2014-11-12 2:45 GMT-02:00 Luis Eduardo Cortes :
> Hello:
>
> I'm newbie in asterisk, please help me.
>
> My context is as follows:
>
> 192.168.4.2 --> Asterisk 11.13.1 complied from source
>
> 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway
>
> Whe
Hello:
I'm newbie in asterisk, please help me.
My context is as follows:
192.168.4.2 --> Asterisk 11.13.1 complied from source
192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway
When I call from a GSM cell phone, my TG100 GSM gateway answers and
dials extension (configured as a hotline on
Another question, what audio format I use in MixMonitor to maintain a
connection with reasonable quality and reduce the use of I / O disk? Today
I use wav.
tks
2014-07-24 9:05 GMT-03:00 Eduardo Leones :
> Thank you all for the answers. I will do tests to find the problem.
>
>
n
> you should have your answer.
>
>
> On Wed, Jul 23, 2014 at 4:03 PM, Eduardo Leones <
> edua...@ypytecnologia.com.br> wrote:
>
>> Thanks for the feedback.
>>
>> In this case SSD disks you think it solves?
>>
>>
>> Eduardo
>>
o increase the concurrent calls, or use a storage appliance.
>
> To confirm this, install the tool nmon and use the v and d options to
> bring up the resource usage indicators and drive busy/throughput statistics.
>
>
>
> On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones <
> edua...
Thanks for the feedback.
In this case SSD disks you think it solves?
Eduardo
2014-07-23 18:01 GMT-03:00 Ron Wheeler :
> I would also do some math on the bandwidth requirement.
>
> If you divide your disk bandwidth by your recording bit rate what is the
> theoretical maximum num
people
I have a running Asterisk 1.8.28 in great Dell server with two xeon
processors and 16gb of ram and HD SAS 15k (Raid 1). This server is
recording all calls (placed to record the audio in a ram disk), the entire
CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation
and AG
e
the performace of the asterisk?
tks
Eduardo
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sends a new call for him not
understanding the range (wrapuptime = 30).
My question is if I can somehow make the queue respect the same range with
a call that was not caused by the queue. Is there any way I wrapuptime in a
reset after a manual call?
tks
Eduardo
Josh, thanks for the feedback. That problem can also occur with dynamic
members, would not be just for those who work with realtime?
tks
2014-06-06 10:14 GMT-03:00 Josh Metzger :
> On Fri, Jun 6, 2014 at 9:03 AM, Eduardo Leones <
> edua...@ypytecnologia.com.br> wrote:
>
>&
Guys, I have a problem. I have a queue on asterisk 1.8 that members are
added dynamically via the AMI QueueAdd. When you run the CLI a
"reload app_queue.so" all members who were in the queue disappear. This is
a bug or some parameter that I do not know?
Would have another way to do the reload queu
2014-04-22 3:39 GMT-03:00 binary dreamer :
> hello there.
> sorry if I bring conversation to a different level, but I would suggest you
> a cheaper solution.
> just get a 3G dongle from the list of http://wiki.e1550.mobi
> it is a VERY reliable solution and I have been using it for a few years in
>
Hello:
I have this situation: I can make calls internally, I can make inbound
calls but I can't make outbound calls.
Thanks in advance.
These are my devices:
* asterisk 11.8.1 = 192.168.1.22
* sipphone grandstream gxp2160 = 192.168.1.5
* gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4
por
Hello:
Is that possible?
Do I have to upgrade first to firmware version 6.4?
Does firmware version 6.4 exist?
Thanks in advance.
Regards.
--
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--> http://linuxcounter.net/ <--
skype --> luedcortes
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Hello!
I wonder what the default context that the Queue application uses to call
extensions. If there is a possibility to change this into a context created
by me possible? Would you like to get this load value to variables before
calling the extension.
tks,
Eduardo
Hello,
How do I track the status of an extension for socket? I'm trying to use the
ExtensionState, but it is returning empty.
thank you,
Eduardo
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any limitation of Asterisk to use more hardware resources?
tks
Eduardo
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I have a question about global variables. Is it possible to somehow keep
global variables unset via Dial Plan even Restarting asterisk?
tks
Eduardo
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minutes just to ask for the password of
the agent.
