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:
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
You can also adjust the Interdigit Long Timer and Interdigit Short
Timer values found in the Regional settings config screen.
- Pedro
On Fri, 28 Jan 2005 13:36:14 +0100 (CET), Remco Barende
[EMAIL PROTECTED] wrote:
On Fri, 28 Jan 2005, David John Walsh wrote:
The delay is a time out
understood - I use the # sign as well, but some users are not used to
using the # sign so decreasing the timer helps those that may forget
to use the # key.
-Pedro
On Fri, 28 Jan 2005 08:08:28 -0600, Michael B. Murdock
[EMAIL PROTECTED] wrote:
Pedro,
You can also instruct your users
You need to create different contexts for each company.
- Pedro
On Wed, 2 Feb 2005 21:49:53 -0500, Jason Brown [EMAIL PROTECTED] wrote:
In order to put a shared pbx in an office building for multiple businesses,
I will have to make sure that the caller ID information going out
different callerid's set.
- Pedro
On Wed, 2 Feb 2005 22:31:57 -0500, Jason Brown [EMAIL PROTECTED] wrote:
Pedro
Exactly my point. I have each company in a different context. How do I
SetCallerID to a number based on the context
Cool idea.
One question - let's say someone specifies their home phone number and
their cell number. How do you take into the account if the cell VM
picks up (ie. if cell is out of coverage and VM greeting is played)?
On Fri, 04 Feb 2005 10:41:28 -0700, Kevin P. Fleming
[EMAIL PROTECTED]
! LOL)
I am not holding my breath that this is a viable solution, but was
just wondering your thoughts.
Thanks!
Pedro
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Noah,
Thanks for your input on this. I am not sure if it handles incomng
connections or not - will have to check. I don't think it will work
either - worth a shot to ask though.
Thanks!
- Pedro
On Tue, 8 Feb 2005 10:26:48 -0500, Noah Miller [EMAIL PROTECTED] wrote:
We have a client
Bridged Call SIP/frontdesk-0461ZOMBIE
SIP/frontdesk-0461ZOMBIE (customercontext 100 1 )
Ring Dial SIP/frontdesk|20|t
2 active channel(s)
--
No one is on a call - how can I get rid of this without restarting asterisk?
Thanks!
Pedro
)
--
No one is on a call - how can I get rid of this without restarting asterisk?
Thanks!
Pedro
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).
Any ideas on why a zombie sip channel would occur?
Thanks in advance for any insight on this.
- Pedro
On Thu, 10 Feb 2005 14:57:17 +1300, Matt Riddell
[EMAIL PROTECTED] wrote:
Pedro wrote:
No one is on a call - how can I get rid of this without restarting asterisk?
soft hangup TAB
. But good to know that if it becomes a
problem, I can try upgrading to 1.0.3 or later.
Thanks!
Pedro
On Thu, 10 Feb 2005 09:19:45 +0100, Florian Overkamp
[EMAIL PROTECTED] wrote:
Hi,
-Original Message-
Does anyone know how to kill a zombie channel?
Here is what I see on a show
What is odd is no meetme is being used. But may be related - thanks!
Pedro
On Thu, 10 Feb 2005 14:37:31 +0100, Florian Overkamp
[EMAIL PROTECTED] wrote:
Hi,
-Original Message-
This is the first time I have seen this so it does not appear to
happen too often. Obviously would
Ok this is odd - caught it again twice today. The more I thought
about what has changed on the server I realized that I was not using a
timing device before, but am now using ztdummy. I if that could be
causing the zombies?
- Pedro
On Thu, 10 Feb 2005 08:50:35 -0500, Pedro [EMAIL PROTECTED
call.
I even got an SPA-2100 in hopes that the g729 would sound better on
that unit, but the same issue is present there as well.
Is it just a bad implementation of g729 compression with the Sipura
product line?
Any thoughts or recommendations are appreciated :)
Thanks!
- Pedro
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
uggg.
Is anyone out there having any luck with the SPA-2000 or SPA-2100
using the g729 codec with decent call quality?
On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote:
On Feb 14, 2005, at 1:25 PM, Pedro wrote:
Is it just a bad implementation of g729
,
Pedro
On Tue, 15 Feb 2005 11:35:12 -0600, Eric Wieling [EMAIL PROTECTED] wrote:
Pedro wrote:
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
Actually the SPA-2100 supports 2 g729 channels which is why I bought
it. Unfortunately, the call quality is just as poor on the 2100 as it
is on the 2000.
- Pedro
On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote:
Is it just a bad implementation of g729 compression
, 15 Feb 2005 11:35:12 -0600, Eric Wieling [EMAIL PROTECTED] wrote:
Pedro wrote:
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
? 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
?
-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
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
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.
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
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
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
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
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
) Cisco 7960
Any thoughts are appreciated.
