Would it be possible to see an example on extensions.conf and voicemail.conf
to see how to do that?
Thanks in advance,
Jonathan
On Mon, Jun 14, 2010 at 10:18 PM, Zeeshan Zakaria wrote:
> You can use realtime architecture. I have a similar setup, voicemails works
> just fine.
>
> Zeeshan A Zakari
By the way, I am currently testing this product from Skype. I would
like to be able to receive calls ona Skype name on our pbx.
1) It works beautifully and you don't have to do anything in particular.
2) It's disproportionally expensive which is why I want Skype for
Asterisk to work.
SfS costs $
I understand that SfA is a binary module? There are processors it will
not work on, correct? Are there limits as to operating system or
distros?
tia,
/r
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
you see lot of documentation on wiki
Google them many success case you see
Ram
On Tue, Jun 15, 2010 at 7:01 AM, Landy Landy wrote:
> Hello List.
>
> I just installed a2billing with asterisk 1.6 and got it working. The only
> problem is that I'm trying to setup something to manage who's using th
On Mon, 14 Jun 2010, bruce bruce wrote:
However, for this project, it seems that I can use php system() along
with grep to see the status of a peer with one line of code:
asterisk -rx "sip show peer $sip_peer" | grep -c "X-Lite"'
Above ^^^ in Linux prompt returns 1 if $sip_peer is registered
Hello List.
I just installed a2billing with asterisk 1.6 and got it working. The only
problem is that I'm trying to setup something to manage who's using the most
minutes in the house. I noticed a2billing only works for callin cards setups,
or maybe I didn't configure it correctly for what I wa
I do have a rather bigger project coming my way and I would really like to
know how to do the *"feeding a text file into the STDIN of your AGI so you
can debug completely external to Asterisk"*
*
*
*However, for this project, it seems that I can use php system() along with
grep to see the status of
On Mon, 14 Jun 2010, C F wrote:
> One more thing, read the comments here:
> http://www.voip-info.org/wiki/index.php?page_id=573&tk=2ff846f8169b7694aed5&comments_page=1
> Don't forget to have a beer ready :P
Now that's really funny.
I read along with this and was thinking this was exactly my expe
One more thing, read the comments here:
http://www.voip-info.org/wiki/index.php?page_id=573&tk=2ff846f8169b7694aed5&comments_page=1
Don't forget to have a beer ready :P
On Mon, Jun 14, 2010 at 7:18 PM, C F wrote:
> Your configs looks good, the only thing left is to figure out:
> 1. You sure you
Quickly:
Do some reading on PICKUPMARK: You need to set this on the channel that
you want to pick up, not the channel that is doing the pickup.
Philipp
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com -
Your configs looks good, the only thing left is to figure out:
1. You sure you have the right cable plugged in to the right port? The
reason I'm asking this is because you started out with the dchannel
being 48.
2. PRI Debug, what are the messages?
3. Contact your provider and try troubleshooting i
On Mon, 14 Jun 2010, bruce bruce wrote:
Thanks for the input. I actually did use verbose() and that is when I
noticed that my path to phpagi was not right since nothing was coming
through. For return value prior to fixing phpagi path, I was getting:
NoOp("SIP/64.111.222.111-0ca7", "")
wh
You can use realtime architecture. I have a similar setup, voicemails works
just fine.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-14 6:08 PM, "Jonathan González" wrote:
Hi there,
I have been taking a look on how to configure voicemail systems with
asterisk and I would like to know if
Thanks for the input. I actually did use verbose() and that is when I
noticed that my path to phpagi was not right since nothing was coming
through.
For return value prior to fixing phpagi path, I was getting:
*NoOp("SIP/64.111.222.111-0ca7", "")*
which actually is just right because if you
Go to http://phpagi.sourceforge.net/ and check the documentation.
Basically you can run any command you can run on the CLI or
extensions.conf. To get the status of a SIP peer I would recommend
using AMI. PHPAGI also has a library to connect to the AMI and send
commands.
