On Wednesday 21 April 2010 17:11:38 Alejandro Recarey wrote:
Thanks Tilghman, this immediatley solved the problem.
Perhaps a mention in cdr_adaptive_odbc.conf that the res_odbc.so
module must also be loaded will help newbies like me ;)
In general, it's a good idea to load all modules that are
the time in the file cdr is right, as mysql. calldate is the time when
the record insert into mysql.
2010/4/22 Alejandro Recarey alexreca...@gmail.com:
Hi all,
I am having a curious problem. I use two cdr backends, csv and MySQL.
These are my settings:
Call Detail Record (CDR) settings
I have installed a fresh installation of AsteriskNOW and have configured
FreePBX. When my users receive a call to their extension the Follow Me
rules call their cell phone. I currently have Call Confirmation enabled.
When the user attempts to press 1 to accept the call they are immediately
When I
comment out the port-parameter (then it defaults to 5060), it is still
the same...
[Apr 22 09:32:49]
--- Transmitting (NAT) to my_pub_ip:5064 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.23:5064;branch=z9hG4bKc46696a2b5;received=my_pub_ip
From: "SIM
3-1"
Hi.
My configuration is Elastix 1.5.2-2 (asterisk 1.4.24, libpri-1.4.3-5,
dahdi-2.1.0.4-7 ) and OpenVox d210e connected to telco provider (Euro ISDN).
Here is my /etc/dahdi/system.conf:
# Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS ClockSource
span=1,1,0,ccs,hdb3
# termtype: te
http://packages.asterisk.org/centos/5/current/x86_64/RPMS/
On 21.04.2010 17:34, David Backeberg wrote:
I didn't know there was an RPM for centos with asterisk in it.
I personally think that's a bad idea. There are a lot of source options.
app_fax.so in particular depends on SpanDSP, and
Un-top-posting...
2010/4/22 Alejandro Recarey alexreca...@gmail.com:
I am having a curious problem. I use two cdr backends, csv and MySQL.
I am finding that the calldate field varies between 3 seconds and 3
minutes between the MySQL database and the CSV files! Is this expected
behaviour?
Hi!
Is there any way to configure a stock Asterisk install to use
wideband mixing or will we have to compile our own?
Not sure!
Look here:
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe#ConfBridge
Philipp
--
_
I appears as though I was a little hasty in saying that it wasn't generating
two calls. It actually was, but I was doing a poor job of searching the logs.
I setup a new-to-me IP 6000 with older firmware on it (3.0.2.0927), and I am
not getting the issue. I am going to start upgrading the
Try reseting the Gateway (soft reset of the settings) and use only IE to do
the setup again. Nothing else comes to my mind.
Also, create a simple extension in Asterisk or if you are using FreePBX you
don't need to tamper with any ports stuff.
-Bruce
On Thu, Apr 22, 2010 at 3:37 AM, Jonas
I'm using
Firefox on Fedora but I don't think the problems lies there.
All goes well when the gateway is connected directly to the internet...
It's when it is behind NAT the 401 is sent from Asterisk...
It must be some NAT-thing combination in how the
GSM-gateway/Zyxel-router sends the
I was able to upgrade asterisk to 1.4.25 and the issue with Find Me Follow
me in FreePBX has been resolved.
Thanks
***
___
Just ask for ASK
Taking the hassle out of technology so you can run your business.
On Thu, 2010-04-22 at 17:45 +0200, Jonas Kellens wrote:
All goes well when the gateway is connected directly to the
internet... It's when it is behind NAT the 401 is sent from
Asterisk...
Is the device registering to an IP address, or do a DNS name? What type
of NAT firewall are you using?
Hi,
Can anybody with previous experience with it guide me on how to setup
asterisk with analog em to connect it to an old style em system which uses
4 pair cables on RJ 45 jacks. All the analog cards I know of use RJ 11
jacks. And there is no choice of modernization of the customer equipment.
Jared,
thank you for your answer.
As I said in my previous mail, I'm using a Zyxel NBG-419 router (which
normally supports VoIP and QoS). Firewall is disabled on the Zyxel.
The MV-374 only accepts IP-address, not a FQDN. Will give it another
try though...
The answer from Portech-support :
On 04/21/2010 07:13 PM, bruce bruce wrote:
How can I find out what the source of the problem is guys?
As I said I didn't change anything, except for making few minor changes
to the firewall today and that was at Amazon firewall level and not
within CentOS.
What causes these bad dahdi_test
Hello asterisk users!
I, like many people, have a cell phone. I also have some SIP phone
devices (software and hardware). I'd like to have one number that
rings all my phones and routes the call to wherever I pick up.
However, my cell phone has its own call forwarding voicemail. I can't
just
Hi!