I thought I might be missing hardware, but both the CPU and Memory are with
low consumption.
Does anyone have any idea what happens to the AgentCallBackLogin?
tks,
Eduardo Leones
ntent-Length: 0
>
>
> <>
>
> 15 min (call ended)
>
>
>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asteris
What is your timing reference
"module show like timing"
where your timing counts > 0
I had strange noises with dadhi 2.5 had to roll back to dahdi 2.4
hth
2012/11/17 Valer Nur
> Carlos,
>
> Echo might be a possible cause of the noise but it is strange you hear it
> also on internal calls sin
Danny, thanks for your input...
Can you tell me if I am wrong with the following or give me a brief guide
of what to look at?
I was planning on using Asterisk + chan_sccp to control the VOIP phone.
Asterisk will NOT replace the current CCM/PBX at work, it will have just
one phone but in a way that
perform also called receptive, and is occurring
when they are in an active call, responsive receive calls at the same time. The
application queue does not mean that the extension is a active link.
Does anyone have any idea how to solve this problem?
tks,
Eduardo
?
Thanks,
Eduardo
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asterisk-users mailing list
Hello all,
Does anyone know if E&M over E1 signalling works on top of R2, ISDN and
where can I find a sample Dahdi configuration? Have done a lot of google
and cannot find a proper E1 configuration.
Thanks,
Edu
Maybe you could try sipaddheader but you will need to use sip trunk instead of
iax trunk
Att.
Eduardo
On 15/04/2012, at 16:28, Olivier CALVANO wrote:
> i am search on google ;=) but no result for this moment hihi
>
>
>
>
> Le 15 avril 2012 21:14, Olivier CALVANO
messages, and after beeing able to do the mentioned above, that
the problem is in our side?
Thanks,
Eduardo
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- Non-Commercial Discussion
Enviadas: Sábado, 3 de Setembro de 2011 16:56
Assunto: Re: [asterisk-users] pickup for extension in asterisk 1.4?
On Sat, Sep 03, 2011 at 11:19:12AM -0700, Eduardo Leones wrote:
> I'm not able?to create?a?dial plan?to pull?links?extension.?I am
> using?asteri
asterisk 1.4?
On 11-09-03 02:19 PM, Eduardo Leones wrote:
> Good afternoon,
>
> I'm not able to create a dial plan to pull links extension. I am using
> asterisk 1.4.18with the following dial plan:
>
> exten => _ * 7XXX, 1, Pickup ($ {EXTEN: 2} @ PICKUPMARK)
>
Good afternoon,
I'm not able to create a dial plan to pull links extension. I am
using asterisk 1.4.18with the following dial plan:
exten => _ * 7XXX, 1, Pickup ($ {EXTEN: 2} @ PICKUPMARK)
exten => _ * 7XXX, n, Hangup ()
But is not working, the following error appears in the CLI:
- Execu
1/7/18 Steve Davies
> On 18 July 2011 12:20, Eduardo Carpes wrote:
> > Hello guys
> > I need some help to do works FAX using SIP, anybody know the secret to
> > this? Have asterisk 1.6.
> > Thanks!!
> >
> > --
> > Enviado do meu celular
> >
> >
Hello guys
I need some help to do works FAX using SIP, anybody know the secret to
this? Have asterisk 1.6.
Thanks!!
--
Enviado do meu celular
Eduardo Carpes
E-mail: car...@bsd.com.br
www.freebsd.org
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is T38 passthrough mode (T38-ATA -> Asterisk -> T38 Gateway),
> Asterisk doesn't support T38-Gateway mode.
>
> See
> http://www.voip-info.org/wiki/view/Asterisk+T.38
> and
> http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway
>
> Markus
>
> Am 14.07.2011 19:
Hey guys
I need of some help...
