On Thu, Nov 19, 2009 at 07:01:13AM +0100, Olivier wrote:
Hi,
I'm using a revision 6822-enabled Dahdi-Tools (see
https://issues.asterisk.org/view.php?id=13897) with a Junghanns QuadBRI.
This patch has now been merged into the trunk of DAHDI.
1. Do I still need qozap driver ? If positive,
Indeed it does. You add contacts and set the softphone number to
extension@server
Leif Neland wrote:
Philipp Kempgen skrev:
Leif Neland schrieb:
Mostly to debug/test BLF, is there a softphone or another app. which can
subscribe to hints on Asterisk?
X-Lite?
Leif Neland schrieb:
Philipp Kempgen skrev:
Leif Neland schrieb:
Mostly to debug/test BLF, is there a softphone or another app. which can
subscribe to hints on Asterisk?
X-Lite?
It does not
subscribe to hints on Asterisk.
It does.
In the contact drawer: Add contact - Contact
Hi,
there are several possibilities do to it
REGISTER Username/Extensions Enumeration
INVITE Username/Extensions Enumeration
OPTION Username/Extensions Enumeration
for more information:
http://www.hackingvoip.com/presentations/sample_chapter3_hacking_voip.pdf
rich...
On Thu, Nov 19, 2009 at
Ok. I do NOT have ports 1-2 opened in. I guess I
should try that and see if it works.
I will open ports 5060 - 5070 and 1 - 100100 and do
some test tonight. I will keep you posted.
I ran this test and there was no difference.
I still can't get through.
---
Retransmitting
A) is easy. - exten = s,1,voicemail(group) - record once, send X
times.
B) Not quite so simple; you could give each person in the group an
email address and make the non-Euro's be regular email (j...@hotmail.com) and
Euro's be the SMS address (0123456...@cingular.com). You would want
The Asterisk Development Team is pleased to announce the release of Asterisk
1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
These releases resolve a large assortment of issues reported by the community.
--- On Wed, 11/18/09, Vieri rentor...@yahoo.com wrote:
From: Vieri rentor...@yahoo.com
Subject: [asterisk-users] asterisk 1.4.26.3 makes kernel panic
To: asterisk-users@lists.digium.com
Date: Wednesday, November 18, 2009, 5:34 AM
Hi,
I'm experiencing frequent kernel panics on a system
Hello,
On a very new system, I've got :
# cat /etc/dahdi/genconf_parameters | grep -v ^#
lc_country fr
context_lines remote
group_lines 1
bri_sig_style bri_ptmp
echo_canoslec
pri_termtype
SPAN/1 TE
SPAN/2
2009/11/19 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Nov 19, 2009 at 07:01:13AM +0100, Olivier wrote:
Hi,
I'm using a revision 6822-enabled Dahdi-Tools (see
https://issues.asterisk.org/view.php?id=13897) with a Junghanns QuadBRI.
This patch has now been merged into the trunk of
Hi All
I would Like to run a macro in a meetme conference when a user presses a
certain digit sequence, but I cannot seem to find how to do this , is it
possible?
if so how?
Thanks for you help
Robb
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You can use the X option to make meetme run the macro - I'd do it something
like this:
- exten = 1500,1,meetme(1500,X)
- exten = 1500,2,hangup
- exten = 6,1,Macro
- exten = 6,2,Goto(default|1500|1)
This will run the macro if the user presses 6 and return
On Thu, Nov 19, 2009 at 03:46:50PM +0100, Olivier wrote:
Hello,
On a very new system, I've got :
# cat /etc/dahdi/genconf_parameters | grep -v ^#
lc_country fr
context_lines remote
group_lines 1
bri_sig_style bri_ptmp
This is the default
snip
Customers in Europe all have mobile phones, while senders in North America
rarely have them ( they have answering machines, though ).
/snip
What planet/year are you/your clients living on/in? I don't know anyone who
doesn't have a mobile. Maybe it's just that they call it a cell phone
You have to remember, these are folks who get their product by boat .
unless they're an auto maker, that's pretty 19th/20th century.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Thursday, November 19,
On 19/11/09 15:37, Asterisk Development Team wrote:
The Asterisk Development Team is pleased to announce the release of Asterisk
1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
The site page still
David Gibbons wrote:
snip
Customers in Europe all have mobile phones, while senders in North
America rarely have them ( they have answering machines, though ).
/snip
What planet/year are you/your clients living on/in? I don’t know anyone
who doesn’t have a mobile. Maybe it’s just
The 'Make sounds' routine into Makefile doesn't seem to pre-fetch all of
the audio tarballs.
Is this an oversight or is there a strategic reason for it?
Specifically it doesn't seem to fetch the MOH tracks for selected codec's.
For example, during the most recent 1.6.1 update, the g.722
On Thu, 2009-11-19 at 10:50 -0600, Karl Fife wrote:
The 'Make sounds' routine into Makefile doesn't seem to pre-fetch all of
the audio tarballs.
Is this an oversight or is there a strategic reason for it?
