On 2/22/2010 10:26 AM, Per Jessen wrote:
Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if
that helps.
/Per Jessen, Zürich
I have use this howto
http://www.voip-info.org/wiki
On 2/22/2010 7:36 AM, Tzafrir Cohen wrote:
On Sun, Feb 21, 2010 at 07:55:39PM +, Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
Which version of Asterisk do you use? Which channel driver?
I have use this howto
http://www.voip-info.org/wiki/view
On 2/22/2010 1:02 PM, Tzafrir Cohen wrote:
On Mon, Feb 22, 2010 at 12:22:39PM +, Pedro Santos wrote:
On 2/22/2010 10:26 AM, Per Jessen wrote:
Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
I've got an HFC-PCI (single channel
Does any one put a HFC-S card working in nt ptp mode?
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hi,
can anyone knows a way to make automatic calls from a list of numbers stored
in a file, one by one, as the calls hangs up.
EX:
1º call - hang up - 2º call - hang up - 3º call ..
thanks,
pn
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Hello everyone, I am new in voip.
I want to use a linux pc as a media gateway controller (with Megaco
protrocol if possible). I heard Asterisk could do it, but in the
documentation I haven't found information about it.
Could someone help me?
Thank you very much.
Pedro Gonzalez
Hi,
can anyone tell me how i do a sip trunk between an asterisk and a alcatel
omnipcx pbx with sip support
tx,
Pedro Santos
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Hi, I have been considering a purchase of the linksys spa3102 for a couple
hours but I would like to know from someone here, wether this device will
support faxing on my local asterisk server, I have had success sending and
recieving faces with an x100p, and recall that in the old
hi guys, is it possible to do a basic 3-or-more-way
conference call when the phones dont support it? I am
fully aware of this concept on expensive phones like
this one:
Grandstream GXP 2000 -Conference call 3-way
http://www.youtube.com/watch?v=hlZ6JqE1MT4
The problem is that the basic plain old
hi there!
I have a couple phones connected to a sipura ata and
if I go into *- IVR, I press options on the regular
phones and it all works fine and dandy.
then I connect an xten softphone, a new extension in
my dialplan, I dial the ivr, * asks me to dial
something to go through it, I press keys
hi there guys!
how can I eliminate this message?
[May 11 11:00:46] WARNING[7039]: res_musiconhold.c:506
monmp3thread: Unable to spawn mp3player
[May 11 11:09:06] WARNING[7039]: res_musiconhold.c:424
spawn_mp3: Found no files in
'/var/lib/asterisk/mohmp3'
This is on debian etch 4.0
asterisk 1.4,
Hello,
Can i identify the sound files that are played in the asterisk
console? I defined the verbose to 100 but i can not see the sound
files that are played in some situations... :(
For example, I need to know what files are played for the message:
Extension xxx is unavailable
The goal is
Hi there list!
I want to catch all numbers that don't exist, play a
nice message and restart operator, this is different
from dial i because that is for incorrect extensions,
an undefined number will give a busy signal, something
I don't like
You can search for the word irc to see my comments,
Ok so I read the Linux 2.6 related README and finally
compiled propperly, I thought but at the end I notice
that lscpi does report the cards, but I cant modprobe
wcfxo nor zaptel and I do have wcfxo.ko in the
/lib/modules/2.6.8/extra/ directory, so what gives?
This is a Debian Sarge, thanks!
Hi
i install Asterisk can register softphones on clients computers but when i
make a call to a extencion this error apear
Call Failed: not found
in the asterisk machine i do commannd sip show peers and i can see the
clients there
can you help me
thanks
Alex was right. The problem is that when i make changes in freepbx,
those changes are not written in the config files.
I only made modifications in files_custom.conf.
The version of freePbx that i use is 2.1.1 (not beta) and Asterisk 1.2.12.1.
Thanks by your help,
Ps.
2006/11/18, Alex Robar
. i dont know what causes this
error but i have noticed that restarting FreePBX or re-installing the
application stops this. Just restart the box
On 11/18/06, Pedro Silva [EMAIL PROTECTED] wrote:
Alex was right. The problem is that when i make changes in freepbx,
those changes are not written
Hello,
From some days ago, when i made changes in web interface to asterisk
that comes with trixbox (freepbx), this dont reflect the changes in
asterisk configuration.
I has reviewed the file permissions in /etc/asterisk and all files are
writable to asterisk user.
