Same as the other spans coming out of the same MUX!
signalling = pri_cpe
I got 2 spans working fine from the same MUX. This one isn't working for
some reason.
The all have similar configs, except where I need to change channel number
and primary channel number.
Mike
From:
there is a link, maybe it can help you, please refer it
http://bbs.openvox.cn/viewthread.php?tid=263extra=page%3D4
2010/6/23 Mike l...@net-wall.com
Same as the other spans coming out of the same MUX!
signalling = pri_cpe
I got 2 spans working fine from the same MUX. This one isn't working
On Tue, Jun 22, 2010 at 09:30:39AM -0400, Mike wrote:
Hi,
I have the following happen to me after the restart of one of my servers:
out of my 3 PRIs (all configured with the same technical settings), the last
one isn't coming back. It's underutilized (chances it didn't get a call
Hello List,
I am looking at implementing auto callwait on my asterisk box ( has
single FXS to which my phone is connected).
Is there a way to terminate the current active call (soft-hangup) and auto
accept the new incoming call ? (I don't care for call indication).
Please advice,
We use a dial option A() that will stream audio as soon as the calle picks
up...
On 23 June 2010 05:50, Zhang Shukun bit...@gmail.com wrote:
2010/6/22 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de:
Hi!
but i want to answer the channel when dial someone and pick up the
Plain asterisk. You only configure it once, and re-use the configuration for
different call centers :-)
On 23 June 2010 00:28, Luciano Moreira lmore...@dxbrasil.net wrote:
We use Vicidial for all size CallCenter. It's very powerful for multi
server and/or multi site. We have vicidial from
to re-read peers from realtime db try: sip prune realtime all
On 23 June 2010 01:22, Jean Chassoul chass...@gmail.com wrote:
anyone know something about this?
On Fri, May 14, 2010 at 10:56 AM, Jean Chassoul chass...@gmail.comwrote:
Hi,
I'm using dynamic realtime with asterisk 1.6.0.24,
VicidialNOW (http://vicidialnow.org/)
On Wed, Jun 23, 2010 at 2:21 AM, Alejandro Cabrera Obed
aco1...@gmail.comwrote:
Dear all, I need to build a PBX based on Asterisk for a call center. I
have worked with raw Asterisk but it's hard to work for big
implementations think.
Also I have worked
Addons module is not installed. There is another pbx with just free pbx 2.7
installed and is showing cdrs on reports panel. So, wondering if I'm missing
some configuration on pbx which is not generating cdrs with free pbx 2.5
installed on that or is it because of the version.
Thanks,
I found this link which help me solve this problem
On reading SIP.CONF it say we can add additional local nets this seems to solve
the problem.
; You may add multiple local networks. A reasonable set of defaults
; are:
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local
Hi,
Looking for some reliable and quality providers of USA DIDs.
Any pointers ?
Thx
Sans
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Asterisk 1.6
CentOS 5.0
All -
I'd like to offer my users the ability to hangup a call by pressing **. I'm
using an attendant, so when ** is dialled I'd like processing to return to
the attendant so the user can place a subsequent call. I have setup
features.conf to include:
[featuremap]
On 6/23/10 7:20 AM, RSCL Mumbai wrote:
Hi,
Looking for some reliable and quality providers of USA DIDs.
Any pointers ?
Thx
Sans
We've had good luck with Vitelity and DIDForSale.com.
N.
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Hi,
Although zonedata.c contains ITU E.180 recommendations for Turkey, we are
still experiencing unrecognized hangups from Turk Telekom PSTN lines when
callers hangup.
Turk Telekom does *not* provide supervised disconnects on analog PSTN, and
the tone we receive we when caller hangs up is
Thanks for taking a look. I am using the latest 1.4.33.1 source packages,
(latest libpri, latest dahdi) as of yesterday.
The link is definitely not down, but since this is delivered via DS3 with a
MUX I assume that only means that the wire from the MUX to the Digium card
is ok, but says nothing
I am getting a SIP 401 unauthorized message.