How i know if T.38 is enable on asterisk?
I saw the file /etc/asterisk/sip.conf and
/etc/asterisk/sip_general_custom.conf, both have the in "t38pt_udptl=yes";
Thanks!!
--
Eduardo Carpes
E-mail: car...@bsd.com.br
www.f
10%
De: Matt Riddell
Para: asterisk-users@lists.digium.com
Enviadas: Quarta-feira, 4 de Maio de 2011 0:32:28
Assunto: Re: [asterisk-users] Fading voice problem
On 3/05/11 10:16 PM, Eduardo Leones wrote:
> Guys,
>
> I'm having problems in the fadi
Guys,
I'm having problems in the fading voice calls, receptive and active, that in
SIP
accounts. While few people
using the system, calls are perfect, but it beats the normal use of
connections (average 30 concurrent), the voice begins to fade from people.
Soon I figured some network problem,
Anyone know a good IAX2 softphone for Windows that has g729 and it is free?
att
Eduardo--
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using this slowdown? I thought of
usingthe AddQueueMember, but would have to change much in
design, so is my second choice for solution.
att
Eduardo--
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Any idea ??
Em 28/12/2010 11:32, Eduardo Lobo Blanco escreveu:
Hi list,
I´m using Asterisk 1.6.2.14.
I have some queues configured in my solution, using dynamics members.
All my members are IAX2 clients.
Each member will be at least in two queues at same time.
So i´m trying to use the
eue}_${CALLERID(num)})
exten => s,n,Queue(${queue}65)
Any idea why Asterisk dosnt´play the announce ??
--
Abraços ...
Eduardo Lobo Blanco
Spacecom Ltda.
edua...@spacecom.com.br
(41) 3270-6000 *03
(41) 9101-4450
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val.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Eduardo
> Langoni
> Sent: Monday, August 31, 2009 10:33 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-user
Hi folks!
I have an asterisk, version 1.4.22.2, with dahdi 2.1.0.4 and OpenR2 2.1.1.0.
Sometimes twice a day, sometimes after 3 days, all sip devices looses
registry, but asterisk doesn't show nothing strange, no error log, and
all calls in E1 trunk continue running, but sending to voicemail.
Wha
mmand just dial both numbers using the
dialplan ??
Am I too far here ? or is this something that already exists
and I don't know it ??
I would appreciate any help.
Thanks a lot,
______
Marco Eduardo Cordeiro
Visioncom IT
&
Eduardo Cordeiro
Visioncom
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Syed Nasruddin
Enviada em: terça-feira, 5 de agosto de 2008 09:40
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: [asterisk-users] Queue Penalties not working properly
Hi,
I am
Hello,
Just wanted to let you know that the XP version works fine on vista.
I was working on a similar program but didnt have enough time to finish, I
was working on Delphi 7 btw.
Thanks
Marco.
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Gerald Hars
cial Discussion
Assunto: Re: [asterisk-users] RES: How can I Disable call-waiting
Hello
thank u for ur attention but I did it and in fact its the same as call-limit
in newer versions.
this cmd limit ur call not disable call-waiting.
best regards
On Wed, Jul 23, 2008 at 5:02 PM, Marco Eduardo Cord
Have you tried incominglimit=1 on sip.conf ??
It worked for me, no matter which softphone or ipphone / ATA I use, it
works.
You have to use it inside the configuration for every sip peer, just like
this:
[1002]
Type=friend
Host = dynamic
Port = 5060
incominglimit=1
.
.
.
De: [EMAIL
We installed na Asterisk System whith 400 Softphone users (Eyebeam 1.5 from
Counterpath).
As far as we know, Asterisk don't support yet IM (Instante Message)
feature,instead Eyebeam have this feature.
Is that true? Is there any new version from Asterisk that supports IM?
> Eduardo
There is something which I could do to execute priority 2? It's
possible my agi have programming error?