On Mon, 2010-06-1
> On Mon, 2010-06-14 at 14:57 -0400, bruce bruce wrote:
>> Carlos, Thanks a lot for getting me started. That helps a great deal.
>> exten => _x.,1,NoOp(${EXTEN})
Since you're just getting started, there is an application specifically
written to send output to the CLI -- verbose(). It's more "ob
Hi there,
I have been taking a look on how to configure voicemail systems with
asterisk and I would like to know if there's any way to define mailbox in
a dynamic way.
I have 100 users and I would like to know if there's any way to avoid the
definition of the 100 mailboxes in voicemail.conf and u
Hi Guys,
Looking for a powerful box that is compact, can take two hard drives for
Raid-1 (no SSD, too expensive), have at least two Gig ports or two
10/100mbps ports. Fit two PCIe or one PCIe card plus it's daughter card
which needs as much room as a PCIe and doesn't need the actual slot. That is
Thanks for pointing to the debug. I found that the path to phpagi was not
right and since I fixed that everything seems to work fine.
Now, I want to know if I can use a phpagi command to check the status of SIP
Peer if it is online and registered or not. I know I can use *grep with
asterisk rx "si
If you do an "agi set debug on" from the CLI and run your AGI you
should see something like this:
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_request: auth.agi
[Jun 10 12:53:00] VERBOSE[8515] res_agi.c: AGI Tx >>
agi_channel: SIP/206-0d9b
[Jun 10 12:53:00] VERBOSE[8515] re
Hi again,
So, I have this but NoOp shows a random SIP info value rather then the one
passed to it. Just to test, I am sending $didin as argument to test.php and
then expect it back as $didgot back into dialplan. But it seems that either
send or receive has problem because no matter what I put as t
Also cheaper to replace flash card than hard drive.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Monday, June 14, 2010 4:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-us
Why no flash?
> * Small pre-built PC (not buying board, case, all parts separately)
> * Low power consumption
> * No fan or very small fan
> * Hard drive (not flash memory)
An ssd uses less power, so generates less warmth, hence less need for
fan in the drive area. Also less noise
Thank You Tarek!
That was the case, and i saw now i had a typo in the extension further down,
but, you solved it.
Now I faced a couple of other problems, alle the announcements and MOH didn’t
play, the settings are default.
Maybe i'll figure it out.
Thank you
Regards
Aksel
Fra: asterisk-u
The hangup is after Dial.
anay suggestion
De : Vardan Harutyunyan
À : asterisk-users@lists.digium.com
Envoyé le : Lun 14 juin 2010, 20h 31min 22s
Objet : Re: [asterisk-users] Re : Asterisk Call routing problem
Hangup is comming after Dial or AGI?
--
Vardan
If I use CONTROL STREAM FILE the messages disapearIt´s this ok?
On Mon, Jun 14, 2010 at 12:22 PM, Warren Selby wrote:
>
>
> On Mon, Jun 14, 2010 at 8:27 AM, equis software
> wrote:
>
>> In Asterisk 1.4.22 it doesn't happend, in version 1.4.23.1 and aboveappear
>> this messages
>>
>>
>
On Mon, 14 Jun 2010, Carlos Chavez wrote:
> You can even send parameters to the AGI via the command line like:
>
> exten => _x.,1,AGI(sample.agi,param1,param2) //Use comma for 1.6 or |
> for 1.4 or below
Comma works fine in 1.2.
--
Thanks in advance,
-
On Sun, Jun 13, 2010 at 2:59 PM, Vieri wrote:
> I'm wondering if anyone knows a good, stable C AGI library (* v. 1.4 and 1.6
> compatible).
> I've taken a look at CAGI and QUIVR but their latest code releases date back
> to 2006.
> I've also seen a more recent project (wildpbx) dated 2009:
> htt
Carlos, Thanks a lot for getting me started. That helps a great deal.
