
tating the username of the Siemens phone.
When I place a call with X-Lite the From: header is altered and reads
[EMAIL PROTECTED] instead of [EMAIL PROTECTED]
Any idea how this is possible?
Kind reagards,
E. Versaevel
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[EMAIL
ve made a SIP scenario trace of the callsetup, I'm a bit puzzled by the 2
time call setup?
Kind regards,
E. Versaevel
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens E. Versaevel
Verzonden: donderdag 18 november 2004 13:53
Aan: [EMAIL PROTECTED]
PROTECTED]
Verzonden: donderdag 18 november 2004 11:41
Aan: E. Versaevel
Onderwerp: Re: [Asterisk-Users] Setup/SIP routing
Hi
On Thu, 18 Nov 2004 11:32:08 +0100, E. Versaevel <[EMAIL PROTECTED]> wrote:
> However, I'm having troubles routing incoming sip traffic to SER,
asterisks
> keeps
hat) and has to authenticate outbound calls (SER
can't do that either)
However, I'm having troubles routing incoming sip traffic to SER, asterisks
keeps messing up the form header (replacing it by the dialed context, ie
[EMAIL PROTECTED] )
Any ideas if this setup is even possible?
Kin
d out how to get *
out of the media path)
I'm still new to all this, but I think this could work.
Kind regards,
E. Versaevel
Extensions.conf
[sip_in_from_carrier]
exten => _XX, 1, Dial(SIP/[EMAIL PROTECTED],20,r)
;Not a 10 digit number
exten => s,1,Answer
exten => s,2,Musi
ot Found" back
from myserbox), but it isn't relayed to the carrier.
I'm also talking with the carrier about skipping the authorization (or
moving it to a lower layer IE vpn oid), but I like to have a solution ready
if the carrier doesn't want that.
Kind regards,
E. Versaevel
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