used? It
would be nice to now the scope and limits of this support
Thanks
2012/11/3 Andrew Latham lath...@gmail.com
On Sat, Nov 3, 2012 at 2:16 PM, Danny Dias ing.diasda...@gmail.com
wrote:
Hello,
I wonder if Digium provides support for Asterisk OpenSource versions as
an
anual fee
Hello,
I wonder if Digium provides support for Asterisk OpenSource versions as an
anual fee or something?
For example, if i download Asterisk 1.8.X (Certified or not...) can i buy
support from Digium to maintain and help on possible future problems in my
configuration?
Thanks
--
It wors. Thanks
El 12/06/2012 22:21, Shaun Ruffell sruff...@digium.com escribió:
On Tue, Jun 12, 2012 at 10:17:46PM +0200, Danny Dias wrote:
It's weird, already installed avahi with yum install avahi
Now the error is:
[Jun 12 16:13:46] WARNING[19949] loader.c: Error loading module
Hi,
I'm just trying to install the DPMA on my Asterisk. I already made the
upgrade from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did:
*mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185
*
*compiling Asterisk-Cert2 1.8.11*
*./configure
make
make install
make config
*
Afther
Thanks Jason,
I didn't see in any document with an advice of packages needed. And yes,
i did open a case yesterday, no answer yet!
BR
2012/6/12 Jason Parker jpar...@digium.com
On 06/12/2012 02:56 PM, Danny Dias wrote:
Hi,
I'm just trying to install the DPMA on my Asterisk. I already
'res_digium_phone.so'
could not be loaded
Something related to voicemail?
Thanks
2012/6/12 Danny Dias ing.diasda...@gmail.com
Thanks Jason,
I didn't see in any document with an advice of packages needed. And yes,
i did open a case yesterday, no answer yet!
BR
2012/6/12 Jason Parker jpar
Hello,
I would like to know the difference between encrypt the rtp and signaling
between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm
trying to understand whether an SBC could fit an Asterisk deployment
Thanks
--
That's my question...the sbc provides security over trunking, right? The
same can do Asterisk or a Proxy..isn't? Does an SBC can provide any kind of
add-value to an Asterisk deployment?
El 11/06/2012 20:20, Paul Belanger paul.belan...@polybeacon.com
escribió:
On 12-06-11 02:06 PM, Danny Dias
Hi Tim,
Unfortunately i can't reproduce the scenario because it was a long time
ago. But it would be nice to hear from you, what things can be verified
within fax and Asterisk? Any TIP on wireshark monitoring?
El 31/05/2012 03:08, Tim Nelson tnel...@rockbochs.com escribió:
- Original
I've used Asterisk 1.4.22 with hylafax and iaxmodem and it wasn't reliable
at all; sometimes the fax reach the destination, sometimes not, and even
worse, asterisk got froozen...(here using analog lines over Sangoma B600
and Digium TDM400P, same behavior with both.
Other history with same
Just to clarify, i were using fax machines connected to fxs ports
El 30/05/2012 20:31, Danny Dias ing.diasda...@gmail.com escribió:
I've used Asterisk 1.4.22 with hylafax and iaxmodem and it wasn't reliable
at all; sometimes the fax reach the destination, sometimes not, and even
worse
Hello,
For those customers with only analog lines, who ask for fax2email and
email2fax, whats the most reliable solution available and tested with
Asterisk?
Thanks
--
_
-- Bandwidth and Colocation Provided by
with analog lines, that would
be better to use. Could you please confirm? any place to check How-To on
Hylafax and Iaxmodem?
Many thanks!!!
2012/5/29 Carlos Alvarez car...@televolve.com
On Tue, May 29, 2012 at 8:03 AM, Warren Selby wcse...@selbytech.comwrote:
On Tue, May 29, 2012 at 3:10 AM, Danny Dias
I did not understand. What do you mean with renumber all the messages?
El 25/05/2012 02:27, Edwin Lam edwin@officegeneral.com escribió:
On 5/23/12 2:42 AM, Danny Dias wrote:
Can i delete like this:
rm -rf /var/spool/asterisk/voicemail/**voicemailcontextcustomer/300/**
INBOX
23, 2012, at 1:03 AM, Danny Dias wrote:
Thanks Jason,
But how to delete them? there are a lot of old voicemails, but i don't
want to break the app_voicemail.