As I understand it, it only pulls the tarballs you have selected in
make menuselect. Is
On Thu, 2009-11-19 at 10:50 -0600, Karl Fife wrote:
The 'Make sounds' routine into Makefile doesn't seem to pre-fetch all
of
the audio tarballs.
Is this an oversight or is there a strategic reason for it?
As I understand it, it only pulls the tarballs you have selected in
make menuselect.
Previously incorrectly sent to asterisk-dev list, sorry.
I tried today while connected to a Jtec QSIG E1 card, with
DAHDISendCallreroutingFacility with the following test dialplan:
Extension 4888 is on the Fujitsu
[incoming]
exten = 8688,1,Answer()
exten = 8688,n,Playback(connecting)
exten =
On 11/12/2009 09:31 AM, Dr. Michael J. Chudobiak wrote:
Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason
a0 on CPU 0.
Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely
on the PCI bus.
Nov 12 08:54:27 steerpike kernel: Dazed and confused, but
At 10:43 PM 11/18/2009, you wrote:
Ira skrev:
At 07:06 AM 11/18/2009, you wrote:
I know that I'm not looking for Dial(SIP/xSIP/y) - as documented, this
handles nothing like what I'm looking for.
It's not the answer you're looking for, but that feature is built
into a Aastra 480i-CT
AXvoice server hacked. Here are few working accounts
USE XLITE to make calls
Registrar/Proxy
magnum.axvoice.com:9060
Free Sample account
username=xMaxwellSmartx
secret=thanksapache
username=woodsy
type=friend
secret=haramikuttasala
username=wumingzi
type=friend
secret=kickyourass
Dave Cotton wrote:
On 19/11/09 15:37, Asterisk Development Team wrote:
The Asterisk Development Team is pleased to announce the release of Asterisk
1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
On 19 Nov 2009, at 19:32, baba jigger wrote:
AXvoice server hacked. Here are few working accounts
USE XLITE to make calls
What a fcking idiot.
S
___
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asterisk-users mailing
Steve Howes wrote:
On 19 Nov 2009, at 19:32, baba jigger wrote:
AXvoice server hacked. Here are few working accounts
USE XLITE to make calls
What a fcking idiot.
My thoughts exactly.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase
On Fri, Nov 20, 2009 at 12:32:31AM +0500, baba jigger wrote:
AXvoice server hacked. Here are few working accounts
USE XLITE to make calls
Registrar/Proxy
magnum.axvoice.com:9060
Free Sample account
username=xMaxwellSmartx
secret=thanksapache
username=woodsy
type=friend
Usually, hackers give out the details they discover so that thousands of
people use the stolen details, and therefore its impossible to detect
which user is the actual hacker.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Fri, 20 Nov 2009, baba jigger wrote:
AXvoice server hacked. Here are few working accounts
USE XLITE to make calls
How was the server hacked? The usernames and passwords do not appear to be
particularly guessable.
This would be much more interesting and relevant to this list.
--
logic says just send his email off to gmail, im sure their security will do
something in regards to his using of a gmail account to forward this info
On Thu, Nov 19, 2009 at 3:06 PM, Steve Howes steve-li...@geekinter.netwrote:
On 19 Nov 2009, at 19:32, baba jigger wrote:
AXvoice server
No way ??
On Thu, Nov 19, 2009 at 8:59 AM, David @ULC ucoms2...@gmail.com wrote:
Anyway to Increase Volume gain in Asterisk ?
USING g729 codec.
___
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asterisk-users mailing list
Hello,
I have an Asterisk system in the UK using ISDN service from BT.
My problem is that the called number is always passed to the provider with the
Type Of Number declared as “national” despite pridialplan (and
prilocaldialplan) is set to “unknown”.
My questions are:
- How can I find out
Actually, either this guy is really stupid or he/she makes some sort of
experiment who will start sending traffic and then try to catch those
_bad_guys_ ...
-
razu
On 11/19/2009 10:31 PM, Doug Lytle wrote:
Steve Howes wrote:
On 19 Nov 2009, at 19:32, baba jigger wrote:
AXvoice
Can someone please share with me a sip configuration to connect an asterisk
server to a voip provider since my configuration isn't working for me.
thanks.
--- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote:
From: Landy Landy landysacco...@yahoo.com
Subject: Re: [asterisk-users]
why would you need to increase gain in Asterisk ? Also do you need it on
E1 (this would be the only place where you would actually need it) ?
-
razu
On 11/19/2009 11:11 PM, David @ULC wrote:
No way ??
On Thu, Nov 19, 2009 at 8:59 AM, David @ULC ucoms2...@gmail.com
mailto:ucoms2...@gmail.com
On Thu, Nov 19, 2009 at 10:18:32PM +0100, Solt Kemecsei wrote:
Hello,
I have an Asterisk system in the UK using ISDN service from BT.
My problem is that the called number is always passed to the provider with
the Type Of Number declared as “national” despite pridialplan (and
Hi,
I just started with Asterisk as I am very unhappy with the functionality
of my current PBX at home. I try to understand everything and play
around, but it is not as easy as I thought. So please be patient if this
is a too easy question for You.
