I have setup a reverse dns for my local subnet and it seems to have resolved
the issue, i was able to make calls even when my asterisk box is not
connected to the net. thanks for all your help!
On Wed, Jun 1, 2011 at 5:57 AM, Hans Witvliet h...@a-domani.nl wrote:
On Tue, 2011-05-31 at 10:29
may i know what domain is asterisk specifically looking for? coz i don't use
domains on the ip phones,
i configure them to register to the IP e.g. 10.10.10.1. forgot to mention
i am using freepbx as a GUI,
does freepbx tells asterisk to look for a specific domain?
TIA.
Regards,
Ron
On Tue,
...@lists.digium.com] On Behalf Of
nhadie ramos
Sent: Tuesday, May 31, 2011 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk fails when DNS or
internet fails
may i know what domain is asterisk specifically looking for?
coz i don't use
Thank you for the information. I will try to install a dns-cache.
Regards,
Ron
On 5/30/11, Alex Balashov abalas...@evaristesys.com wrote:
On 05/30/2011 02:44 AM, gincantalupo wrote:
- do not use urls, only ip addresses in sip.conf
or put your urls inside /etc/hosts (is what I do especially
By the way, is this only an issue for asterisk 1.4? or is it the same with
1.6 and/or 1.8?
TIA.
Regards,
Ron
On Mon, May 30, 2011 at 2:50 PM, Alex Balashov abalas...@evaristesys.comwrote:
On 05/30/2011 02:44 AM, gincantalupo wrote:
- do not use urls, only ip addresses in sip.conf
or put
Hi,
Would just like to inquire why asterisk fails to send calls in / out
when the DNS is failing
or when the server with asterisk has no internet. Ip phones are
connected via IP address and i am using an FXO card, so even if
internet fails i should still be able to make calls thru the fxo. but
to be registered to the same server, or
their client needs to be configured to register to both.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nhadie
Ramos
Sent: Tuesday, July 22, 2008 21:52
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] sometimes extensions
Hi All,
I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on
both asterisk. users register via domain, i have that domain on round-robin.
users can register and sometimes can call each other, but sometimes even if an
extension is register and i tried calling it, i got
, or
their client needs to be configured to register to both.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nhadie
Ramos
Sent: Tuesday, July 22, 2008 21:52
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] sometimes extensions can't be called
Hi All
Hi,
How can i setup conference when i have 2 asterisk servers?
my setup is 2 asterisk servers using realtime, i'm simply using DNS SRV just
for redundancy (not really high availability). i have a web interface, wherein
i can create extension, conference etc.
adding extension is ok, even if
Hi,
Would just like to inquire if anyone here has a setup of asterisk to send
traffic to AS5400 connected to an SS7-PRI. this is more of a AS54 question, as
i've been reading and i always stumble upon PGW2200 as a requirement to handle
SS7 signaling on the AS54. Has anyone able to send calls
Hi,
I don't know what i'm doing wrong but i already reinstalled the system. still
using ubuntu 64-bit.
made sure i had the correct local date time.
then did all this:
ntpdate pool.ntp.org
tzselect , i chose Asia/SIngapore
/etc/timezone is Asia/Singapore
i added TZ='Asia/Singapore'; export TZ to
, June 13, 2008, 12:31 PM
On Friday 13 June 2008 02:35:09 Nhadie Ramos wrote:
Hi,
I don't know what i'm doing wrong but i already reinstalled the
system.
still using ubuntu 64-bit. made sure i had the correct local date time.
then did all this:
ntpdate pool.ntp.org
tzselect , i chose Asia
: Tilghman Lesher lt;[EMAIL PROTECTED]gt;
Subject: Re: [asterisk-users] time on asterisk
To: Asterisk Users Mailing List - Non-Commercial Discussion
lt;asterisk-users@lists.digium.comgt;
Date: Thursday, June 12, 2008, 1:42 AM
On Wednesday 11 June 2008 17:52:15 Nhadie Ramos wrote:
gt; I'm using
: Thursday, June 12, 2008, 8:20 AM
Nhadie Ramos wrote:
gt; Hi Sir,
gt;
gt; I tried restarting asterisk, but still it has the wrong time.
gt;
gt; I tried restarting the system, then start asterisk it still uses the
gt; wrong time.
gt;
gt; I also tried recompiling asterisk, checked i have the correct time
a timezone for a city and usually the correct
daylight parameters are used
nbsp;
Stelios S. Koroneos
Digital OPSiS - Embedded
Intelligence
http://www.digital-opsis.com
nbsp;
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nhadie
Ramos
Sent: Thursday, June 12, 2008 12:00 PM
Hi,
I'm using gotoiftime on asterisk, but it seemsnbsp; there is a difference
between the asterisk time and the system time. could it be because i adjusted
the system timezone on my linux? do asterisk not detect the change of timezone
on the system? How can I fix this prob?
Regards,
nhadie
canceled the call?
Another is if i dial any number, even invalid ones, my script get-total.php
still thinks it is an answered call, so it still does deducting on the balance.
will really appreciate any help.nbsp; TIA.
--- On Sat, 6/7/08, Nhadie Ramos lt;[EMAIL PROTECTED]gt; wrote:
From: Nhadie
for that.
l.
On Sat, 07 Jun 2008 01:25:37 +0200, Nhadie Ramos lt;[EMAIL PROTECTED]gt;
wrote:
gt; Hi,
gt;
gt; How can i get the deadAGI to work at this scenario
gt;
gt; Basically when someonc calls international,amp;nbsp; i will get the
gt; remaining balance using AGI get-available.php.
gt
, June 7, 2008, 12:50 PM
You should use it on the hang-up extension and only after the channel is
technically dead.