My public IP or PIP is being pre-routed with iptables to goto an
internal IP or IIP
All the polycom phones in the office point to the IIP. they work fine.
I have 2 external phones that are registering to the PIP. I see the
register attempt
as I am
Thank you for your comments and suggestions !!!
Now I will start to read about the different products you mentioned,
and take a decission.
By the way, a friend of mine suggest to me Trixbox Pro Call Center
Edition (paying some $$$) because he says this product has several
features directed for a
On Wed, Jun 23, 2010 at 8:44 AM, Jerry Geis ge...@pagestation.com wrote:
--- Transmitting (NAT) to X.X.X.X:1024 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK6ea01bc7;received=X.X.X.X
From: sip:x...@x.x.x.x.;user=phone
To: sip:x...@x.x.x.x;user=phone;tag=as21ab1732
Please, I need help with this...
Anahi Ludueña
From: a_ludu...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 18 Jun 2010 15:12:25 +
Subject: Re: [asterisk-users] Music on Hold problema
The list of /var/lib/asterisk/mohmp3 is:
-rw-rw 4 asterisk asterisk
The moh conf file seems good. It is the standard implementation and should
have worked. Just wondering if your end devices, whether they are IP phones
or softphones, are setup to listen to some different codecs than ulaw and
slin? Or in your sip.conf when declaring extensions you are not putting
Hello,
I look ARI (Asterisk Recording Interface)
the publisher site is closed...
http://www.littlejohnconsulting.com/ari
Thank you,
Mickael
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New to
didforsale.com is one of the best and reliable DID providers in the USA
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
Date: Wed, 23 Jun 2010 16:50:48 +0530
From: rscl.mum...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place
four thousand calls to what appears to be a toll number in Zimbabwe last
night. Filter 82.150.165.5.
A more overriding problem for me is how do we know what *destinations* to
filter so this idea of war dialing a toll
i faced a similar situation with my ISP .. they block INBOUND UDP port 5060
which means if i try to register.. the server would receive my registration
message.. but when it sends the acknowledgement .. the ISP Firewall rejects the
message so the server responds with Unauthorized.. i simply
On Wed, 23 Jun 2010, Jeff LaCoursiere wrote:
Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place
four thousand calls to what appears to be a toll number in Zimbabwe last
night. Filter 82.150.165.5.
A more overriding problem for me is how do we know what
Agreed! Didforsale.com is THE way to go.
--
Rick Hall
Senior Vice President
ReadyWire Multimedia Solutions
Affordable Website Reseller Hosting
http://www.readywire.com/
(312) 278-4446 x5446
Technical Support:
24 hours a day / 7 days a week
Customer Login...:
It's one of the bad modules that goes with FreePBX anyhow. The moment you go
over 3000 recordings you are already in trouble. It's about time someone
come up with a better moduel.
On Wed, Jun 23, 2010 at 11:05 AM, Mickael Monsieur
mickael.monsi...@gmail.com wrote:
Hello,
I look ARI (Asterisk
you can start by simply telling us what is the purpose of your server.. and
does it have long distance of overseas?? do you use Numeric usernames? simple
passwords? passwords the same as your username? this way you can offer more
info so we can help you.a quick answer will be.. opening a few
On Tue, Jun 22, 2010 at 6:33 PM, Ryan Wagoner rswago...@gmail.com wrote:
--
The out of dialog support was the trick for 1.6.2.9 since it has
support for sending a keep-alive. I have attached a modified version
of your patch that worked for me. Do you mind if I attach the modified
version of
On Tue, Jun 22, 2010 at 8:57 PM, Andres and...@telesip.net wrote:
completely as well.
Below I've posted a patch that responds with a 200 OK to these
keep-alive requests, and I believe
also solves the temporary loss of registration problem, though more
testing in different environments
for
You can look at it a few different ways. Use one or more methods:
1. If you are allowing SIP phones to register from anywhere (inside and
outside your network), make sure all the extensions have VERY strong
passwords (12 characters or more of absolute jibberish).