I installed Asterisk 1.4.26.3 on a Debian Lenny
Sorry if this is off topic
I have a loud talker in our call center and was asked if I can make
his voice louder to make him talk softer :-)
Does anyone know if you can do that with Polycom 430's
I found voice.gain.tx.headset but wasn't sure if that will make his
voice louder to the calling
Hi friends,
I want opinions to resolve a problem about interferences with Dahdi
Channels.
I have rooms with many users in each one, when I try to spy this
channels all conferencist listen a noise like a cavern or tunnel and
all rooms are mixed, people listen people of anothers rooms.
I'm using
I was dealing with an issue for a few weeks with my Gateway randomly
crashing (Didn't matter what version of asterisk, sangoma firmware,
etc)... I finally hooked up a modem cable to serial console and was able
to catch the crash.. Wanpipe was causing it
I spoke with Sangoma and they said
On Thu, Nov 19, 2009 at 3:36 PM, Landy Landy landysacco...@yahoo.comwrote:
Can someone please share with me a sip configuration to connect an asterisk
server to a voip provider since my configuration isn't working for me.
thanks.
Who is your voipprovider? Did they give you the settings
Tzafrir,
Thanks for the tip, I am going to give it a try.
Any hint on how to check the actual value of pridialplan variable without the
need of making a test call?
Thanks,
Solt
On Thursday 19 November 2009 at 22:43:53 Tzafrir Cohen wrote:
On Thu, Nov 19, 2009 at 10:18:32PM +0100, Solt
This isn't necessarily apparent, but users.conf has control over some
CALLERID variables. You might want to look there as well.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Solt Kemecsei
Sent: Thursday,
Nothing. I don't know what in the world is going on with my setup.
Here's my FORWARD rules:
eth0 = external nic, eth1 = lan
0 0 ACCEPT udp -- eth0 eth10.0.0.0/00.0.0.0/0
udp dpts:5060:5070
0 0 ACCEPT udp -- eth0 eth10.0.0.0/0
On Thu, Nov 19, 2009 at 4:53 PM, Landy Landy landysacco...@yahoo.comwrote:
Nothing. I don't know what in the world is going on with my setup.
Here's my FORWARD rules:
eth0 = external nic, eth1 = lan
Did you try the config I provided in my previous email? You should copy /
paste it to avoid
What does your Dial command look like? It should be something like
this:
exten = _9.,1,Dial(SIP/voipprovider/${EXTEN:1})
Also, do you have a register statement for voipprovider in sip.conf?
Does sip show registry show that it's registered successfully?
At 2:53 PM on 19 Nov 2009, Landy Landy
Hi,
I'd say Linphone configuration. I suggest to check Linphone
configuration and also asterisk debug - one of those will give you an
answer (most likely asterisk debug as it shows you what it receives ...
if it doesn't receive anything then Linxphone fails)
--
razu
On 11/19/2009 11:48 PM,
On 19 Nov 2009, at 20:38, Barry Miller wrote:
I forwarded this to techsupp...@axvoice.com, just in case they didn't
already know. I also apologized if I was the 10,000th person to do
so.
Ditto, and left a voicemail for them too (number from whois).
S
Hi list
I am having trouble getting asterisk to perceive the firewall's ip address as
outside localnet (setting in sip.conf). The situation is this:
- phones inside lan work fine when localnet is set to 192.168.0.0/255.255.255.0
- phones outside the lan can't ack the invite from asterisk because
I have the conf provided in last post.
exten = _9.,1,Dial(SIP/voipprovider/${EXTEN:1})
Yes, I have that in the dialplan.
Does sip show registry show that it's registered
successfully?
*CLI sip show registry
Host dnsmgr Username Refresh State
In SIP setting on the e71 I set the public user name as
1...@10.10.11.180. There is a sip.conf context [1995]
On the asterisk CLI I get:
Registration from 'sip:%201...@10.10.11.180:5060' failed for
'10.10.11.98' - No matching peer found
So I changed the sip.conf context to [%201995]
Then:
In SIP setting on the e71 I set the public user name as
1...@10.10.11.180. There is a sip.conf context [1995]
I can confirm that the Nokia E71 works perfectly fine with Asterisk.
It looks like you have a space between sip: and your username in your
SIP Profile on the phone. If in doubt, remove
%20 usually represents a space in escaped URL format - perhaps you've
inadvertently got a space in front of the username in the SIP account on the
e71?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean
Hi Anees,
Have you tried to monitor the number of active channels?
Something like:
watch 'asterisk -rx show channels | grep active'
According with your setup the maximum number of active calls should be
7x120=840 - or near this number.
Maybe the calls are not closed properly and you
2009/11/19 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Nov 19, 2009 at 03:46:50PM +0100, Olivier wrote:
Hello,
On a very new system, I've got :
# cat /etc/dahdi/genconf_parameters | grep -v ^#
lc_country fr
context_lines remote
group_lines 1
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