But it draws attention to me between the PC with softphone and the
telephone I see traffic ARP or ICMP that could make to try between the
equipment but does not see RTP. Is there some special consideration that
it must to observe?
Your English is seriously twisted, making your question
Hi All,
I would like to know if you can confirm that, if using origination via AMI, as
documented here:
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate
it is not possible to set the max duration of a call.
I mean: what you would do with the L (limit) parameter of the
Hi,
currently I am writing a sound recognition software that will suit
here pretty well - it can recognize your cell phone's our of radio
coverage or similar operator message. It's GPL, link here:
http://github.com/Motiejus/SoundPatty
Now the program can say if 2 WAV files match (tested with out
On Thu, 22 Apr 2010, Philipp von Klitzing wrote:
Hi!
But it draws attention to me between the PC with softphone and the
telephone I see traffic ARP or ICMP that could make to try between the
equipment but does not see RTP. Is there some special consideration that
it must to observe?
Your
Hello List,
I need to connect with an Avaya PBX (this part is done), and i would
like to get and send back User-to-User Information (UUI) with the
call. The UUI need because I need to identify the call based on
something witch is available on Asterisk and Avaya too.
It is possible, or have
I have a quick question. I am using Asterisk 1.4. I have a user that has
changed phones (grandstream budge tone 200 to a polycom 330). I have changed
the sip.conf and extensions.conf. I have also unplugged the old phone and
plugged in the new phone. I get the ext showing on the phone, but when
Are you using Databases and/or realtime as opposed to plain-text conf files?
If not, I'd try a restart when convenient. If so, something is hung up.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony LaMear
Sent:
Check out the 'p' option for the Dial command.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
It enables call screening, so you have to press 1 to answer. This can also
prevent the voice mail from being left on your cell phone.
--
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
--
_
-- Bandwidth and Colocation Provided by
On Thu, 22 Apr 2010 15:58:34 -0400
Ryan Bullock rrb3...@gmail.com wrote:
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
Hi Ryan, thanks for your comment.
Unfortunately
One way to do what you want is to create an extension and then in your
originate action use a local change with that extension.
Action: Originate
Channel: Local/allow_caller_id:415111:541222:3...@context
Exten: do_echo
Context: cfmc_cdi_private
Priority: 1
Variable:
Here is how I do it, Mike
-- Perl Code --
my $phone_number=4918802;
my $testfile = /tmp/testin_$$.wav;
unlink $testfile;
my %resp = $astman-sendcommand( Action = 'Originate',
Channel =
DAHDI/$key/w$phone_number,
Take out the router/firewall and connect directly to the net to test your
NAT problem theory.
On Thu, Apr 22, 2010 at 12:15 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
Jared,
thank you for your answer.
As I said in my previous mail, I'm using a Zyxel NBG-419 router (which
normally
I have a list of CLIDs prefixes that I want to use in a context.
Basically, I want to do this but the list of prefix numbers is much longer.
List of prefixes (556,557,557,989.)
[custom-inbound]
exten = _556,1,answer
exten = _556,n,playback(beep)
exten = _557,1,answer
exten =
Use an AGI to do a database lookup and return a value
[custom-inbound]
Exten = s,1,AGI(lookup.agi,${EXTEN})
Exten = s,n,GotoIf($[${AGI_RETURN} = blah}?1:2)
I would recommend the built-in database for simplicity, but for hundreds of
numbers you are going to want either a compiled C agi
Catches 555 through 559:
exten = _55[5-9],1,answer
exten = _55[5-9],n,playback(beep)
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
--
_
-- Bandwidth and Colocation Provided by
As I already
said 2 times earlier in this thread : when I connect directly to the
internet, then the registration goes through normally. So according to
me it is definitely a NAT-problem. Don't need to find this out another
20 times.
Just don't know just what setting is needed when the
Zeeshan Zakaria wrote:
Can anybody with previous experience with it guide me on how to setup
asterisk with analog em to connect it to an old style em system which
uses 4 pair cables on RJ 45 jacks. All the analog cards I know of use
RJ 11 jacks. And there is no choice of modernization of the
Ryan,
Thanks, but as I said, part of the problem is that I can't use DTMF in
my car. So having to 'press 1' is unacceptable.
Bryan Jacobs
On Thu, 22 Apr 2010 15:54:47 -0400
Ryan Bullock rrb3...@gmail.com wrote:
Check out the 'p' option for the Dial command.