It works fine for that.
l.
On Sat, 07 Jun 2008 01:25:37 +0200, Nhadie Ramos lt;[EMAIL PROTECTED]gt;
wrote:
gt; Hi,
gt;
gt; How can i get the deadAGI to work at this scenario
gt
again for all the help.
regards,
nhadie
--- On Sat, 6/7/08, Nhadie Ramos lt;[EMAIL PROTECTED]gt; wrote:
From: Nhadie Ramos lt;[EMAIL PROTECTED]gt;
Subject: Re: [asterisk-users] Question on DeadAGI
To: Asterisk Users Mailing List - Non-Commercial Discussion
lt;asterisk-users@lists.digium.comgt
Hi,
How can i get the deadAGI to work at this scenario
Basically when someonc calls international,nbsp; i will get the remaining
balance using AGI get-available.php.
but after the call i would like to get the usage by calling get-usage.php so i
can update users balance, but looking at the
Hi All,
What could be the cause why there is no audio coming form the participants.
ztdummy is loaded, ZTDUMMY/1 (source: HRtimer) 1.
I can hear Please enter your PIN, User blah blah has enttered...etc etc
But when the particpants talk, we hear nothing. What are the possible mistakes
i did on
Hi,
i will try to setup 3 * box, 1 ser.
if none, let's say i have 4 extensions 101, 102,103 and 104, 101 registered on
* 1, 102 on * 2, 103 on * 3 and 104 on * 1 also.
i will define this dial plan:
[dial-extension]
exten = _1XX,1,Dial(SIP/${EXTEN}) - look it up on the local first
exten =
Hi all,
got some issues with meetme, i created meetme 8000, i have users 200 and 201
dial to 8000, meetme works fine.
but i need an outside user to join, so that user will dial-in via the sip
trunk. sip trunk has did e.g. , outside user dials-in to that DID,
asterisk rcvs it forward
if i have this setup:
[sip users] -- [asterisk] --- [as5300] --- [pstn]
asterisk will talk to as5300 using sip. i will use as5300 as a trunk on the
asterisk so sip users can call out to pstn.
what i would like to is do prepaid on those trunks, not on the sip users. sip
users can call any
). Is that possible?
Regards,
Nhadie
On Tue, 2008-04-22 at 22:59 -0700, Nhadie Ramos wrote:
i want to create a billing system to monitor only the trunks and also
to load amounts on those trunks. is this possible? will i be able to
use app_prepaid for this?
TBH, I don't really understand your
for funds with a prepaid vendor?
Darren Wiebe
[EMAIL PROTECTED]
Brian J. Murrell wrote:
On Wed, 2008-04-23 at 09:38 -0700, Nhadie Ramos wrote:
Hi, sorry to confused you with my question.
the normal prepaid application like astcc, if i'm not mistaken, monitors the
amount left on the user
thank you sir, i will try to check on that. i haven't really tried astcc yet so
i really dont understand how it works right now.
also, do you have any reference on using app_prepaid? can't find some sample
config, i would like to see how i can use that. do you think app_prepaid is
suited for
Hi All,
How can i enable time condition on meetme? below i would like to deny
callers if the time is not yet the scheduled time of the conference, but it
seems like its still goes to 600,2, hope anyone can help.
[meet-me-test]
exten = 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,600,3)
Hi All,
How can i enable time condition on meetme? below i would like to deny
callers if the time is not yet the scheduled time of the conference, but
it seems like its still goes to 600,2, hope anyone can help.
[meet-me-test]
exten =
,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
Nhadie Ramos [EMAIL PROTECTED] wrote: Hi All,
How can i enable time condition on meetme? below i would like to deny
callers
Hi,
I was able to store voicemail following the tutorial
http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage
i would just like to inquire how can i create a web interface (will use php)
to play the voicemail stored in the database.
the field in the database is recording longblob
Hi All,
If i were to develop a softphone, how can i add call transfer, call on
hold and 3-way conference on it? linksys Ip phone has those built-in button
to xfer, conf, on hold.
and x-lite also has those, how can i have those if i develop my own?
Thank You
Regards,
Nhadie
Hi All,
is it possible to choose outbound route by checking the extension of the
caller?
e.g extension that starts with 3 goes to outbound route 1 extension that
starts with 4 goes to outbound route 2. Basically, i'm hosting two(2)
office, extension 3XXX is office 1 and extensions 4XX is
Hi john,
Thank you for your reply, i finally stumbled on google what the problem is.
The driver does not compile on kernel newer than 2.6.19.
Regards,
Nhadie
John Novack wrote:
Sangoma gives EXCELLENT technical support.
I would suggest you try there first.
The few problems I have had with
Hi,
I'm trying to install asterisk(v1.2.22) with FreepBX(v2.2.3) with a
4-Port FXO Sangoma card A200.
I'm using Fedora 7 (x86_64) kernel version 2.6.22.1-27.fc7, but i'm
having these errors:
$ ztcfg -
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart
Hi,
Would just like to ask, I have an SER SIP Proxy and I setup an Asterisk,
i used an SER local as a trunk for the Asterisk.
When the Asterisk box register to SER it will have this URI
sip:[EMAIL PROTECTED], instead of sip:[EMAIL PROTECTED]
Anyone has encountered this problem? Because I'm
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