19, 2011 10:05:45 PM
Subject: Re: [asterisk-users] FAX on PRI to MFCR2
On Fri, Feb 18, 2011 at 3:23 AM, leonimar cape leo_mac...@yahoo.com wrote:
Hi,
I am having issues sending and receiving fax on my asterisk setup.
Currently I have a server that has 2 x E1 TDM cards one is sangoma
Hi,
I am having issues sending and receiving fax on my asterisk setup.
Currently I have a server that has 2 x E1 TDM cards one is sangoma and the
other
one is openvox. Both support echo cancellation.
One of the e1 is connected to our telco provider via mfcr2 where all our
incoming calls
Hi Group,
I am currently facing a dead end and any help will be much appreciated.
I have an a104d installed in an asterisk box, two of which is configured on
ISDN
pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am
getting one way audio when a local on the hipath
Hi Group,
I was able to resolve the problem by disabling the echo cancellation in a104d
and using the same dahdi config.
Thanks...
- Original Message
From: leonimar cape leo_mac...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 15, 2010 6:12:35 PM
Hi Group,
Is there anyway to force asterisk to use the ip address instead of the hostname
in the sip via header.
Our client's gateway is using a not FQDN as the hostname of their gateway. And
I
am suspecting that the asterisk is dropping the call because it could not
resolve the hostname.
.
Regards,
Mac
- Original Message
From: Gareth Blades list-aster...@skycomuk.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, July 15, 2010 4:45:31 PM
Subject: Re: [asterisk-users] Invalid host name
leonimar cape wrote
Try also installing the devel version of your kernel.
I manage to find the link below:
http://download.openvz.org/kernel/branches/rhel5-2.6.18/028stab064.7/ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm
Try installing it and then recompile the dahdi module.
Hope this helps.
-- Mac
Hi Group,
I have an asterisk running as media gateway with a Digium TE405P 2nd Gen
rev 2 with echo cancellation. It is interconnected to a telco carrier via ISDN
Pri. The voice quality is clear except that sometimes a hear a beep sound that
occure around 5 to 10 secs in the middle of the
Hi Steve,
I notice that everytime I command a reload on the
asterisk my unicall channels gets also reset. So all
going calls got cut-off. Any suggestion that I should
do avoid this problem?
Regards,
Leonimar
--- Steve Underwood [EMAIL PROTECTED] wrote:
Thierry Querette wrote:
Hi
Thanks for the info...
I was able to make if work after recompilation and
relocation of library files.
Regards,
Leonimar
--- Steve Underwood [EMAIL PROTECTED] wrote:
leonimar cape wrote:
Hi Steve,
I need to enable the Unicall channel in my asterisk
box to be able to interconnect
Hi Steve,
I need to enable the Unicall channel in my asterisk
box to be able to interconnect to a local telco
provider using MFCR2. I use the unicall release
unicall-0.0.3pre9 and a patch for asterisk 1.2.
Compilation was done with ease. The problem is that I
got an error Unable to read
Hi to all,
Has anybody in this group encounter this kind of error
in their recompilation of zaptel. My asterisk box has
a for E1/T1 digium card with echo cancellation. I want
to upgrade the zaptel driver. So I downloaded the new
driver from digium and recompile it but I got errors.
Please the
Hi Group,
Just want to asked if some of you have experience 1
way audio. Currently I am using two asterisk box. One
handles the prepaid platform, and the other one is for
media gateway connection. I am using asterisk version
1.2.4 and a 4E1 digium card with echo cancellation. My
interconnection
: leonimar cape [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 14, 2006 9:24 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Adding entries on company
directory
Hi,
I'm really lost on the directory functionality of
the
asterisk. I am currently setting a ACD using
guide me please.
Regards,
Leonimar Cape
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Why don't you use shorewall for your firewall instead.
Works for me.
--- Andres Baravalle [EMAIL PROTECTED]
wrote:
Hi,
I have a problem with asterisks on Linux.
Looks like it is a iptables problem. My external
client (eyebeam, on a
different computer) cannot register to the asterisk
Yap, I have already tried it. But what I want to is to
be able to map a rate table per CID. Do you know how I
can implement it?
Regards,
Leonimar Cape
--- Dovid Bender [EMAIL PROTECTED] wrote:
why not use astcc ? it comes with asterisk and does
all that you have requested. we have scripts
]
[mailto:[EMAIL PROTECTED]
Behalf Of leonimar cape
Sent: Tuesday, March 07, 2006 12:29 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] and H323
Hi Viktor,
What is the version of the asterisk you are using?
You
should use
feasible in A2billing but
the problem is I cannot find a proper documentation of
setting it up. Can anyone show point to the right
direction? Does any one has a better suggestion?
Thank you very much in advance!
Leonimar Cape
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Tired
You can activate the on demand recording on the [EMAIL PROTECTED] by
adding w and W in the asterisk dial command option. It
is located on the general settings under setup. Either
the caller or the called party can initiate the
recording by pressing *1.
Leonimar,
--- Giridhar Bandi [EMAIL
Hi Viktor,
What is the version of the asterisk you are using? You
should use the right version of Openh323 and pwlib to
be able to compile chan_oh323 successfully.
Currently using asterisk 1.2.4 used
openh323-Mimas_patch2-src-tar.gz and
pwlib-Mimas_patch2-src-tar.gz for compiling
chan_oh323.
somehow resets or
crashes after my second reload command.
Has anyone have an idea what causes this?
Regards
Leonimar Cape
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Hi,
Can please someone help me. I have successfully
installed Asteriskathome 2.5 on a server with a Intel
Server Board SE7525RP2. May issue is after placing the
TE405P in the server, it is not booting anymore. Has
anyone in here have the same experience? Can someone
please point me to the right
Hi,
I am using nagios to monitor asterisk via nrpe plugin
and it is working fine. But I want also to monitor the
status of the pri. I have found a plugin which uses
the asterisk manager configuration. But it doesnt
return the result of the command being executed. As
much as I want to include it,
details.
thnx
--- yusuf [EMAIL PROTECTED] wrote:
leonimar cape wrote:
Hi,
I just want to inquire which of the available h323
modules for asterisk is more stable and better
quality. My boss asked me to setup asterisk with
and I
am having a hard time choosing which one should I
Hi,
I have already determine the reason why my incoming
got release after one ring. The telco that I am
connected is waiting for an immediate answer
supervision from my side. Is there anyway immediate
answer supervision be included on the ISDN messages.
Thanks
--- leonimar cape [EMAIL
Hi,
I just want to inquire which of the available h323
modules for asterisk is more stable and better
quality. My boss asked me to setup asterisk with and I
am having a hard time choosing which one should I
used.
Any advice and suggestion will be greatly appreciated
Thanks in advance!
Thanks for the info yusuf... Im gonna check it out...
Cheers!
--- yusuf [EMAIL PROTECTED] wrote:
leonimar cape wrote:
Hi,
I just want to inquire which of the available h323
modules for asterisk is more stable and better
quality. My boss asked me to setup asterisk with
and I
am
Hello,
Can somebody please assist me with my problem.
Currently I am using a [EMAIL PROTECTED] version 2.4 with
a TE406P digium card. One the E1 is connected to a
telco switch via an ISDN. May issue is that may
incoming calls in the zap channels gets disconnected
or release after 1 ring. I really
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