Hello, when is use SIP realtime and i (re)start asterisk, then all SIP user are
Unavailable, and never go to Not in use because the phones are
registered on opensips.
And res_config_mysql does load the user only, when the SIP does a call
(or get called)
an then chan_sip give app_queue the
Hi,
I'm currently using a cheap SIP provider for outbound calls.
I do have 6 channels to them.
In their terms of service there is the following limit:
The total duration of calls during one single day should not exceed 24
hours or we do have the right to terminate the contract...blah blah
If your server is good enough to handle those queries, I would have a cron
job use the CDR to calculate the time spent this current day every minute or
so, put this value in a text file (or anywhere really) and read it every
call you make to check, and use this as the absolute timeout value for
Make sure that you only have the one grammar active when doing your test.
You want the voice engine to basically only have 11 possibilities to chew on
(0-9 plus oh).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
I will be out of the office starting
Wed June 9th and returning Wed June 16th.
Please contact Mary at m...@accessgate.net cell 407-267-1463
or Jonathan at jsny...@accessgate.net cell 407-267-0056
or call our main number 888-227-9337.
--
My .02 - I would set up a context for dialing with this provider that counts
minutes and stops the dial with a message once you get to 1321 minutes (22
hours).
Exten = _8X.,1,noop(call using Cheap sip)
Exten = _8X.,2,macro(call_out,${EXTEN:1})
Exten = _8X.,3,hangup
[macro-call_out]
Exten =
Asterisk 1.4.31 IAX2/RSA BUG
If A tries to register on B and the RSA key from A does not match the
key on inkeys on B, B do not send a REGREJ, instead it sends a
REGAUTH with a new CHALLENGE, then A send a new REGREQ for this
CHALLENGE with the same wrong RSA key and it loops forever on this
I'm fairly new to FreePBX/Asterisk/Trixbox, but have Googled myself
into submission here, so any assistance is appreciated.
We had a user with a weak SIP secret recently that allowed it to be
used by an outside user. The extension was 3799. I could see the
intruder's calls (including the
On 08/06/2010 15:21, Danny Nicholas wrote:
My .02 - I would set up a context for dialing with this provider that counts
minutes and stops the dial with a message once you get to 1321 minutes (22
hours).
Exten = _8X.,1,noop(call using Cheap sip)
Exten = _8X.,2,macro(call_out,${EXTEN:1})
On 08/06/10 14:50, J wrote:
I'm fairly new to FreePBX/Asterisk/Trixbox, but have Googled myself
into submission here, so any assistance is appreciated.
We had a user with a weak SIP secret recently that allowed it to be
used by an outside user. The extension was 3799. I could see the
intruder's
I hope I'm correct, I don't have time to verify every bit of this,
but
The message
[Jun 7 17:04:16] NOTICE[7422] chan_sip.c: Failed to authenticate user
asterisk sip:3...@206.205.124.247 sip%3a3...@206.205.124.247
;tag=as23bacb61
indicates the user asterisk. Do you have a sip account for
Since you mentioned FreePBX, unfortunately, it's not only the GUI that
drives the system and it can be that at some point someone planted
the extension in one of your .conf or other file if they had access to SSH
or some other way.
Going back to occurrence in sip.conf as mentioned, of course
Hello,
I noticed that changes to realtime sip peers are not applied until a
'reload'. A 'sip reload' does not make any changes to realtime sip peers.
When changing for instance the mailbox-parameter in the realtime
sip_buddies table, the change is not applied with a 'sip reload'.
For every
I just made a change to one of my *.gram files for my LumenVox IVR. I was just
wondering if anyone knows the command in Asterisk to reload the .gram files.
Thanks for your help
--
_
-- Bandwidth and Colocation Provided by
Speechunloadgrammar followed by speechloadgrammar should do the trick.
Let's say you changed /etc/asterisk/grammars/foo.gram that you are using as
foobar.
Speechunloadgrammar(foobar) will knock it out
Speechloadgrammar(foobar,'/etc/asterisk/grammars/foo.gram) will reactivate
it.
_
Hi J,
When I used FreePBX, I faced these situations occasionally. It is normal to
see these entries in your CDR when hackers are trying to misuse your system.
There doesn't need to be a real extension for it to appear it in the CDR.
Based on what SIP URI the hacker sends, the CDR will display
On 8 Jun 2010, at 16:40, Jonas Kellens wrote:
I noticed that changes to realtime sip peers are not applied until a
'reload'. A 'sip reload' does not make any changes to realtime sip peers.
sip prune ?
S
--
_
-- Bandwidth and
On my asterisk 1.4.22-4 sip reload works just fine, and occasionally I have
to use it when sip secrets are updated. What version are you using?
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-06-08 11:46 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello,
I
On Tuesday 08 June 2010 10:40:44 Jonas Kellens wrote:
Why does a 'sip reload' only checks the following files :
[Jun 8 17:31:16] == Parsing '/etc/asterisk/sip.conf': [Jun 8
17:31:16] Found
[Jun 8 17:31:16] == Parsing '/etc/asterisk/users.conf': [Jun 8
17:31:16] Found
[Jun 8
On Tue, 8 Jun 2010, Laurent CARON wrote:
I'm currently using a cheap SIP provider for outbound calls.
I do have 6 channels to them.
In their terms of service there is the following limit:
The total duration of calls during one single day should not exceed 24
hours or we do have the right
We hit this issue and are reviewing the patch to install now...
Any updates?
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux
Using 1.4.30
Jonas.
On 06/08/2010 05:53 PM, Zeeshan Zakaria wrote:
On my asterisk 1.4.22-4 sip reload works just fine, and occasionally I
have to use it when sip secrets are updated. What version are you using?