2012/5/22 Jason Parker jpar...@digium.com
On 05/22/2012 04:54 PM, Danny Dias wrote:
There are 4 files for each voicemail
Hello,
I was checking how to DELETE old voicemail from Asterisk, for my extension
300, i have 20 MB
[root@pbx INBOX]# pwd
/var/spool/asterisk/voicemail/default/300/INBOX
[root@pbx INBOX]# du -s -h
20M
There are 4 files for each voicemail:
msg.gsm
msg.txt
msg.wav
msg.WAV
I've
Thanks Jason,
But how to delete them? there are a lot of old voicemails, but i don't want
to break the app_voicemail.
2012/5/22 Jason Parker jpar...@digium.com
On 05/22/2012 04:54 PM, Danny Dias wrote:
There are 4 files for each voicemail:
msg.gsm
msg.txt
msg.wav
Hello,
I have a question regarding DPMA for Digium Phones, if i install the DPMA
on my Asterisk Server A, and then, i move the phone to register into
another Asterisk Server B, can i install for free another DPMA license
for my digium phones on this second server? can i move the DPMA from one
2012/5/21 Danny Dias ing.diasda...@gmail.com
Hello,
I have a question regarding DPMA for Digium Phones, if i install the DPMA
on my Asterisk Server A, and then, i move the phone to register into
another Asterisk Server B, can i install for free another DPMA license
for my digium phones
Thanks Kevin. Buying one for Spain right now ;)
2012/5/15 Kevin P. Fleming kpflem...@digium.com
On 05/12/2012 12:07 PM, Danny Dias wrote:
What about the Database and recording calls replication? as i could see,
the RSeries does not take into account these data.
The Digium R-series
Thanks,
What about the Database and recording calls replication? as i could see,
the RSeries does not take into account these data.
Thanks
2012/5/12 Kevin P. Fleming kpflem...@digium.com
On 05/11/2012 10:46 PM, Danny Dias wrote:
Hi,
I would like to know if the servers (A and B) could use
Hi,
I would like to know if the servers (A and B) could use boards non-digium
with the R-Series HA product from Digium, i have a couple of B600E Sangoma
to put on each server and use the R-series to provide HA. Is that possible?
Thanks
--
www.danntel.net
*sip:danny4...@thesipschool.com*
Does the D40 will support the option to develope apps? As i could see on
videos only the D70 has the apps button, and also, the lcd screen is
smaller. Right?
Enviado desde mi Samsung Galaxy S II
El 10/05/2012 12:44, Kevin P. Fleming kpflem...@digium.com escribió:
On 05/09/2012 08:38 PM, Danny
Hello,
Im looking to buy a digium phone D70 unit just for testing on lab; to
really understand the phone and features.
I cant find any website with opinions; any here? Are they really valuable
to the price? (D70 quite expensive)
Does the SDK for building apps is usable? Can you build powerfull
Ok understood. The signaling wont be a problem, but not the same with rtp
as it uses randomly ports. The idea is to have an intermediary who could
delivers both ports and ping them to both sides to keep nating open on
routers, this is what i do with rtp proxy within opensips.
But in this case no
Did you asked OpenVOX for support?
El 27/04/2012 01:48, John Millican j...@millican.us escribió:
Hello,
I have an OpenVox A400E02 (2FXO) in a box running Debian 6.0.2 running
Asterisk 1.8.6.0. I have to POTS line on it from Verizon in Virginia, USA.
Whenever I place a call to one of the two
Btw, red alarms means phisical problemscheck cable first.
El 27/04/2012 10:23, Danny Dias ing.diasda...@gmail.com escribió:
Did you asked OpenVOX for support?