Eduardo wrote:
Hi all. I've written a AGI in C language. It
receive the asterisk variables to identify the caller. After, it dial
to destination. When caller or the cal
Hi all. I've written a AGI in C language. It
receive the asterisk variables to identify the caller. After, it dial
to destination. When caller or the called hangup the phone, asterisk
returns me '200 result=-1'. For this, asterisk never execute next step,
priority 2. This is very important to m
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We are using MFC/R2 driver successfully in at least
three places in Brazil.
I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hicom
300. I can get a good audio quality with Grandstream, Polycom, and X-Lite
softfones, but SIPURAS and Linksys get a garbled audio, something
Hi!
I'm not familiar with this card, but it seems you are not using the correct
syntax when using command Dial or your * box isn't recognizing the channel
"Zap". Did you compile and install zaptel? _How_? Is Asterisk loading
"chan_zap.so"?. Please, more info!
BR
!
==
Eduardo J. López Martínez
<[EMAIL PROTECTED]>
Isabel Operation
Center <[EMAIL PROTECTED]>
DIT - Dept. Ing. Sist. Telemáticos Tlf: +34 91 3367366 (3036)
UPM - Univ. Politecnica de
Madrid Fax: +34 91 3367333
ETSI
Telecomunicacion
28040 M
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Eduardo
> Kaftanski
> Sent: Thursday, June 23, 2005 11:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] H323 vs OH323 GK registra
the OH323 channel
works much better, sounds better and DTMF works.
Any ideas?
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Eduardo Kaftanski
[EMAIL PROTECTED]
Red Hat Certified Engineer/Instructor/Examiner
Gerente Ingenieria LinuxCenter S.A.
Mariano Sanchez Fontecilla 310, 2do piso, Edificio Birmann24, Las Condes, Chile
http://
l originates from
the pabx.
from which version on is the noSilenceSupression parameter understood?
--
Eduardo Kaftanski
[EMAIL PROTECTED]
Red Hat Certified Engineer/Instructor/Examiner
Gerente Ingenieria LinuxCenter S.A.
Mariano Sanchez Fontecilla 310, 2do piso, Edificio Birmann24, Las Condes, Chil
r?
thanks.
ps. the PABX is a Nec Aspire. anybody can help me configuring it for
Asterisk integration? thanks.
--
Eduardo Kaftanski
[EMAIL PROTECTED]
Red Hat Certified Engineer/Instructor/Examiner
Gerente Ingenieria LinuxCenter S.A.
Mariano Sanchez Fontecilla 310, 2do piso, Edificio Birmann24,
Hi list,
I am trying to install a TDM22B in my * box. I am
using a Pentium III with SuSE 9.2 (2.6.8-24-default) and * 1.0.7. I installed
the TDM22B (no IRQ conflicts), recompiled zaptel, libpri and asterisk. I
modified a file (can’t remember which one) to create my devices entrien
in /d
Cisco http://www.cisco.com/univercd/cc/td/doc/product/
iaabu/pix/pix_62/config/pixclnt.htm). I see a whole bunch of
references to BOOTP but the Polycom manuals are somewhat superficial
in the network related stuff so that one is not able to draw any
conclusions from it.
Thanks,
Eduardo
Hey I had the exact same thing happen to me yesterday. Let me know
how easy the exchange process is. Who did you buy them from?
Thanks,
Eduardo Jimenez
On May 12, 2005, at 8:26 PM, Wilson Pickett wrote:
Hi,
I just got and setup a new ip500 yesterday and it worked for about 15
minutes. Then it
hanged
and the reboot keystroke did not work)….but I think that last thing is
just an issue with everything else not being setup on the phone itself.
Any ideas?
Thanks,
Eduardo Jimenez
Software Engineer
Job Performance Systems, Inc
Ph: 703-683-5805 x 210
www.jps-usa.com
Also,
I
t "Meetme rooms" (although I don't
know if I can archive my goal using it). I only want to dial a new SIP agent
depending the selected room.