Thank You,
Pedro
TRACI.net
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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
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
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
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
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
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
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
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
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
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
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
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,
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
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
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 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
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 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 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
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
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
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
.
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,
We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using
restrictcid=yes and doesn´t work.
What can I do, thaks
Pedro
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de
[EMAIL PROTECTED]
Enviado el: miércoles, 31 de marzo de 2004 12:00
Yes, my phone company has enabled the Caller ID hiden possibility, thats
because with a Panasonic PBX works fine but with Asterisk not. Thanks for
your aproach, what can I do now?
Regards,
Pedro
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Manuel Wenger
So I basically have park working but when the call gets parked it
doesn't announce the line it parked on.
How can I get this to work?
Pedro
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Hi,
I have one Cisco
IP Conference 7935. Somebudy have any idea how I can config this phone to work woth *.
My *
server is now working with GrandStream Phone and X-Lite SoftPhone, I need to add
this Cisco 7935 but
I dont
know how I can convert to SIP.
Thanks,
Pedro
Mansilla
Hi all,
is it possible to make a queue for outgoing calls? That's for preventing
Device '/dev/ttyI 0' is busy error when having only one line to dialout
and many files in /var/spool/asterisk/outgoing folder. So it would call
only one call at the time and when it's done it would move to next.
is wrong ?
Regards,
Pedro
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?. What can
I do to find the problem ?
Thanks.
Regards,
Pedro
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.
Which can be the problem ?. What can I do to find the problem ?
Thanks, in advance,
Pedro
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Hi,
When have a incoming call from E1 to a extension FXS, and this extension is
busy, the incoming call recive ring tone, and it is wrong. What can I do?
Thanks in advance
Pedro
Here is the trace:
asterisk-1*CLI
Protocol Discriminator: Q.931 (8) len=41
Call Ref: len= 2 (reference 66/0x42
Roger,
Maybe you are using extensions like _9. try to put de complete number in
your estension.conf
ej; exten = _9XXX,1,Dial(.
exten = 101,1,Dial(Zap/1)
in that case send congestion if the 3 digits extensions are not in
extensions.conf.
Regards,
Pedro J. Vela Ruiz
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedro Vela
Sent: Friday, April 23, 2004 7:52 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Fax problem
Hi,
We have a machine with an *'s with Digium TDM400P and connected wit other
machine with *'s an TDM400P
information you need to
help us?, very thanks.
Regards,
Pedro
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group = 1
signalling = bri_net_ptmp
channel = 1-2,4-5,7-8,10-11
Thanks,
Pedro
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Hi,
Maybe I have similar problem. I have a Junghanns.net quadBRI PCI Card in
wiht Telefonica ISDN BRI line, and we have in zapata.conf usecallerid=yes
and hidecallerid=no, but we have not the caller ID.
Can I make some configuration to solve this?
Thanks,
Pedro
-Mensaje original-
De
Hi,
I have a Junghanns.net quadBRI PCI Card with Telefonica ISDN BRI line, and
we have in zapata.conf usecallerid=yes
and hidecallerid=no, but we have not the caller ID.
Can I make some configuration to solve this?
Thanks,
Pedro
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Hi Alf,
Have you got a Junghanns.net quadBRI PCI Card ?
If yes, Have you received CallerID number ? How you have got configured
zaptel and zapata ?
Im collapssed at this point, thaks in advance,
Pedro
PD: maybe... around your question, are you using cdr in csv or in mysql?, if
you are using
directly to ISDN lines and
voice and fax work fine.
Now, we have between ISDN lines and Panasonic PBX the Asterisk, and voice is
ok but fax doesn´t work fine. What can I do?
Thanks,
Pedro
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Hi,
Thanks Tim, we try this and works fine at first page but when the page
graphic dense or more than one page, we have an error.
Un saludo,
Pedro
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Robinson
Tim-W10277
Enviado el: miércoles, 13 de octubre de
Hello,
I have teh same problem with:
QuadBRI - * - TDM400 - Modem
Thanks in advance for your help.
Regards,
Pedro
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Carl Sempla
Enviado el: jueves, 23 de septiembre de 2004 3:56
Para: [EMAIL PROTECTED]
Asunto
I'm currently setting up a PBX system using the T100P card, and was
wondering if it can handle the 2-way trunk type of T1s. Do 2-way trunk
T1s use RBS signaling?
Please excuse my ignorance, I have mostly dealt with PRI B and D channel
type of T1s.
Thanks
Pedro
Just trying to get a feel for how these protocols have progressed and
what is recommended from experience.
Also, what phones work the best.
Thanks for the info
Pedro
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Hi,
Somebody
have any idea how I can config a CISCO IP CONFERENCE
STATION Model 7935 that work with * .
Thanks.
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To
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 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
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
;
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
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
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 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,
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 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
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