Currently, the *$agi->request['agi_extension']; returns the SIP channel
info with IP and I want that to be the incoming DID number. *
*
*
*My dialplan output is this for line one:*
**exten => _x.,1,NoOp(${EXTEN})
*415444555*
Using asterisk 1.4.30
sip.conf is realtime sip_buddies in mysql database. What settings here
affect the Pickup() ?? If you think about pickup/call-groups, I have
none defined.
extensions.conf :
exten => _**XX,1,NoOp()
exten => _**XX,n,Macro(GetKlantIDfromCALLnum,${CALLERID(num)})
exten => _*
Hangup is comming after Dial or AGI?
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Adil Zaaraoui wrote:
> Dear Vardan,
> I
Randy R wrote:
> Hi,
>
> I'm looking to build an Asterisk box that can run at a remote
> location. Here are most of the specs of what I'm looking for:
>
> Physical hardware
>
> * Small pre-built PC (not buying board, case, all parts separately)
> * Low power consumption
> * No fan or ve
On Mon, 2010-06-14 at 13:41 -0400, bruce bruce wrote:
> Hi Carlso,
>
>
> Thanks for the input. I have done this in php and am not familiar with
> phpagi.
> So, there is absolutely no way to temporarily solve this problem by
> getting the value back from php file?
>
>
> Wondering if it would req
sip.conf and extensions.conf would be helpful as well as knowing what
version you are running. Based on what you went, I would say you have a
config error, but I can't tell where without seeing the config.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digiu
Hi Carlso,
Thanks for the input. I have done this in php and am not familiar with
phpagi.
So, there is absolutely no way to temporarily solve this problem by getting
the value back from php file?
Wondering if it would require a lot of work to change the php file to
phpagi?
Thanks,
Bruce
On Mon,
Dear Vardan,
I had before the same problem, i reinstalled asterisk and it worked; now i get
the same problem;
i am using asterisk 1.4.22, it forwards if i call operator A (the one i was
testing); but all other numbers are not forwarded, it just hangup, here is the
output:
Accepting AUTHENTICA
Thanks everyone for your suggestions.
Is it feasable to run Skype for Asterisk on the Atom processors? It's
a feature I'd really like to have. As for conferencing, we rarely use
it but never would need more than 3 seats.
--
Thanks for the reply.
Actually my problem is not related to sip.conf and extensions.conf. I have
used only standard files from martin pdf which are given as example.
I am able to call some system connected over LAN, when each has a softphone
and are registered on a asterisk server. But now what i w
On Mon, 14 Jun 2010, Chris Bagnall wrote:
> Actually, the Atom seems to be surprisingly powerful. We have a couple of
> Atom boxes with transcoding and conferences enabled without issue. I
> wouldn't pretend it'll cope with hundreds of conference participants, but
> with ~10 or so it seems to be f
I knew somebody will send me these links, so I mentioned in my question that
wiki was of no help. These were the first pages I went through, and they
don't tell how to *disable* DST. To have DST picked up automatically by
these phones based on timezone is so nice of Snom but that is exactly what I
Hi!
> Snom wiki was not helpful. My client wants to keep his phones pointed
> to UTC time, no DST, no change in timezone, i.e. to stay at 0 hours
> difference.
How about this?
XML Syntax: Settings/dst/xml
http://wiki.snom.com/Settings/dst
Settings/utc offset:
Signed UTC offset in seconds. This
I would think AGI would be better. ?
I don't think system() returns anything, except maybe a success/fail ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Monday, June 14, 2010 12:00 PM
To: Asterisk User
> As I mentioned, I'm not inclined to mess with the secrets, too much
> hassle for users.
