Hi Everyone,
I am writing an open source application that brings desktops widgets
to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I
am trying to get my head around the Asterisk Manager Interface.
I had been using the Event: NewCallerid to detect a new call which my
Asterisk
Hello good ppl,
Any way of stopping asterisk writing into syslogs or any other file, if
I set verbose 6 on the console?
All I want is the verbose output only on the console, nowhere else.
My logger.conf says :
console= notice,error
;messages = notice,warning,error
Thanks in advance.
- Benjamin
Btw, even the syslog line in logger.conf is commented :
; syslog.local0 = notice,warning,error
Benjamin Jacob wrote:
Hello good ppl,
Any way of stopping asterisk writing into syslogs or any other file, if
I set verbose 6 on the console?
All I want is the verbose output only on the console,
Quoth Moises Silva...
May be I am missing something, but, manager command DBPut should do
the trick of putting the DB value. And, since you are already using
the manager interface, you are using PHP or PERL to connect to the
Database, why not wait for the DBPut command response and from the
On 9/4/07, Devraj Mukherjee [EMAIL PROTECTED] wrote:
Hi Everyone,
I am writing an open source application that brings desktops widgets
to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I
am trying to get my head around the Asterisk Manager Interface.
I had been using the
What logs are coming out to /var/log/messages?
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
Jacob
Sent: 04 September 2007 07:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] stop
On Sun, Sep 02, 2007 at 03:47:45PM +0100, Chris Bagnall wrote:
There's both a 7960 and a 7960G (and a 7961 to confuse matters further).
The 7960 is the earlier version. The easiest way to identify it from a
picture is to look at the messages/services/etc. buttons. On the 7960 the
words
On Tue, Sep 04, 2007 at 10:43:15AM +0100, Adrian Marsh wrote:
What logs are coming out to /var/log/messages?
Ask asterisk
logger show channels
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
Exactly the same lines as on the console.
Adrian Marsh wrote:
What logs are coming out to /var/log/messages?
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
Jacob
Sent: 04 September 2007 07:58
To: Asterisk Users Mailing List -
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
Because it seems my mail from 30th august didn't make it to the list i
send it again. If the mail _did_ get to the list and i didn't see it
please excuse the duplicate post
Below is the mail from the 30th:
I have a setup like this:
An
Hi Atis,
Is your code open source, or are you willing to share your PHP code
snippets with me? And thanks for the information on Asterisk's
stability. Do you think there is an issue in the implementation or
just network/traffic issues?
Thanks for your time.
On 9/4/07, Atis [EMAIL PROTECTED]
Here it is :
SIP01*CLI logger show channels
Channel Type StatusConfiguration
--- ---
Console Enabled- Notice Error
Tzafrir Cohen wrote:
On Tue, Sep 04,
On Sun, Sep 02, 2007 at 04:03:51PM +0200, Jonathan GF wrote:
Hi folks,
i'm trying to configure my extensions.conf as small as posible and for
that reason i'm using macros. The problem is that maybe I have a
misunderstood the concept for the directive mailbox in sip.conf.
What mailbox= seems
On Sun, Sep 02, 2007 at 05:15:41PM -0500, Anthony Messina wrote:
is asterisk capable of generating the off-hook warning tone for the us?
1400+2060+2450+2600/100,0/100
i have placed it into indications.conf, but all i get is one high-pitched
screech instead of alternating tones.
I am
On Tuesday 04 September 2007 06:54:55 am Robert Lister wrote:
On Sun, Sep 02, 2007 at 05:15:41PM -0500, Anthony Messina wrote:
is asterisk capable of generating the off-hook warning tone for the us?
1400+2060+2450+2600/100,0/100
i have placed it into indications.conf, but all i get is
Hey Robert,
you can't imagine how much i appreciate your post, which is most a
tutorial than a post :)
Really, many thanks for your thoughts. Take for sure i will try to
implement the options you showed me here in asap.
Thank you again!
Best regards,
Jonathan GF
On 9/4/07, Robert Lister
All,
I've been experimenting with shortcodes for SIPgate etc. Passing calls
to SIPbroker seems a good way to go, but the message I've had back from
SIPgate is we don't support SIPBroker...
So whats the easiest way to support SIP SIP network calling?
At the moment, I've setup some local
When you access the A*k console, is this via a tty connection
(ssh/telnet), or actually on the physical console of the server?
I don't think it's A*k that's directly logging to the console - the
config doesn't show that... I'm guessing, that you're accessing A*k via
the local terminal, and that
Adrian Marsh wrote:
All,
I've been experimenting with shortcodes for SIPgate etc. Passing calls
to SIPbroker seems a good way to go, but the message I've had back from
SIPgate is we don't support SIPBroker...
So whats the easiest way to support SIP SIP network calling?