In freepbx all appears to be
Hi,
2006/11/17, Alex Robar [EMAIL PROTECTED]:
Hi Pedro,
Did you press the red bar at the top of the page? Until you do this, the
config files are not written out.
Yes, i press the red bar and freepbx dont return any error.
For example, If i add a new extension, the files
Hello,
When i try to install the sfftobmp3.1, the tribbox box give me the
following error:
...
checking for TIFFOpen in -ltiff... yes
checking jpeglib.h usability... no
checking jpeglib.h presence... no
checking for jpeglib.h... no
configure: error: jpeglib.h not found
I try to find packages
Hello,
Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?
I tried to apply this patch (from http://www.mlkj.net/asterisk/) but
it give me errors...
Also i tried define one extension for fax receptions but this dont works:
exten = 1,1,Goto(handle_fax,s,1)
exten =
Excellent, Michiel! This works :)
You know what kind of file it is created (SFF)?
Can you send to me the example faxreceive.php?
Thanks and best regards!
PS.
2006/11/7, Michiel van Baak [EMAIL PROTECTED]:
On 15:03, Tue 07 Nov 06, Pedro Silva wrote:
Hello,
Anyone knows if chan_capi-0.7.1
2006/11/1, Armin Schindler [EMAIL PROTECTED]:
On Wed, 1 Nov 2006, Pedro Silva wrote:
As you can see in the log below, the called number is just '0':
CalledPartyNumber = 810
It seems DDI 0 of your line was called. So just do
exten = 0,n,Dial...
Armin
Is that right! Thanks
Hello all,
To test some configs i forgot the trixbox web config (freepbx) and i
made changes directly in asterisk config files (sip.conf,
extensions.conf, etc). Result: asterisk is working ok but the the web
config is totaly confused and, if i made a change via freepbx this not
works ok. Only
Hello,
The problem was wrong contexts defined like Marco said, and is solved.
Now, i have another problem...of course :)
On incoming calls, i only can receive calls if i define a line like
the following, in extensions.conf:
exten = _.,n,Dial(SIP/500,30,tr) (all incoming calls are redirected
to
... but is
problem in asterisk or is before asterisk, on diva card...?
Tanks by any possible help!
Best regards,
PS.
2006/10/29, Pedro Silva [EMAIL PROTECTED]:
Finally this works!!! :)
Tanks to Alberto and Marco by your help!
The problems are:
- the cable was connected to the wong card port
problem?
Thanks and best regards,
PS.
2006/10/29, Alberto Pastore [EMAIL PROTECTED]:
Pedro Silva ha scritto:
Hello,
I need to connect one diva server 4bri to a portuguese BRI interface.
The operator (PT) said that this bri is in point-to-multipoint mode
(S0). Previously one PBX has connected
that the problem is between diva card and BRI access.
So i need to solve first this problem and only after that im care with
asterisk... :)
Obrigado desde já pela disponibilidade de ajuda!
PS.
May be i can help.
Sou Português:)
On 10/29/06, Pedro Silva [EMAIL PROTECTED] wrote:
Thanks Alberto!
I
Hello again Alberto!
Anyway, to get more info, try to open a second shell
and run /usr/lib/eicon/divas/xlog
then on the first shell redo the telsampl test, then
post the output of xlog off the list to my address
(alberto at msoft-italia.com)
This is the xlog output:
4:1736:074 - CREATEID ok:
Finally this works!!! :)
Tanks to Alberto and Marco by your help!
The problems are:
- the cable was connected to the wong card port... :(
- the card config needs to be: ETSI; TE; Point-to-Point (I thought
that was point-to-multipoint).
Best regards,
PS.
2006/10/29, Pedro Silva [EMAIL PROTECTED
Hello,
I need to connect one diva server 4bri to a portuguese BRI interface.
The operator (PT) said that this bri is in point-to-multipoint mode
(S0). Previously one PBX has connected to that interface.
The asterisk and diva drivers are working ok but i cannot communicate
to outside via this
I just installed rpm binaries in a new mandriva and I
see a frew error messages with asterisk -vvvcfg,
btw I would also like a little guidance to just set up
a couple sip phones to start playing with soft phone
communication with 3 pcs on the network
thanks :)
ng
to declare all contexts in extensions.conf so we can use it
on Realtime??
Another question:
Its possible to include contexts in Realtime like we made on
extensions.conf?