El 27/04/2012 01:48, John Millican j...@millican.us escribió:
Hello,
I have an OpenVox A400E02 (2FXO) in a box running Debian
on the
firewall also you should set the phone to send a nat keep alive each 30
seconds (asterisk also sends a options packet to keep the nat open but
doesn't always work ok )
-Original Message-
From: Danny Dias ing.diasda...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date
Hello,
I have a doubt (basic i guess, but not for me). I have an escenario where
customer site has Asterisk PBX behind Nat/firewall with private IP address
and sone phones also; BUT there are some other phones on different sites
and of course behind its nat/firewalls; with IAX i have no problem,
behind nat
El 26/04/2012 19:31, Carlos Alvarez car...@televolve.com escribió:
On Thu, Apr 26, 2012 at 9:54 AM, Danny Dias ing.diasda...@gmail.comwrote:
I have a doubt (basic i guess, but not for me). I have an escenario where
customer site has Asterisk PBX behind Nat/firewall with private IP
Does not work for me!
El 26/04/2012 20:14, Carlos Alvarez car...@televolve.com escribió:
On Thu, Apr 26, 2012 at 10:47 AM, Danny Dias ing.diasda...@gmail.comwrote:
I cant put public ip adress to the asterisk server.
The main problem i see is with the sip headers (contact, sdp ip and
ports
not
had any crashes since applying the patch.
ABE-2147
How can i apply this patch on my asterisk versions: 1.4.24.1 and
1.4.23.2? do i have to apply this patch manually?
Thanks in advance for your help
--
Ing. Danny Dias
www.DannTEL.net
this patch for several weeks and has not
had any crashes since applying the patch.
ABE-2147
How can i apply this patch on my asterisk versions: 1.4.24.1 and
1.4.23.2? do i have to apply this patch manually?
Thanks in advance for your help
--
Dog is my Co-pilot
--
Ing. Danny Dias
2010/11/22 John Novack jnov...@stromberg-carlson.org
Danny Dias wrote:
Hello John,
What i am asking is if i can apply this patch manually or something like
this without making any upgrade of Asterisk, has anyone done this before?
I can't answer that question.
ummm why
Hello friends,
I'm trying to make a simple call from asterisk CLI, but is quite confuse
i followed the information here:
http://www.voip-info.org/wiki/view/Asterisk+CLI+dial
and changed my extensions.conf like this:
alsa.conf
[general]
autoanswer=no
context=consolecontext
extension=100
By
Zakir,
Have you checked the RFC3261?
21.4.2 401 Unauthorized
The request requires user authentication. This response is issued by
UASs and registrars, while 407 (Proxy Authentication Required) is
used by proxy servers.
2010/10/20 Zakir Mahomedy z...@mayfair2000.com
Hi
I am trying to get
[:port][/extension]
2010/10/20 Danny Dias ing.diasda...@gmail.com
Zakir,
Have you checked the RFC3261?
21.4.2 401 Unauthorized
The request requires user authentication. This response is issued by
UASs and registrars, while 407 (Proxy Authentication Required) is
used by proxy servers.
2010
Hello Asterisk Community,
Is there a way to check in asterisk cdrs and extension forwarded?
I mean, i'm calling to a ISDN number, wich goes to extension 8222, but
this extension is forwarded to another one, the problem is that in
CDRs i am able to see the the first step of the call, but never
, 2010 at 1:02 PM, Danny Dias ing.diasda...@gmail.comwrote:
Hello my friend Ingmar,
I would like to know the cable you used? how was the connection? i'm using
this one:
http://wiki.sangoma.com/Pinouts#A108 Loop Back
Is this ok? what should i do my friend, my problems are understand the
fisicall
...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias
*Verzonden:* vrijdag 24 september 2010 11:05
*Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
*Onderwerp:* [asterisk-users] How to test BIG traffic through
DAHDI/WANPIPEinterfaces
Hello
Hello,
I'm getting a KErnel Pannic every time i restart the server, what could be
happening?
I just make: shutdown -r now and the server gets Kernel Panic. I'have to
go on site and press the power button
Here you have my sotware versions:
Asterisk 1.4.24.1
DAHDI Tools Version - 2.1.0.2
DAHDI
Thanks Tim
That solved my problem, thank you very much...but now i'm having another
problem, when the server starts, it doesn't start asterisk automatically,
should i change the start script?