How can achieve this? What additional tools will be necessary?
Thanks a lot!
Eduardo.
=========
Hi guys,
I would llike to ask for your help on a problem I'm having with the cdr
functionality. I installed asterisk 1.0.4, and asterisk-addons-1.0.4 and
followed the procedures for installation and mysql configuration. Everything
seems fina. The cdr_mysql module is loaded, and I get no error
dnid'
chan_capi.c:1724: structure has no member named
`dnid'
chan_capi.c: In function `load_module':
chan_capi.c:2793: warning: passing arg 4 of
`ast_channel_register' from incompatible pointer type
make: *** [chan_capi.o] Error 1
taiwan:/usr/
r
Urgent handler
Jan 1 18:58:55 WARNING[29394]: chan_sip.c:694 retrans_pkt: Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 1
(Critical Response)
Calls from the ISDN phone to a softphone WORK.
Can anyone help me??? Thanks in advance ;)
Eduardo.
My configuration is the fo
Hello all,
I'm trying to make phone calls from a softphone through an ISDN line. The
problem I have is that when I try to make a call (outgoing) my ISDN card
does not respond.
The point is that i am being able to make phone calls from an ISDN phone
connected to a ISDN-PBX (the same ISDN-PBX wher
Hello,
You can fix it adding in your "modem.conf":
outgoingmsn=*
I'm not sure is you have to write a "*" or a "0" or simply "andrew".
Then, if you make it work, tell me if your outgoing ISDN calls are working.
I'm having problems whit the
”
using Asterisk by adding two blocks in “sip.conf” and two lines in “extensions.conf”
(I read how to do it somewhere J).
The idea is to make calls from a softphone to the
PSTN and viceversa.
Thanks a lot!
Eduardo.
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imple set of config files to thest an analog modem?
Thanks a lot!!!
Eduardo Lopez
PD.- I have no aditional hardware to thes the ISDN card.
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To U
Hi,
suppose I have agents waiting on a queue and I configure asterisk to dial out
and to forward the call to the first agent enqueued. Asterisk will do it even if
the answer to the call is "busy".
Is it possible to configure asterisk to detect the busy signal and, in that
case, dial another num
Hi,
I installed asterisk-addons and configured it so that the cdr is done on a mysql
database. Everything was fine, until I originated outgoing calls using the
manager API. The call itself is performed perfectly, but when I hangup, I get
the following warning on asterisk CLI:
Aug 31 14:29:23 WA
I'm trying to use the ChangeMonitor command on the asterisk manager API, but I
can't find the syntax anywhere. Asterisk only tells me:
Action: ChangeMonitor
But I don't know the parameters. Can anybody help me?
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Hi,
I just installed astguiclient, following the SCRATCH_INSTALL, without errors.
But when I try to enter the administration page (http://127.0.0.1/astguiclient/
admin.php), it's blank. The browser shows me the following page source:
The same happens with http://127.0.0.1/astguiclient/welcome.
Dear Jimenez,
You have to configure a dial-peer in Cisco box. A 2611 with a NM-HDV-E.
It works. The configuration is something like:
[Cisco]
dial-peer voice 8000 voip
protocol sipv2
codec g711
dest pattern 4... (Whatever says your dialing plan)
session target ipv4:(ip address of you
I have configured several ISDN cards in Brazil in Cisco Routers. There
is a configuration called compand-type (ulaw alaw) (Cisco). They are
different between US and Brazil. The sound is very distorted when in the
wrong configuration. The difference is between 56 bit and 64 bit ISDN.
Maybe that s yo
On Fri, 30 Jan 2004 19:22:21 -0500
Andres <[EMAIL PROTECTED]> wrote:
> Eduardo Goncalves wrote:
>
> >Hi list,
> >
> > I'm with a little problem on my E1 (E&M signaling) link. Every
> > call a
> >make hangs up after 2 or 3 seconds of c
ler
Jan 30 18:46:17 WARNING[81926]: chan_sip.c:486 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Response)
Could someone help?