Thanks in advance,
Pedro Nunes
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;
Default context gets used in siutations where you are using
the GK routed model
or no type=user was found. This gives you the ability to
either play an invalid
message or to simply not use user
authentication at all.
Thanks in advance.
Pedro
leading digit I couldn't do it yet.
Pedro Nunes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sven
Fischer
Sent: quinta-feira, 9 de Fevereiro de 2006 9:23
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Leading 0 on caller ID with internal S0
Hi there guys,
Does anyone know what this is??
Every time a mISDN channel connects to anything, I get this message on
the CLI of asterisk.
Unhandled Message: prim 281 len -22 from addr 51400101, dinfo 0 on port:
1
Thanks
Pedro Nunes
Hello,
Do you try
Answer() and then Dial(SIP/xyz,,m)???
Exten = ???,1,Answer()
Exten = ???,2,Dial(SIP/xyz,,m)
You need to answer the call before you can hear music on hold.
Pedro Nunes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Clauson
te_choose_channel=no
dialplan=0
use_callingpres=yes
;always_immediate=no
;immediate=no
;hold_allowed=yes
;callgroup=1
;pickupgroup=1
;presentation=not_screened
;echocancel=no
echocancelwhenbridged=no
echotraining=yes
[group1]
ports=1
context=bri_card_1
msns=*
Thanks in advance
Pedro Nunes
Hello all,
I need to install a production server with BRI support.
I know that exists bristuff, misdn, chan_capi ...
I have hcfpci based cards.
For a very stable environment, what driver should I use??
Thanks in advance
Pedro Nunes
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I think you are thinking of iLBC:
http://www.voip-info.org/wiki-iLBC
Be aware that this codec is known to be pretty CPU intensive to accomplish its compression.
- PedroOn 11/22/05, Sam Tam [EMAIL PROTECTED] wrote:
I think I have heard in the past that someone mentioned to me there is acodec
to match the 1.0.10 version or will 1.0.9 be the
final release of those packages?
- Pedro
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no changes, so they have
not been updated.
It is very likely that this will be the final release of the 1.0
branch of Asterisk. Users are strongly encouraged to begin upgrading to
version 1.2.
Thanks!
On 11/22/05, Pedro [EMAIL PROTECTED] wrote:
I noticed that asterisk.org now has asterisk
will be appreciated.
- Pedro
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database server on . Check debug for more info.
Nov 21 13:17:08 WARNING[31192] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug.
Any thoughts?
- Pedro
On 11/21/05, Alexander Lopez [EMAIL PROTECTED] wrote:
It is a better practice to use a noload option in
modules.conf
to connect database serveron . Check debug for more info. Nov 21 13:17:08 WARNING[31192] res_config_mysql.c: MySQL RealTime:
Couldn't establish connection. Check debug. Any thoughts? - Pedro On 11/21/05, Alexander Lopez [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote: It is a better practice to use
!
-PedroOn 11/21/05, Olle E Johansson [EMAIL PROTECTED] wrote:
Pedro wrote: Yeah - tried that.Here are 2 lines I have in my modules.conf file: noload = pbx_realtime.so noload = app_realtime.so For some reason, I still get the following in my logs even after a
restart of Asterisk. Nov 21 13:17:08 ERROR
What jitterbuffer issues are you having with connecting to 1.0.x servers?On 11/17/05, Eric ManxPower Wieling [EMAIL PROTECTED]
wrote:Asterisk guy wrote: does it include the patch for VAD?
( dropping extra frame of G.729 since we already have a VAD frame at the end )It does not include several
I just installed asterisk 1.2 rc2 and ran a 'make samples' and asterisk
starts just fine with no errors in the logs. However, if I issue
a reload I get the following:
Nov 15 17:08:22 WARNING[27009] res_musiconhold.c: Music on Hold class 'default' already exists
It is almost like the previous
What chipset that card use??
Pedro Nunes
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Arulraj
Sent: terça-feira, 1 de Novembro de 2005 23:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Fritz!Card PCI
in advance
Pedro Nunes
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Outubro
de 2005 15:30
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: R: [Asterisk-Users]
Bristuff question
http://www.voip-info.org/wiki-Asterisk+zaphfc
look this
Giordano
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Per conto di Pedro Nunes
Inviato
Thanks,
That will fix my problem... And agent skills, is that possible too??