2010/9/30 Tim Nelson tnel...@rockbochs.com
- Danny Dias ing.diasda...@gmail.com wrote:
I'm
Hello,
I'm experiencing some weird problems on my server:
- 1) The following messages are filling up my logs:
[Sep 29 08:24:59] WARNING[7077]: chan_dahdi.c:2789 pri_find_dchan: No
D-channels available! Using Primary channel 140 as D-channel anyway!
[Sep 29 08:24:59] WARNING[7078]:
-a
Linux Sangoma-Testing 2.6.26-2-amd64 #1 SMP Thu Sep 16 15:56:38 UTC 2010
x86_64 GNU/Linux
r...@sangoma-testing:/home# uname -r
2.6.26-2-amd64
2010/9/28 Paul Belanger paul.belan...@polybeacon.com
On Mon, Sep 27, 2010 at 6:57 PM, Danny Dias ing.diasda...@gmail.com
wrote:
r...@sangoma-testing
You have to increase the time of expiration for the Register...on linksys
devices is located on Proxy and Registration section under the EXTN: (Where
N is the extension number)
Try putting this to: 3600
To check wheter or not is loosing Register, try with ngrep-sip and check it:
ngrep -p -q -W
Hello,
I'm checking this:
[Sep 28 13:32:46] NOTICE[30360] chan_zap.c: PRI got event: HDLC Abort (6) on
Primary D-channel of span 1
[Sep 28 13:32:46] NOTICE[30363] chan_zap.c: PRI got event: HDLC Abort (6) on
Primary D-channel of span 4
[Sep 28 13:32:46] NOTICE[30363] chan_zap.c: PRI got event:
:456...@sip.mydomain.com sip%3a456...@sip.mydomain.com
.
Call-ID: c9bd8b57-f7bdc...@192.168.1.52.
CSeq: 116228 REGISTER.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Content-Length: 0.
On 9/28/10 7:24 PM, Danny Dias
Hello,
I'm trying to compile DAHDI on DEBIAN but i have the following error:
r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make
echo You do not appear to have the sources for the 2.6.26-2-amd64 kernel
installed.
You do not appear to have the sources for the 2.6.26-2-amd64 kernel
installed.
installed on every install.
This link should help:
http://www.cyberciti.biz/faq/howto-install-kernel-headers-package/
Dean Hoover
Milwaukee, Wisconsin
On 9/27/2010 11:09 AM, Danny Dias wrote:
Hello,
I'm trying to compile DAHDI on DEBIAN but i have the following error:
r...@sangoma-testing
installed.
You do not appear to have the sources for the 2.6.26-2-amd64 kernel
installed.
exit 1
make: *** [modules] Error 1
The same result :(
2010/9/27 Daniel Tryba dan...@tryba.nl
On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote:
r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4
¿ahhh?
2010/9/27 Roger Burton West ro...@firedrake.org
On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote:
What should i do?
aptitude install module-assistant
m-a a-i dahdi
--
_
-- Bandwidth and Colocation
I've these versions of DAHDI running into another Server with no
problem...it seems to be a problem with dependencies, but i can't find the
trick :(
2010/9/27 Paul Belanger paul.belan...@polybeacon.com
On Mon, Sep 27, 2010 at 12:09 PM, Danny Dias ing.diasda...@gmail.com
wrote:
What should i
paul.belan...@polybeacon.com
On Mon, Sep 27, 2010 at 12:09 PM, Danny Dias ing.diasda...@gmail.com
wrote:
What should i do?
Try with the lastest DAHDI version, 2.4.0.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode
-la /usr/src/linux
lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux -
linux-headers-2.6.26-2-amd64
Seems to be OK, isn't?
Thanks!
2010/9/27 Paul Belanger paul.belan...@polybeacon.com
On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias ing.diasda...@gmail.com
wrote:
The same problem!
What
./configure was run? If so redo the
./configure command and see what that does.
--
Jim Dickenson
mailto:dicken...@cfmc.com dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 27, 2010, at 3:57 PM, Danny Dias wrote:
Hello Paul,
Here is the output of the commands:
r...@sangoma-testing:/home
: cannot access /usr/src/linux: No such file or directory
Is that Ok?