At the moment,
On Tue, Sep 04, 2007 at 12:17:02PM +0530, Benjamin Jacob wrote:
Hello good ppl,
Any way of stopping asterisk writing into syslogs or any other file, if
I set verbose 6 on the console?
All I want is the verbose output only on the console, nowhere else.
My logger.conf says :
console=
On 9/3/07, Arinze Izukanne [EMAIL PROTECTED] wrote:
Can you show me a sample fo config? The link schematic should look like
this:
E1 == TDMoE==E1.
Refer to the section Sample configs for setting up TDMoE between 2 servers
without TDM hardware, using ztdummy on this page:
Yeah,
I can see that now after testing it all - but this is what raised my
question.. What IS the best mechanism for all the VoIP servers/networks
to interact ? Setting up individual agreements for each network is so
1980's, and in this modern world there must be a better solution..
A.
Hello All,
Anyone knows what does this error message means and where to check for
the cause and why it happened?
Asterisk on hyperion exited on signal 11. Might want to take a peek.
But when I check Asterisk, its running fine...
Cheers,
Nitesh
To answer my own question I found a way to acheive what I wanted so
here's my solution for the record (might help someone else if they
search the archives).
In the Dialplan setup the following entries:
[snom_setdndon]
exten = _.,1,NoOp(Dummy Routine Called for ${EXTEN})
Do you know where to find clear developers' guides (with some examples)
for developing apps that run *on* Cisco 79xx phones (especially the
7970)? Examples that can run against Asterisk (not CallManager) with SIP
firmware (not SCCP), and/or LDAP directories (or other open servers)
would be
Signal 11 is a segmentation fault, if you are not running unsupported
patches on Asterisk you should compile without Asterisk optimizations
and open a bug attaching the debugging backtrace.
Read This:
http://www.voip-info.org/wiki-Asterisk+debugging
Regards,
On 9/4/07, Nitesh Divecha [EMAIL
I've got an Asterisk switch that is going to run an IVR menu with a database
interface that will be doing lookups based on the user entered data and then
reading back strings with the appropriate data integrated into the text. I
have found quite a bit of data on using MySQL as a database with
I haven't come across any wireless devices that support IAX2, but we have
successfully used the Linksys WIP300, Linksys WIP330, Nokia N80, Nokia E61i,
and Nokia N95 with asterisk.
If you just need wireless and not mobility, the Linksys WBP54G also works well
to interconnect Ethernet based VoIP
You can use Asterisk's AGI and PHP/Perl or whatever else. You'll need
to install connecting software, such as FreeTDS, to connect to SQL.
Then you can either pass arguments to your script or use environment
variables to set Asterisk variables.
Here's a good place to start:
Seriously, from our experience, SIPBroker IS the best way to interact
with all the open networks. For any closed networks, you might create
special rules for interaction, but that would rely on setting up a deal
with the respective destination network to actually ALLOW your calls.
There are
On 9/4/07, Larry Costigan [EMAIL PROTECTED] wrote:
I've got an Asterisk switch that is going to run an IVR menu with a database
interface that will be doing lookups based on the user entered data and then
reading back strings with the appropriate data integrated into the text. I
have found
On Tue, Sep 04, 2007 at 07:41:53AM -0500, Anthony Messina wrote:
well i'm looking for the feature that the telco provides where, if you've
left
the phone off-hook for 60 seconds or so without input, it gives you the
loud put the damn phone back on the hook noise.
it works if i set
While attempting to install zaptel I received the following output in
response to make install:
...
Install -d /etc/udev/rules.d
Build_tools /genudevrules /etc/udev/rules.d/zaptel.rules
Build_tools /genudevrules :line 1: udevinfo : command not found
Make: *** [devices] error 1
And the install
The correct term for this tone is howler. I'm surprised it is not in
indications.conf
Robert Lister wrote:
On Tue, Sep 04, 2007 at 07:41:53AM -0500, Anthony Messina wrote:
well i'm looking for the feature that the telco provides where, if you've
left
the phone off-hook for 60 seconds or
On Tuesday 04 September 2007 03:24:59 pm Robert Lister wrote:
On Tue, Sep 04, 2007 at 07:41:53AM -0500, Anthony Messina wrote:
well i'm looking for the feature that the telco provides where, if you've
left the phone off-hook for 60 seconds or so without input, it gives you
the loud put the
On Tue, Sep 04, 2007 at 03:44:46PM -0500, Eric ManxPower Wieling wrote:
The correct term for this tone is howler. I'm surprised it is not in
indications.conf
There are two sorts. The 'howler' is usually a single frequency tone
-- 1500ish Hz, I think -- that gets progressively louder over
I have a customer that has two buildings that are connected with a
fiber link. We have a single Asterisk server to cover both buildings.
Now the customer went and bought an overhead paging system for the
remote building and they want to integrate it with Asterisk. Is there a
device that
Anthony, Robert here is mentioning a SIP phone, but I didn't see you
specify what kind of phone you have. Is this accurate, or is it a
Zaptel FXS port?