Thanks again
Pedro Nunes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lorenzo
Emilitri
Sent: quinta-feira, 13 de Outubro de 2005 8:17
To: Asterisk Users Mailing List - Non
Thanks,
That will fix my problem... And agent skills, is that possible too??
Thanks again
Pedro Nunes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes
Sent: quarta-feira, 12 de Outubro de 2005 23:39
To: Asterisk Users Mailing List - Non
to an mssql DB, try to find the callerID number in table
extensions, and then sets a variable named Name to the value of
table Name. Cool hah...
Pedro Nunes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Serge
Lhermitte
Sent: quarta-feira, 12 de Outubro de
feature available on most PBX that support ACD functionality.
Does anybody knows how to
do it with asterisk??
Thanks in advance
Pedro Nunes
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Asterisk
it to
our VoIP product offerings.
- Pedro
http://www.traci.netOn 10/8/05, Cory Andrews [EMAIL PROTECTED] wrote:
The F3000 is not anticipated to be available for distribution until lateDecember/January, FYI.
Cory AndrewsSenior Partner+++VOIPSupply.com454 Sonwil DriveBuffalo, NY 14225
Had not seen a response on the following question - wondering if
anyone may have any insight on this?
Original Question-
Is there a way to log SIP response codes without enabling verbose
logging? Reason being is that from time to time I see a call fail on
our primary provider and
Is there a way to log SIP response codes without enabling verbose
logging? Reason being is that from time to time I see a call fail on
our primary provider and roll-over to our backup providers. If I
happen to catch it on the console I can see the code 484 or similar.
It would really help in
I have searched quite a few places and have not seen this discussed.
Basically I was wondering how would you go about having an option for
a user to be notified every 15 minutes until their new voicemail
message is checked. Since the notification e-mails we send get sent
to cell phones or actual
Thanks - a cronjob for the user was going to be my last resort. Was
not sure if there was a setting like repeatnotify=15 to repeat the
notice every 15 minutes.
Thanks for your feedback though!
On 7/1/05, Michiel van Baak [EMAIL PROTECTED] wrote:
On 13:33, Fri 01 Jul 05, Pedro wrote:
I have
Looks like 9 out of 10 calls are failing on voipjet at the moment (at
least terminating to South Florida numbers). Keep getting message
that says number can not be completed as dialed. Anyone else seeing
this?
On 6/15/05, Pedro [EMAIL PROTECTED] wrote:
Couple of days. Apparently the new US
I need help to make a conection form FWD to my pbx,
I can receive a call from PSTN for a FXo card but know I need to receive call
via IP form FWD I have activate hte IAX on freeworlddialup but does not work I
can't make or receive calls. I virtually new in this can please somebody help
me.
Couple of days. Apparently the new US carrier has some changes that
needs to be made.
On 6/14/05, Wiley Siler [EMAIL PROTECTED] wrote:
Did they say when it would be corrected?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedro
Sent
Caller ID is still not working to certain areas. This problem was
confirmed by voipjet tech support in their last e-mail to me.
On 6/13/05, Matt [EMAIL PROTECTED] wrote:
I never noticed any problems.. so I can't comment :) hehe
On 6/11/05, Pedro [EMAIL PROTECTED] wrote:
Finally got
Finally got a response from voipjet support and they say they have
switched to a new provider for US termination. I have yet to test
this out as I have not had a chance to build them back into our routes
but will report my findings once I do. Anyone else notice any
improvements?
On 6/9/05,
Seems things have just got worse. Just got reports that 800 numbers
are not terminating. For example, can not dial:
800-888-9358
or
800-922-4684
Had to pull voipjet out of our routes until this gets fixed.
On 6/9/05, Moody [EMAIL PROTECTED] wrote:
We have been having serious quality problems
will definately check them out.
On 6/10/05, Pedro [EMAIL PROTECTED] wrote:
Please provide the SIP or IAX provider you are using that allows you
to terminate to 800 numbers for free.
On 6/10/05, Matt [EMAIL PROTECTED] wrote:
Why would you even be routing 800 numbers out voipjet? They CHARGE
Definately problems with voice quality and caller ID is not working
very well. I have e-mail a couple times and still no response from
their tech support on this. This is very concerning since I tried all
3 servers with the same results.
On 6/8/05, Julio Arruda [EMAIL PROTECTED] wrote:
Roman
I have done some searching and not sure this is even possible, but
here it goes...