2010/9/28 Danny Dias ing.diasda...@gmail.com
Hello Paul,
Here is the output of the commands:
r...@sangoma-testing:/home# ls -la /lib/modules/
total 12
drwxr-xr-x 3 root root 4096 2010-09-24 10:21 .
drwxr-xr-x 13 root root
Hello Community,
I need to test or simulate many calls through dahdi/wanpipe, i have a
Sangoma A108D, and i need to test the stability of the
card/drivers/firmwares with a test environment, do you think is possible?
What should i do? using some loopback cable maybe?
Thanks in advance
DD
--
*Danny Dias
*Verzonden:* vrijdag 24 september 2010 11:05
*Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
*Onderwerp:* [asterisk-users] How to test BIG traffic through
DAHDI/WANPIPEinterfaces
Hello Community,
I need to test or simulate many calls through dahdi/wanpipe, i
the diration of the call to can load the box very well.
Danny Dias wrote:
ummm but how do you do that?
SIPp is only for SIP calls...i need to check in some way the dahdi
driver, i need in someway stress de card, is that possible? may be it
has no sence at all :(
Thanks!
2010/9/24 Ingmar
Any hints please?
I would appreciate your valuabl help
Thanks
2010/9/16 Danny Dias ing.diasda...@gmail.com
Hello my friends,
I would like to understand the output from pri intense debug span X, the
Telco always says that their side is OK, but i always receive alarms,
loosing connection
Hello my friends,
I would like to understand the output from pri intense debug span X, the
Telco always says that their side is OK, but i always receive alarms,
loosing connection, take a look:
[Sep 16 13:18:19] WARNING[30364] chan_zap.c: Detected alarm on channel 1:
Recovering
[Sep 16 13:18:19]
Hello,
I'm having some problems with a total SIP Asterisk scenario, some extensions
when make internal and outgoing calls can't hear very well the other party,
not echo, not packet lostthe problem is that the volume seems to be very
low...what could be happening? i'm not sure what to check
Yes my friend...CONFIRMED!!! G729 on both sides
2010/9/15 Ishfaq Malik i...@pack-net.co.uk
Have you checked that the codec order on the phone matched the order set
on the server?
On Wed, 2010-09-15 at 17:04 +0200, Danny Dias wrote:
Hello,
I'm having some problems with a total SIP
Hello Adriá...
We are using Linksys 942, softphones Xlite...it's a macro pbx, with almost
1000 users, we've checked the gain and volume on the phones :(
2010/9/15 Adrià Vidal adriavi...@gmail.com
On Wed, Sep 15, 2010 at 6:08 PM, Danny Dias ing.diasda...@gmail.comwrote:
Yes my friend
Thanks Sebastian,
It's the same firmware version for all our linksys phones...and we have
hundreds of pbx's runnning this firmwares versions without any problem
2010/9/15 Sebastian s...@open-t.co.uk
Hi,
On 09/15/2010 04:04 PM, Danny Dias wrote:
Hello,
I'm having some problems
please?
Thanks!
2010/9/10 Moises Silva moises.si...@gmail.com
On Thu, Sep 9, 2010 at 11:08 AM, Danny Dias ing.diasda...@gmail.comwrote:
Thanks Kevin,
But today i saw a Kernel Panic into my server, for no any apparent
reasondoes
this parameter could help: pci=routeirq
By the way, we
Thanks Miguel,
Excellent TIP! :)
I will try and let you know
Best Regards!
2010/9/10 Miguel Molina mmol...@millenium.com.co
El 10/09/10 03:14, Danny Dias escribió:
There used to be a problem with some Dell servers though, but that
was already fixed some weeks ago.
HEllo Moises
Hello Asterisk community,
I'm experiencing some problems with a Digium TE4XXP, the thing is that i'm
sharing IRQ with some megasas device:
169: 69917985 0 0 0 0 0
0 0 IO-APIC-level megasas, wct4xxp
I've been searching here:
Thanks Kevin,
But today i saw a Kernel Panic into my server, for no any apparent
reasondoes
this parameter could help: pci=routeirq
By the way, we are using DELL servers, i've also used Sangoma, and always
the same problem
Thanks!
2010/9/9 Kevin P. Fleming kpflem...@digium.com
On
Any particular reason you don't want to put the logic of the macro in your
AGI?