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
--
Thank you all very much for your replies. I've gone ahead and made a
few tweaks that might help, including disabling anonymous inbound SIP
calls. I also exhaustively grepped /etc/asterisk for 3799, and there
are absolutely no occurances of it. The files do not include files in
any other
Do you have rtcachefriends=yes in your sip.conf?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-08 1:30 PM, Jonas Kellens jonas.kell...@telenet.be wrote:
Using 1.4.30
Jonas.
On 06/08/2010 05:53 PM, Zeeshan Zakaria wrote:
On my asterisk 1.4.22-4 sip reload works just fi...
--
The Asterisk Development Team has announced the release of version 1.4.11.2 of
libpri. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri/
This release fixes situation where Q.SIG calling name in FACILITY message was
not reported to the upper
See your extensions.conf for context [from-sip-external] and it'll all make
sense.
Install and configure Fail2Ban, search for instructions on voip-info.org and
make your life easier.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-08 1:44 PM, J jmau...@2ergo.com wrote:
Thank you all very
Off course...
Jonas.
On 06/08/2010 07:44 PM, Zeeshan Zakaria wrote:
Do you have rtcachefriends=yes in your sip.conf?
Zeeshan A Zakaria
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
The Asterisk Development Team has announced the release of versions 1.6.0.6 and
1.6.1.4 of asterisk-addons. These releases are available for immediate download
at http://downloads.asterisk.org/pub/telephony/asterisk
The Asterisk-Addons releases for 1.6.0.6 and 1.6.1.4 are the last maintenance
Can't think of anything else at the moment.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-08 2:17 PM, Jonas Kellens jonas.kell...@telenet.be wrote:
Off course...
Jonas.
On 06/08/2010 07:44 PM, Zeeshan Zakaria wrote:
Do you have rtcachefriends=yes in your sip.con...
--
Make sure that you only have the one grammar active when doing your test.
You want the voice engine to basically only have 11 possibilities to chew on
(0-9 plus oh).
I always only load one grammar. In the test I did below, there were
exactly TWO possibilities:
I'm having a lot of problem
Maybe your engine is tone deaf. Try showing the ${SPEECH_SCORE(0)} when
you get the foobared result.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner
Sent: Tuesday, June 08, 2010 1:36 PM
To:
I actually commented all this in safe_asterisk and now asterisk loads all
fine at the beginning. Is this okay to do? Also at the beginning of the file
I commented #TTY=9 as well. Can someone shade some light as to what TTY is
and how it can have an adverse effect if it's not available?
#if test
Hello
I want to use my own caller id, instead of the caller id of PSTN line, for the
outbound calls through DAHDI channel. Is there any way ??
Regards
Taimur Hasan
-THQ- !!!ONE
_
I have an installation 100% dialplan and mysql-addons.
I find out that mysql-addons is working great, but I suspect there may be a
memory leak involved.
Anyone else facing a memory leak recently ?
Regards
--
_
-- Bandwidth
No unfortunatlely, there is no way.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-08 3:40 PM, taimur hasan taimurh...@hotmail.com wrote:
Hello
I want to use my own caller id, instead of the caller id of PSTN line, for
the outbound calls through DAHDI channel. Is there any way ??
On Tue, 8 Jun 2010, taimur hasan wrote:
I want to use my own caller id, instead of the caller id of PSTN line,
for the outbound calls through DAHDI channel. Is there any way ??
It depends on your technology (POTS, PRI, etc) and your provider.
Tell your provider you want to set the outgoing
http://leyvacrystaljd.blog23.com
_
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969--
G'day,
I am testing a new server with a Supermicro X7SBA motherboard
and TE405 cards. When I load Dahdi and run dahdi_cfg I get
the following errors:
kernel: Uhhuh, NMI received for unknown reason b1 on CPU 0
You probably have a hardware problem with your RAM chips
We can set our own CallerID. Telco gives us 100 different numbers comming in
our PRI and we may choose one of those 100 as a CallerID
We had to ask telco to permit us this change.
They allowed us to set on the initial SETUP message if we use our own
presentation.
This we we can also use
hi folks. i have the following puzzle:
when i call certain cell phone# using a regular phone POTS.
the called cell phone co. usually return a message such as
phone travel out of range or phone is busy etc. if the phone is
unreachable. now when i have the following setup:
sip phone - asterisk -
-Original Message-
From: Asterisk Development Team asteriskt...@digium.com
Sent: Tuesday, June 08, 2010 11:20 AM
To: asteriskt...@digium.com
Subject: [asterisk-users] Asterisk-Addons 1.6.0.6 and 1.6.1.4 Now Available
The Asterisk Development Team has announced the release of versions
We hit this issue and are reviewing the patch to install now...
Any updates?
Nope. I think any of the patches posted to either of the issues will
work, though the official one is obviously the best.
--
_
-- Bandwidth and
Maybe your engine is tone deaf. Try showing the ${SPEECH_SCORE(0)} when
you get the foobared result.
I repeated the experiment, this time noting the score, which I output.
This time, the result was always 2 and the score was pretty
high: 711, 743, 752.
--
Leif Madsen wrote:
sean darcy wrote:
Richard Kenner wrote:
Is this bug alive in 1.6.2.9-rc1? I'm getting segfaults from chan_dahdi.
If it does effect 1.6.2.8-rc1, I'll just wait for rc2 to see if this
is my problem, instead of filing.
I reported another instance of this today and it was
The latest libpri is supposed to handle this properly, but doesn't
seem to. Here's the debug info. CALLERID(name) is set to empty.
Protocol Discriminator: Q.931 (8) len=66
TEI=0 Call Ref: len= 2 (reference 256/0x100) (Sent from originator)
Message Type: SETUP (5)
[04 03 80 90 a2]
Bearer
45 matches
Mail list logo