**Scenario**
Let's say you have an asterisk server that you use to connect to a SIP
provider that you push your PSTN-bound calls to using g711 and
out-of-band DTMF. The SIP phones in use are Cisco 7960's and are
http://www.lightreading.com/document.asp?doc_id=73943site=lightreading
I know e911 has been discussed on ths list before, but I just read
this and it got me thinking that if you have a Wholesale VoIP carrier
- wouldn't they have to pass e911 on to you as a VoIP provider to, in
turn, pass on to
Actually called Digium with this exact question last week. They said
that you can register the new license on the new server provided that
you ony registered it once before. They said there is no unregister
script to unregister the license from the old server, however. If you
have already used
What I did once was create an announcement that got played to the
receptionist announcing who the call was for based on the number that
was called. This allowed the receptionist to know which greeting to
recite.
On 5/2/05, Michael Welter [EMAIL PROTECTED] wrote:
Chris Mason (Lists) wrote:
The
In your case, where you will need the license is on the box that your
phones register to. For exampe, when someone checks voicemail,
encoding takes place, therefore you need a license.
Look at it this way:
[g729 provider] -(SIP or IAX)--- [g729 asterisk server]
- no license
Just wanted to correct this last post - apparently, you can configure
the other speed buttons to also be separate lines with their own SIP
account.
On 4/22/05, Pedro [EMAIL PROTECTED] wrote:
Just got the new Mitel 5220 Dual Mode IP Phone and I have to say I am
impressed so far. Changing to SIP
Thank you for your feedback.
I was mearly wondering if others had experienced this issue in their
environments. Was not trying to open a bug report or officially
report an issue. Strictly a curiousity request. Really do not want
to upgrade if everything else works fine. Since this issue
Have you tried to enable NAT translation on the Grandstream?
On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote:
I'm trying to register BT100s ... (doesn't work)
X-Lite seems to work though
Tomas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
across anything else
important I will let you know.
Pedro
TRACI.net
On 4/4/05, Kris Edwards [EMAIL PROTECTED] wrote:
Here's a good sign:
Mitel is also addressing economy in adding SIP compliance to two of its
IP phones. The Mitel 5215 and 5220 run Mitel's proprietary MiNET
protocol
Yes, same problem here. Sign-ed up with VoipJet and seems to work
just fine (prices for most areas we call are cheaper too from what I
saw). Only been using them for 24 hours so can't say much about
long-term stability, but so far so good.
Pedro
On 4/19/05, Matthew Asham [EMAIL PROTECTED
) Cisco 7960
Any thoughts are appreciated.
Thank You,
Pedro
TRACI.net
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Will let you know - getting one soon to test.
On Feb 13, 2005 10:58 PM, eric m [EMAIL PROTECTED] wrote:
Hi,
I really appreciate the look and design of newer Mitel Ip phone.
I search througt the list and found only fews notes about the use Mitel 5055
phone on *. Anyone use other model
Just a note that you will need to perform quite a few incremental
upgrades to get to a current firmware version. So if you do get
someone who will sell you the firmware, make sure you get the all of
them.
On Fri, 18 Mar 2005 22:04:29 -0500, Patrick M. Gray, Jr.
[EMAIL PROTECTED] wrote:
Sounds like you are having a codec issue with 2 of your providers.
Make sure you find out what codecs are supported and that your config
is set up accordingly.
On Sun, 06 Mar 2005 00:14:05 +, w fm3 [EMAIL PROTECTED] wrote:
Hi
Hope someone can help :)
I am testing 4 PSTN termination
transfer.
Now, I am running Asterisk CVS-v1-0-11/12/04-15:32:45. I would be
interested in knowing if later versions of asterisk exhibited this
same behavior. Any feedback would be appreciated.
Thanks,
Pedro
On Fri, 11 Feb 2005 08:32:43 +0100, Florian Overkamp
[EMAIL PROTECTED] wrote:
Hi
.
When I try
to call somebody the * show me an error : RECEIVED UNKNOWN MESSAGE TYPE:
4
Somebody have
a skinny.conf sample file for Cisco 7935 or any trick to fix this
problem
Thanks,
Pedro.
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Hi, my name is Pedro Caria I'm new to this list.