I'm afraid that I have to consider that attitude to be a bit like
saying "It's too much hassle for us to insist that our employees
lock their desk drawers and the front door... or wash their
hands after goi
On Mon, 2010-06-14 at 12:00 -0400, bruce bruce wrote:
> Hi Everyone,
>
>
> I have a php file that if an argument is passed to it, it will echo a
> number back. I am looking to use system() in dial-plan to send
> ${EXTEN} to it and then to get that processed value back from the php
> file and put
Hi Everyone,
I have a php file that if an argument is passed to it, it will echo a number
back. I am looking to use system() in dial-plan to send ${EXTEN} to it and
then to get that processed value back from the php file and put it in $var
back into asterisk dial-plan. While trying this method doe
Greetings,
Sounds like a simple thing to do, but I was not able to do it on these
particular phones. Snom wiki was not helpful. My client wants to keep his
phones pointed to UTC time, no DST, no change in timezone, i.e. to stay at 0
hours difference.
The phones are provisioned from a tftp server.
Hello list,
I try to pick up a ringing extension but nothing works.
To be clear, I'm trying to pick up extension 10.
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:4]
Pickup("SIP/testcorp3-0041", "1...@123456") in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159
On Mon, Jun 14, 2010 at 8:27 AM, equis software wrote:
> In Asterisk 1.4.22 it doesn't happend, in version 1.4.23.1 and aboveappear
> this messages
>
>
This message was added around 1.4.23 to let you know that you're violating
the AGI protocol. Read up on the AGI protocol then check through
Hi
I've just had a request from a customer who wants to use Busy Lamp Feed.
I've had a look around and it would appear that you have top use the
'hint' priority. We are using asterisk 1.4.17 with realtime and the
priority column in the extensions table is a tinyint so obviously I
can't put hi
On Sun, Jun 13, 2010 at 3:06 PM, sean darcy wrote:
> But I'm struck with your notion of having sip user ids different from
> extensions. That would not require any user effort, or messing with each
> phone. But...
>
It'd be just as much effort as changing the passwords for each phone.
You'll hav
Hi,
I am trying to find documents on multiple parking lots in Asterisk 1.6,
which was announced as a new feature. Although 1.6 has been out a while, I
see no info on how to set this up.
The wiki is stuck in the past (apr 2008). Is there anything available on
how to create multiple parking
Actually, the Atom seems to be surprisingly powerful. We have a couple of
Atom boxes with transcoding and conferences enabled without issue. I
wouldn't pretend it'll cope with hundreds of conference participants, but
with ~10 or so it seems to be fine.
Likewise with transcoding - we've only rea
In Asterisk 1.4.22 it doesn't happend, in version 1.4.23.1 and above appear
this messages
On Mon, Jun 14, 2010 at 9:08 AM, equis software wrote:
> This are the console messages with AGI debugging
>
> AGI Rx << STREAM FILE msgBienvenida112 1234567890*#
> -- Playing 'msgBienvenida112'
does that phon has a static IP? does it register with the server? posting your
SIP.con and extensions.conf related to this issue could help us to understand
what you are doing.
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
From: niksingha...@gmail
when you add an agent to a queue the agent should log in try adding
member=SIP/301member=SIP/302instead of agent directives.this will ring both
phones.. from your output it doesn't seem to be ringing the agents at all.
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 3
How limited are you in the no-no category when using a small machine
like this. Do you set you system to prevent transcoding? Does it
prevent conferencing? Just curious. I kind of like those features but
love the idea of the small machines.
On 06/14/2010 07:37 AM, Chris Bagnall wrote:
I'm
Hi everybody,
This is the console output of the asterisk server.
debian-te410*CLI> sip set debug peer 2002
SIP Debugging Enabled for IP: 172.26.48.113:5061
I have a sofphone with user 2002 registered on the server on the ip 113.
I am trying to place a call to the sofphone on this ip. I have wri
This are the console messages with AGI debugging
AGI Rx << STREAM FILE msgBienvenida112 1234567890*#
-- Playing 'msgBienvenida112' (escape_digits=1234567890*#)
(sample_offset 0)
[Jun 14 09:06:14] WARNING[21576]: file.c:1300 waitstream_core: write()
failed: Broken pipe
[Jun 14 09:06:14] WAR
On Mon, Jun 14, 2010 at 1:37 PM, Chris Bagnall
wrote:
> We've used the Asus eeeBox (desktop version of their little netbooks) quite
> successfully in past projects: Atom 1.6, 1GB RAM, 160GB HDD.
