Hi John,
Em 17/08/2014 12:04, Tech Support aster...@voipbusiness.us escreveu:
So my question is this: Does anyone know which options can and can't
be overridden on a per-mailbox basis? Are there 9 options, are there 39, or
are there 26? Where can I find a definitive answer? It would be
Hi Matthew,
Em 28/05/2014 15:09, Matthew Jordan mjor...@digium.com escreveu:
* now - tell all CDRs to go submit themselves. Tell all channels to
Where CDR is quoted is it also valid for CEL?
Tks.
--
_
-- Bandwidth and
Dear friends,
In Asterisk 11.7.0, is it possible to receive CEL APP_START and
APP_END events from Park application?
Queue and Dial apps are generation these events, but Park no. It
doesn't seem to make difference when configured.
cel.conf
...
apps = dial,queue,confbridge,park
events = ALL
...
Do you have any idea when the voice is heard only when the call is from my
local network to the Internet and not in the other direction ?
Hi Neo,
In the documentation look for this options:
externip
localnet
nat
It helps to understand them because this kind of situation is common with VOIP.
a conference ended should I use
CEL or is there another better approach?
Best.
2014-03-06 21:28 GMT-03:00 Rusty Newton rnew...@digium.com:
On Wed, Mar 5, 2014 at 1:30 PM, Jairo jairomolin...@gmail.com wrote:
Dear friends,
Need to know filenames of conference recordings in Asterisk 11
Great :)
Thank you very much.
Best.
2014-03-07 18:07 GMT-03:00 Rusty Newton rnew...@digium.com:
On Fri, Mar 7, 2014 at 7:21 AM, Jairo jairomolin...@gmail.com wrote:
Thank you very much Rusty.
It really works. Even if ${MyCustomFileName} gets a different value when
the
second
Dear friends,
Need to know filenames of conference recordings in Asterisk 11.
Besides directory scanning the recordings could use CEL:
Filter MySQL rows with eventtype equal CHAN_START and channame like
ConfBridgeRecorder and then get the eventtime field and convert to
timestamp to complete
Hi Eric,
Take a look at the call pickup code, maybe you need to change it to not
conflict with your dialplan:
localhost*CLI features show
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #1
Thanks for the feedback.
Best.
2014-02-13 13:16 GMT-02:00 Eric Cooper e...@cmu.edu:
On Thu, Feb 13, 2014 at 10:09:04AM -0200, Jairo wrote:
Take a look at the call pickup code, maybe you need to change it to not
conflict
with your dialplan:
localhost*CLI features show
Builtin Feature
: ATTENDEDTRANSFER
eventtime: 2013-11-20 15:05:33
userdeftype:
cid_name: Jairo desktop
cid_num: 311
cid_ani: 311
cid_rdnis:
cid_dnid: 310
exten: 310
context: entrada-canal
channame: SIP/311-351096-048d
appname: Dial
appdata: SIP/310-777940,40,kKtT
amaflags: 3
Hi Jean, you mean what each event indicates? As this link explain?
https://wiki.asterisk.org/wiki/display/AST/CEL+Events+and+Fields
2013/11/19 Jean-Denis Girard jd.gir...@sysnux.pf
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nobody, really?
Thanks,
- --
Jean-Denis Girard
SysNux
Dear list.
This is probably a complex subject but is that right to consider:
a) each distinct linkedid field value in a mysql CEL table as a unique call?
b) the duration of a call as the period (eventtime fields) between
BRIDGE_END and BRIDGE_START events of the same linkedid sequence? (not
Maybe this can help:
http://ofps.oreilly.com/titles/9781449332426/asterisk-Fax.html
Best.
2013/6/13 vortex binary.vor...@gmail.com
Hello. i am running debian 6 with asterisk 11.4. The system has exim4 to
send to email the voicemails.
i would like to get rid of the analog fax machine and
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Jairo Molina Jr∴
http://www.intermol.com.br
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk
it? If so, there is no
Asterisk analog to calldate. You would need an alias set up. Mine looks
like:
alias start = calldate
so that the start of my call is what gets logged to the database as the
calldate.
Kevin Larsen
From:Jairo ja...@intermol.com.br
To:Asterisk Users
signalling=fxs_ks
channel = 1
channel = 2
channel = 3
I was using the cvs head version because I need the
wctdm driver for the TDM04B (4*fxo modules)
Does anybody know what is wrong?
Thanks in advance!!
Jairo Barahona Garita
inCom Developer
Hi!
I have an Asterisk Box with one E1. This is connected
with PSTN. My problem is that periodically the
Asterisk console shows the following message.
-- B-channel 0/1 succesfully restarted on span 1
-- B-channel 0/2 succesfully restarted on span 1
-- B-channel 0/3 succesfully restarted on span 1
Hi!
I have an Asterisk Box with one E1. This is
connected
with PSTN. My problem is that periodically the
Asterisk console shows the following message.
-- B-channel 0/1 succesfully restarted on span 1
-- B-channel 0/2 succesfully restarted on span 1
[..etc...]
I don't know if this
You can use Ethereal to see what your phone (stun) is
sending. Of this way you can see the RTP ports and IP
public that your phones are going to use. You can see
that information in INVITE and OK packets.
For other hand, If you use one router with symmetrical
NAT then Stun won't work
Hi Rafa!
Hi Jairo,
Try with other values for the jitter in your Gateway
(H323).
One customer have a scenario like this:
Phone/Fax Gateways H323 -with 16/8/2/1 Port
FXS- --- GNUGK
--- Asterisk --- Zap (E1)
We have the same configuration, and I think that our
problem is the jitter
Hi!
I want to send fax to PSTN with Asterisk, but by now I
can't.
I am using the following boxs:
Internet Zap E1
Phone/Fax Gateway(H323)--- Asterisk--- PSTN
The gateway H323 has T38 and T30. Before I began with
Asterisk, I used Cisco to connect with PSTN,
client could be connected to Asterisk... am I
right?
With for instance, the projects of the openh323 (http://www.openh323.org)???
Thank you,
Jairo Cavanus
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