Motiejus,
I'm not sure my cell phone plays these - the behavior I observe is that
the call is forwarded to an external number I can control if:
a) The cell phone is out of the service area or off
or
b) I'm busy or reject the call
Currently, I have this number set to my Asterisk
You could use the non-followme option from this link
http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe
and use Lumenvox or Vestec ($50 or $25 for a 1 port license) to be able to
verbally do the 1/yes/2/no thing.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Danny,
That sounds like a decent idea. The dial screening macros are not well
documented and difficult to get right (for example: if one channel
returns BUSY and another returns CONTINUE, what happens?).
I feel that this should be an option built into app_followme - if there
were a
Maybe I'll get brave and try this as a patch :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryan Jacobs
Sent: Thursday, April 22, 2010 4:57 PM
To: asterisk-users@lists.digium.com
Subject: Re:
Thanks Kevin for your reply. We tried this option with two MultiVoIP devices
but results were not satisfactory. I was hoping I could do it without any
external device. My team doesn't want to take any more third party-asterisk
integration risk for this mission critical communication system after
If you're saying the equipment in your car won't generate DTMF tones, a
quick-and-dirty solution would be to use a pocket DTMF dialer.
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryan Jacobs
Sent:
Ah, sorry, I totally missed that in your description.
Other than the speech recognition that Danny is suggesting, my only thought
is to use an agi that will originate another leg, run AMD (answering machine
detect) and then dump the two parties into a conference to re-join them(or
use the Bridge
Zeeshan Zakaria wrote:
Thanks Kevin for your reply. We tried this option with two MultiVoIP
devices but results were not satisfactory. I was hoping I could do it
without any external device. My team doesn't want to take any more third
party-asterisk integration risk for this mission critical
If all the dialplan follow the exact same patten, you may try use
realtime and put the dialplan into mysql.
Just my 2 cents.
bruce bruce wrote:
I have a list of CLIDs prefixes that I want to use in a context.
Basically, I want to do this but the list of prefix numbers is much
longer. List
Don,
No, I'm not trying to say there's a problem with generating the tones.
The issue is that my phone is still holstered, connected to the car via
Bluetooth. I have steering-wheel buttons for receiving calls and
hanging up, but I don't have a safe way to press buttons.
Bryan Jacobs
On Thu, 22
Thank you for this info. I'll look into this equipment and other similar.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-04-22 6:27 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Zeeshan Zakaria wrote:
Thanks Kevin for your reply. We tried this option with two
Hi guys,
I just ran into a funny issue here.
I'm trying to virtualize our asterisk pbx onto vmware esxi. Here's a quick
glance of the system:
* Ubuntu 9.10 i386 with linux-rt kernel (to get 1000Hz timer) everything
up2date.
* Asterisk 1.6.2.6
If I run asterisk using the debian init script
Thanks for the comments, this did the trick :)
On Thu, 22 Apr 2010 13:51:35 -0700
Jim Dickenson dicken...@cfmc.com wrote:
One way to do what you want is to create an extension and then in your
originate action use a local change with that extension.
Action: Originate
Channel:
2010/4/22 Steve Edwards asterisk@sedwards.com:
Un-top-posting...
2010/4/22 Alejandro Recarey alexreca...@gmail.com:
I am having a curious problem. I use two cdr backends, csv and MySQL.
I am finding that the calldate field varies between 3 seconds and 3
minutes between the MySQL
I just ran into a funny issue here.
I'm trying to virtualize our asterisk pbx onto vmware esxi. Here's a
quick glance of the system:
* Ubuntu 9.10 i386 with linux-rt kernel (to get 1000Hz timer)
everything up2date.
* Asterisk 1.6.2.6
If I run asterisk using the debian init script in
On Wed, Apr 21, 2010 at 1:09 PM, Robert Grignon rgrig...@fleetone.com wrote:
I am investigating High Availability solutions for my front end servers.
Always good to hear.
I got into a discussion regarding replicated local databases versus
a single fiber connected shared database on an EMC.
http://shotojukuindia.com/default/index.php
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
bruce bruce wrote:
I have a list of CLIDs prefixes that I want to use in a context.
Basically, I want to do this but the list of prefix numbers is much
longer. List of prefixes (556,557,557,989.)
[custom-inbound]
exten = _556,1,answer
exten = _556,n,playback(beep)
exten =
And I've just done another test. With stock ubuntu 9.10 i386 and sample
asterisk config files, I have the same result. VMWare shows no crazy
stats of disk access nor memory usage. Just 100% cpu load. I selected
Mail Server and OpenSSH server at tasksel screen during
installation.
Same
I opened a ticket about this:
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=17217
Remove -c on the init script of asterisk, line 85. Should help.
I was trying it with a xen guest.
On Fri, Apr 23, 2010 at 6:52 AM, Kelvin Chan
kelvin.c...@positronics.com wrote:
And I've just done
56 matches
Mail list logo