Yes...i've no idea how to do it...it's a PERL script, i'm already checking
how to do this...but it will be a little complicated :(
2010/9/3 Steve Edwards asterisk@sedwards.com
On Thu, 2 Sep 2010, Danny Dias
);
}
#
#
By the way, is it necessary to Hangup the Macro if the AGI is already doing
this?
BR ;)
2010/9/3 Steve Edwards asterisk@sedwards.com
On Thu, 2 Sep 2010, Danny Dias wrote:
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro
from my AGI
Hello community,
I need to finish an AGI script when it invokes a macro from dialplan, how
can i do that? it's quite confusing...the macro is making a hangup but the
script continues
Thanks
--
Salu2
--
_
-- Bandwidth and
)
exten = forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden)
What should i do to finish the macro if this macro reachs the Hangup?
Thanks for your help my friend!
2010/9/2 Steve Edwards asterisk@sedwards.com
On Thu, 2 Sep 2010, Danny Dias wrote:
I need to finish an AGI script when
What should i do to finish the macro if this macro reachs the Hangup?
I tried to say: What should i do to finish the *AGI* if this macro reachs
the Hangup?
2010/9/2 Danny Dias ing.diasda...@gmail.com
Hello Steven...
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro
Nicholas da...@debsinc.com
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Dias
*Subject:* Re: [asterisk-users] How to finish an AGI
snip
This isn’t really a task for AGI since it is by nature single-call
specific
YES YES...that's what i want ;)
so simple but i was so tired :(
I will try it and let you know ;)
THANKS my friend
2010/9/2 Danny Nicholas da...@debsinc.com
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Dias
Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Saludos
Danny Dias
--
_
-- Bandwidth
'
-= Registered Asterisk Dial Plan Hints =-
8340 at pbx9: SIP/8340 State:Idle
Watchers 0
- 1 hints registered
And phone does not show any light with the the extension 8349 in use...
Thanks in advance for your help
--
Saludos
Danny Dias
Asterisk Dial Plan Hints =-
8...@pbx9: SIP/8340 State:Idle
Watchers 0
- 1 hints registered
And phone does not show any light with the the extension 8349 in use...
Thanks in advance for your help
--
Saludos
Danny Dias
Thanks as always Zeeshan ;)
I've changed my configuration, take a look:
[8250]
type=friend
callerid=Extensión 8250 8250
canreinvite=no
context=pbx9
dtmfmode=rfc2833
host=dynamic
insecure=no
language=es
nat=yes
pickupgroup=
callgroup=
qualify=2000
secret=cyx2mo
type=friend
username=8250
Hello Asterisk community,
I'm trying to use BLF with Asterisk Realtime, i've been searching for
some info but nothing seems to be clear, can anyone help me eith some
ideas to make this work ok?
I'va my dialplan with Realtime
Thanks in advance
--
Saludos
Danny Dias
SkypeID: danny.dias1
manually every
time i reboot the machine (my laptop for testing)
So, what should i do in order to solve this situation?
Thanks in advance
Regards
--
Saludos
Danny Dias
SkypeID: danny.dias1
--
_
-- Bandwidth and Colocation
to make this work better?
Regards
--
Saludos
Danny Dias
SkypeID: danny.dias1
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
this work ok, what should we do?
Thanks in advance
--
Saludos
Danny Dias
SkypeID: danny.dias1
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? Please your
help, we really need to put this working
Thanks in advance for all your help!
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Saludos
Danny Dias
SkypeID: danny.dias1
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New to Asterisk
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5ad99e891003180820l56882922gd84102d158918...@mail.gmail.com
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Do you properly hang up the calls. Does 'zap show channel channel
number'
shows that the channel is 'on hook' after its hang up?
On 2010-03-18 10:06 AM, Danny
, 2010 at 5:00 PM, Danny Dias ing.diasda...@gmail.com
wrote:
This digium card has 3 FXO ports and 1 FXS port where we have a fax
machine
connected!