I live in Portugal and find myself in the position to talk often to
various parts of the world, very often the Telco line has a delay
superior to 1s, I also fax in the same conditions, so to my experience
faxes do work with delays far superior
on the mediatrix to include asterisk as the
realm. The other fields are pretty self explanatory (username,
password, etc.). You will also want to turn off silence suppression
as it is on by default.
- Pedro
On 25 Feb 2005 20:07:04 +0100, Edward Banfa [EMAIL PROTECTED] wrote:
Hello all,
Hi I would
First off -
change:
exten = _2000,1,Dial(IAX/master:[EMAIL PROTECTED]/[EMAIL PROTECTED])
to:
exten = _2000,1,Dial(IAX2/master:[EMAIL PROTECTED]/[EMAIL PROTECTED])
On Mon, 21 Feb 2005 13:00:39 -0500, kurt x [EMAIL PROTECTED] wrote:
I have two * boxes running two differnet versions of *.
Box
If you use the MySQL CDR add-on, you could just query the CDR DB for
the numbers you are tracking. No need to add anything fancy.
On Sat, 19 Feb 2005 21:42:31 +0100, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi all.
I am going to do a simple voting application for a radiostation.
Vonage, to my knowledge, does not let you connect your own SIP device
to their service. They provide their own IAD.
As for Broadvoice, I know people that have successfully deployed
asterisk with many people sharing the same account.
- Pedro
On Fri, 18 Feb 2005 17:54:38 +0800, el Flynn [EMAIL
Rich - thanks! Glad I am not the only one seeing this :)
Would be very interested in your results. No problems that I see yet
with these settings.
On Fri, 18 Feb 2005 10:21:08 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
That does not sound right at all. The difference between the two
Make sure you have the proper licenses to use the codecs:
g729
http://www.digium.com/index.php?menu=asterisk_g729
g723
http://www.dspg.com/technology/LicensePricing.html
On Fri, 18 Feb 2005 18:17:58 -0800, Nitesh Divecha
[EMAIL PROTECTED] wrote:
Hello All,
Any one has success with codec
?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Pedro
Sent: Wednesday, February 16, 2005 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN
FYI - Seems
That does not sound right at all. The difference between the two Time=
values should have been 10 (milliseconds).
Did you reboot the Sipura after making the change? There are some values
in the Sipura that don't take effect until after the next reboot; I don't
have a clue whether this
? Look at the timestamps between RTP packets if
you can't see/modify this setting.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Pedro
Sent: Tuesday, February 15, 2005 6:30 PM
To: Jeffrey Chan
Cc: Asterisk Users Mailing List
Forgot to mention that when I set the RTP Packet Size to 20ms that the
difference was 160 (like the Cisco) but call quality was much worse.
On Wed, 16 Feb 2005 15:36:44 -0500, Pedro [EMAIL PROTECTED] wrote:
Thanks for the suggestion. Changing the RTP Packet Size in the Sipura
to 40ms did
, Pedro [EMAIL PROTECTED] wrote:
Forgot to mention that when I set the RTP Packet Size to 20ms that the
difference was 160 (like the Cisco) but call quality was much worse.
On Wed, 16 Feb 2005 15:36:44 -0500, Pedro [EMAIL PROTECTED] wrote:
Thanks for the suggestion. Changing the RTP Packet
as well.
Pedro
On Tue, 15 Feb 2005 09:52:56 +0100 (CET), Remco Barende
[EMAIL PROTECTED] wrote:
Hi list!
I have some sip phones and Sipura ATA 2000's. However after dialling a
number I need to dial a # to control a device.
When I dial # Asterisk kicks in and puts the call on hold. How can I
just need to know if it will be weeks/months/ or days
[15:21] sixtel9: days
- Pedro
On Tue, 15 Feb 2005 09:41:12 -0500 (EST), Paul Dugas
[EMAIL PROTECTED] wrote:
On Tue, February 15, 2005 9:27 am, Rob Risner said:
I'm just wondering, how long should a vanity number transfer really take
Is there a way to somehow do an escape # so that you can still use
the # key to control devices that require a #, but still keep the T in
the dial plan? We have clients that need to check external voicemail
systems that require the use of the # sign, but still want to have the
call parking
)
and a Fritz Capi to connect to my telecom provider.
I can send faxes with some success, but receiving rate of success is
less than 30%.
Fax information for Asterisk is difficult to come by is everybody using
spandsp's way ?
Thx
Pedro Caria
On 15/fev/2005, at 15:05, Rich Adamson wrote:
On Tue, February
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