Moj
Robert Lister wrote:
On Tue, Sep 04, 2007 at 07:41:53AM -0500, Anthony Messina wrote:
well i'm looking for the feature that the
For me as well, if I append a #, asterisk tries to match it to an
extension and I get congestion indication when it fails to do that. This
would be nice though!
(TDM card too)
Moj
Giuffredi wrote:
If I press # I get “incorrect number” as asterisk passes to the telco
the numbers and the
Carlos Chavez wrote:
I have a customer that has two buildings that are connected with a
fiber link. We have a single Asterisk server to cover both buildings.
Now the customer went and bought an overhead paging system for the
remote building and they want to integrate it with Asterisk.
Something like this should work
http://www.voipsupply.com/product_info.php?products_id=3517osCsid=85766c1b3901219249d6fdea9bc7b7c0
On 9/4/07, Carlos Chavez [EMAIL PROTECTED] wrote:
I have a customer that has two buildings that are connected with a
fiber link. We have a single
On Tue, Sep 04, 2007 at 01:39:55PM -0700, Markham, Craig (FRTC Contractor)
wrote:
While attempting to install zaptel I received the following output in
response to make install:
...
Install -d /etc/udev/rules.d
Build_tools /genudevrules /etc/udev/rules.d/zaptel.rules
Build_tools
You can use an inexpensive PC with a sound card. Install Asterisk on
it and set an extension that calls /dev/dsp. This will send audio
out the speaker port on the sound card.
Setup a trunk between this unit and your primary Asterisk server and
you should be in business.
Bryan M. Johns
Most of the overhead paging systems I have worked with had an CO
input, which worked with the FXO port on an ATA. A couple brands had
multiple source options, it is worth checking, I had problems with
poor audio quality using the sound card with asterisk.
On 9/4/07, Carlos Chavez [EMAIL
Hi,
I want to detect a tone while Dial() through pri.
When a secial tone(eg, #), I want to send the call to
another extenison.
Regards.
Take the Internet to Go: Yahoo!Go puts the Internet in your
Bruce Reeves wrote:
multiple source options, it is worth checking, I had problems with
poor audio quality using the sound card with asterisk.
I did as well using the built-in sound on the motherboard that I was
using, switched to a Sound Blaster Live Value card and that problem went
away.
Hi,
I'm new to asterisk, in order to enable X-lite to make a call, what should i do
before making a call?
Current stage,
1. i have create a few accounts in sip.conf.
2. Registration are successful.
Pls advice me how to continue then...
Thanks
asterisk-usershello,everyone!
I was setting up ChanSpy in an Asterisk dialplan today and it just wasn't
working. Here is the snippet:
extensions.conf:
[test]
exten=3001, 1, Set(__TRANSFER_CONTEXT=tranfer)
exten=3001, 2, Dial(SIP/3001,10,tr)
exten=2002, 1,
I have a TDM800P with 5 POTS lines hooked into my university's PBX.
CID information is being picked up properly from phones inside the
PBX, but not from phones outside. I've tried mucking with rxgain with
no result, likewise cidstart and cidsignalling.
The CID number shows up properly if a
On Tuesday 04 September 2007 06:31:08 pm Mojo with Horan Company, LLC wrote:
Anthony, Robert here is mentioning a SIP phone, but I didn't see you
specify what kind of phone you have. Is this accurate, or is it a
Zaptel FXS port?
a zap fxs port is correct.
--
Anthony - http://messinet.com
Helps us help you further, what do you intend to do?
- Dial using a normal telephone line
- Dial using a VoIP provider?
What hardware do you have, etc
On 9/5/07, neoh kumyee [EMAIL PROTECTED] wrote:
Hi,
I'm new to asterisk, in order to enable X-lite to make a call, what should i
do before
To make call to X-lite or any sip phone ,
1. create extension in sip.conf for soft phone.
2. register sip phone with that exentension which is in sip.conf
-give your asterisk server ip in softphone and also username and password.
3. wait for some time
4. if all are good then your phone has been
Hi all,
My name is Ricardo and unfortunately I'm just crawling in this
telecomm/asterisk world. So, after reading all day long i still don't
understand a few things. :D
I'm trying to develop a call recorder for a costumer. He has a small
call center ( 10 agents ) and want to record all
How are you playing the voice? Do you use something like app_swift
or app_cepstral? Just fixed app_swift for my own installation by
changing the framesize constant definition from 160*4 to 20,
after googling for a similar issue. Works like a charm now. It only
broke recently, i.e. not with the
Hi,
Thanks for your reply..
I am intend to dial using a VOIP provider.(developed by us)
Software: x-Lite (SIP softphone)
Registration of account number is fine, but for the case when i dial a number,
it prompt out a message that the number not found.
From my understanding, asterisk can be
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