Wow, we used to benefit from the space program that handed down
technologies madre cheaper, now it's t
> I'm looking to build an Asterisk box that can run at a remote
> location.
We've used the Asus eeeBox (desktop version of their little netbooks) quite
successfully in past projects: Atom 1.6, 1GB RAM, 160GB HDD.
Generally we run Gentoo Linux with Asterisk 1.4., but no reason why
you couldn't r
Hello there
I have been struggling with queues, because i think this is the right module
for our business.
My main goal, is when we receive external calls, the receptionist should be
able to transfer the call to us
Technicians, and I am trying to add 2 extensions to a queue name [teknisk]
Exten
I don't think it's a disk space issue :
bash-3.2# df -h
FilesystemSize Used Avail Use% Mounted on
/dev/sda1 25G 5.0G 19G 21% /
tmpfs 256M 0 256M 0% /dev/shm
bash-3.2# df -h /var/log/
FilesystemSize Used Avail Use% Mounted on
/dev/
On 14/06/10 11:04, Jonas Kellens wrote:
Hello list,
I noticed today that the last logfiles dates 3 days ago !
The logfiles are rotated every night. The logfiles of 2 days ago, 1
day ago and today are empty !
vps*CLI> module show like logger
Module
Description
Hello list,
I noticed today that the last logfiles dates 3 days ago !
The logfiles are rotated every night. The logfiles of 2 days ago, 1 day
ago and today are empty !
vps*CLI> module show like logger
Module Description
Use Count
0 module
along with all the previous suggestions.. i found out that fail2ban is a good
safe tool to be used along with hard passwords and not using numeric
usernames.. for me using A2Billing along with Asterisk was a pain because it
needs to create usernames numeric.. so i had to create strong SIP users
Voip Asterisk wrote:
> Ya I'm passed that part now. I have dahdi properly loading the card,
> and both links are green. Asterisk recognizes the channels, but still
> shows the span as down.
Some telcos turn down a span when too many errors have occurred. You
may still want to contact your p
Andrew Joakimsen wrote:
> I'm running into a strange issue with Asterisk + Iaxmodem + hylafax on
> the same machine. After rebooting the iaxmodems don't register to
> asterisk. Stoping and starting the relevant services gets it working,
>
>
I'm using init to spawn a script that kicks off iaxmo
> * Skype for Asterisk needs to run on this <- so this means x86, right?
or x86_64 is fine
--
Thanks, Phil
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introduc
Hi all,
i got a lot of this messages if only one caller is in a meetme
conference and it playing a MusicOnHold Sound. If a second Caller
entry the Conference the messages ended.
DEBUG[11794] channel.c: Internal timing is disabled
(option_internal_timing=0 chan->timingfd=61
What does this message
AsteriskNow is better
On Mon, Jun 14, 2010 at 2:26 PM, Randy R wrote:
> Hi,
>
> I'm looking to build an Asterisk box that can run at a remote
> location. Here are most of the specs of what I'm looking for:
>
> Physical hardware
>
>* Small pre-built PC (not buying board, case, all parts separ
Hi,
I'm looking to build an Asterisk box that can run at a remote
location. Here are most of the specs of what I'm looking for:
Physical hardware
* Small pre-built PC (not buying board, case, all parts separately)
* Low power consumption
* No fan or very small fan
* Hard drive (n
I'm running into a strange issue with Asterisk + Iaxmodem + hylafax on
the same machine. After rebooting the iaxmodems don't register to
asterisk. Stoping and starting the relevant services gets it working,
but what is the point of using init scripts if it does not work right?
I already tried to ad
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