Thanks in advance
Eduardo
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On Fri, 23 Jan 2004 15:49:31 -0300
"CW_ASN - Gus" <[EMAIL PROTECTED]> wrote:
> < In Brasil, Telefonica offers ISDN, but it's a diferent comercial
> < service (if you want voice and data in your E1), and it's more
> < expensive. If you only want voice, the only choice is R2.
> Very weird, in Arge
eservices... I'm sure
> of that.
In Brasil, Telefonica offers ISDN, but it's a diferent comercial
service (if you want voice and data in your E1), and it's more
expensive. If you only want voice, the only choice is R2.
Small carriers are more flexive and off
igure timing parameters for an X100P?
thanks in advance
Eduardo
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I'm
doing measure tests with SIP and IAX2 trunking. I'll finish today and
post the results.
Thanks for the tips
--
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On Mon, 5 Jan 2004 11:20:08 -0600 (CST)
Brian West <[EMAIL PROTECTED]> wrote:
> Why not use IAX2 trunking you can accomplish the same results with ..
> I tried SIP to SIP with asterisk you must do it without passwords.
Because cisco doesn't compress IAX headers, only rt
On Mon, 05 Jan 2004 10:19:24 -0700
Jared Smith <[EMAIL PROTECTED]> wrote:
> On Mon, 2004-01-05 at 09:24, Eduardo Goncalves wrote:
> > I must use sip, cos we'll use cisco rtp header-compression to
> > save
> > bandwidth.
> >
> > Could
nce I cannot use IAX trunking?
Thanks in advance
Eduardo
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seen before:
asterix:~# lspci
00:03.0 Bridge: PLX Technology, Inc.: Unknown device d00d (rev 01)
any clue?
thanks in advance
Eduardo
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Hi list,
I have a asterisk box with an E400P that was running ok until last
week.
The machine just stop responding and after a reboot, the module (tor2)
doesn't load anymore.
anyone could help?
regards
Eduardo
modprobe returns this:
as
I
try disallow=g729, the same occurs, but this time asterisk shows codec
gsm.
The only way to make a call is allowing only alaw. But this is not
convenience, since i need to use g279 with another endpoint (working
ok).
Why this negotiation problem happens?
Thanks
Ed
Hi list,
Does anyone use the .deb package of asterisk? Is it stable? woks fine?
thanks
Eduardo
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ot;D-channel on span 2 up".
So I stopped asterisk, unload the module, load the module, ant them
started asterisk again. After this, asterisk ran ok.
Could anyone give me a clue on where to look at to discover what
happened?
[ ]'s
Eduardo
erisk
directory or might have problens after asterisk upgrade. So, a run
registration again, this time from /usr/src/asterisk.
But after asterisk upgrade, my sip calls using g729 are not working.
What can I do know to solve this problem?
thanks
Ed
ap.c, Line 6410 (pri_dchannel): Got event No
event (0) on D-channel for span 1
anyone could help?
thanks
Eduardo
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HDLC Abort (6) on D-channel for span 1
Anyone know why?
thanks
Eduardo
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On Tue, 04 Nov 2003 14:38:34 -0500
Brian D Heaton <[EMAIL PROTECTED]> wrote:
> Eduardo,
>
> Hmm, the coax is 75ohm correct? Also, since you are pushing the
> signal
> a little over 250ft you will probably need to set a different LBO
> value in the "spa
--[baloon]---short LAN cable--[*]
Eduardo
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ne = us
defaultzone=us
[ ]'s
Eduardo
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ut 5 meters). Now, asterisk is connected
with a 1.5m cable. I hope this help.
Thanks
Eduardo
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.
> >
My telco checked the circuit last night and didn't find anything
wrong.
Now I'm lost. What should I check to find what's going on?
Eduardo
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ere the second 1 says to use the timing from the incoming E1 line.
My zaptel.conf is like the above, except the crc4. My telco doesn't use
CRC.
Eduardo
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