The problem is that we can receive fax very good, but we can't make any
outbound fax call, in fact, our asterisk get freezed in this case
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x2saa4c40ff1004091730p192f37det33a5283a4ca85...@mail.gmail.com
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On Fri, Apr 9, 2010 at 5:17 PM, Danny Dias ing.diasda...@gmail.com
wrote:
Hello my
!
Message: 9
Date: Fri, 9 Apr 2010 19:22:05 -0430
From: Danny Dias ing.diasda...@gmail.com
Subject: [asterisk-users] Problems with Fax over TDM410P
To: asterisk-users@lists.digium.com
Message-ID:
y2l5a64fbaa1004091652k8393c88anf30c96809f8a9...@mail.gmail.com
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Hello my friends...
We are having some problems with the fax in our asterisk server...
We have:
Asterisk 1.4.21.2
Zaptel Version: 1.4.11
libpri version: 1.4.5
Digium Card TDM 410P
This digium card has 3 FXO ports and 1 FXS port where we have a fax machine
connected!
The problem is that we can
Hello my friends,
I want to make fax work in the following scenario:
My versions are:
Asterisk 1.4.21.2
WANPIPE Release: 3.4.7
Zaptel Version: 1.4.11
libpri version: 1.4.5
Digium Card TDM 410P
The E1 pri is connected to our Sangoma A102DE, we also have a SIP
Mediant Audiocodes 1000 where we
://en.wikipedia.org/wiki/Tux
On Tue, Mar 23, 2010 at 7:16 PM, Danny Dias ing.diasda...@gmail.com
wrote:
Hello my friends,
I'm very worry about a problem i'm having...my asterisk got freez some
times, every 5 or 6 days with NO trace in /var/log/asterisk/messages
What i want to know
://en.wikipedia.org/wiki/Tux
On Tue, Mar 23, 2010 at 7:16 PM, Danny Dias ing.diasda...@gmail.com
wrote:
Hello my friends,
I'm very worry about a problem i'm having...my asterisk got freez some
times, every 5 or 6 days with NO trace in /var/log/asterisk/messages
What i want to know is if safe_asterisk
:
626964fc1003240209i6a48bd5cj93fbacd25e2cf...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
@Danny: How do you start your Asterisk ?
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Regards,
Prince Singh
Drishti-Soft Solutions Pvt Ltd
On Wed, Mar 24, 2010 at 6:35 AM, Steve Edwards asterisk@sedwards.com
wrote:
On Tue, 23 Mar 2010, Danny
Hello my friends...
Currently we are using the following firmware versions on ours aastra 55i:
Firmware Information
Attribute Value
Firmware Version 2.1.0.2145
Firmware Release Code SIP
Boot Version 2.0.1.1055
Date/Time Jun 20 2007 06:20:29
Can we make a firmware upgrade to the latest one:
Thanks Zeeshan,
In fact,i have RealTime configured and working...
What i want is to make an upgrade of libpri and wanpipe at least, asterisk
and zaptel will be like i have now...
Do you think that recompile/upgrade this softwares version will produce a
problem? what steps should i do?
Is it
Hello my friends,
I'm very worry about a problem i'm having...my asterisk got freez some
times, every 5 or 6 days with NO trace in /var/log/asterisk/messages
What i want to know is if safe_asterisk has something to be with this?
This is what i have on my server:
[r...@mypbx ~]# ps -A | grep
Hello my friends,
I want to make upgrades for all my software, currently i have the following
versions:
Asterisk 1.4.21.2
Zaptel Version: 1.4.11
WANPIPE Release: 3.4.7
libpri version: 1.4.5
I want to make upgrade for the last version of Asterisk 1.4, the last
version of Zaptel (dahdi will be
Thanks Zeeshan,
SAngoma told me that the asterisk problem is unrelated to wanpipe drivers,
they told me to reinstall asterisk again
But, i still having doubts about the problem :(
Thanks in advance
Message: 10
Date: Thu, 18 Mar 2010 00:21:06 -0400
From: Zeeshan Zakaria zisha...@gmail.com
...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Do you properly hang up the calls. Does 'zap show channel channel number'
shows that the channel is 'on hook' after its hang up?
On 2010-03-18 10:06 AM, Danny Dias ing.diasda...@gmail.com wrote:
Thanks Zeeshan,
SAngoma told me
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