2. Use deny/permit for
On Wed, 23 Jun 2010, Jeff LaCoursiere wrote:
Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place
four thousand calls to what appears to be a toll number in Zimbabwe last
night. Filter 82.150.165.5.
Ouch. 82.0.0.0/8 is on my block list, available at:
On Wed, 23 Jun 2010, Gordon Henderson wrote:
On Wed, 23 Jun 2010, Jeff LaCoursiere wrote:
Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place
four thousand calls to what appears to be a toll number in Zimbabwe last
night. Filter 82.150.165.5.
A more overriding
On Wed, Jun 23, 2010 at 9:50 PM, Hall, Rick r...@readywire.com wrote:
Agreed! Didforsale.com is THE way to go.
--
Rick Hall
Senior Vice President
ReadyWire Multimedia Solutions
Anyone having experience with didww.com ?
Sorry, I forgot to mention I am looking for wholesale DID --
On Wed, 23 Jun 2010, Tarek Sawah wrote:
you can start by simply telling us what is the purpose of your server..
and does it have long distance of overseas?? do you use Numeric
usernames? simple passwords? passwords the same as your username? this
way you can offer more info so we can
On Wed, 23 Jun 2010, Steve Edwards wrote:
On Wed, 23 Jun 2010, Jeff LaCoursiere wrote:
Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place
four thousand calls to what appears to be a toll number in Zimbabwe last
night. Filter 82.150.165.5.
Ouch. 82.0.0.0/8 is on
On 23 Jun 2010, at 18:39, Steve Edwards wrote:
Ouch. 82.0.0.0/8 is on my block list, available at:
http://www.sedwards.com/class-a-block-list
Would advise people in the UK do not use that list... 82.0.0.0/8 would block a
reasonable chunk of my users for starters..
Steve
--
On Wed, Jun 23, 2010 at 1:57 AM, Tiago Geada tiago.ge...@gmail.com wrote:
to re-read peers from realtime db try: sip prune realtime all
Hi, I don't have a problem with peers or realtime sip! the problem is with
realtime queues I obviously have a queue_member_table for
INSERT/UPDATE/DELETE
On 23 Jun 2010, at 19:26, Steve Howes wrote:
On 23 Jun 2010, at 18:39, Steve Edwards wrote:
Ouch. 82.0.0.0/8 is on my block list, available at:
http://www.sedwards.com/class-a-block-list
Would advise people in the UK do not use that list... 82.0.0.0/8 would block
a reasonable
i consuleted didforsale.com regarding the wholesale thing and their response
was that you should buy a bulk of numbers and make your own api.. one more
thing.. if you are in the USA ..be sure to start your FCC registration (if you
don't have it yet) because it can be a disaster for US
http://www.spamhaus.org/drop/ is a good resource that I use.
~
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 http://en.wikipedia.org/wiki/Tux
On 23 Jun 2010, at 18:39, Steve Edwards wrote:
Ouch. 82.0.0.0/8 is on my block list, available at:
http://www.sedwards.com/class-a-block-list
On Wed, 23 Jun 2010, Steve Howes wrote:
Would advise people in the UK do not use that list... 82.0.0.0/8 would
block a reasonable chunk of my
One thing to take into account and I haven't said before, sorry...
I have 2 pbx, one is connecting to the other by a SIP trunk... The first pbx
has the setting which I put some days ago... the second pbx has the extensions
and I'm trying to use them in the call. Everything is working, except
On Wed, 23 Jun 2010, Jeff LaCoursiere wrote:
On Wed, 23 Jun 2010, Steve Edwards wrote:
On Wed, 23 Jun 2010, Jeff LaCoursiere wrote:
Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place
four thousand calls to what appears to be a toll number in Zimbabwe last
night.
Not sure what kind of provision server you have there. But do not use
http as your provision protocol. Use https instead.
Jian
Jeff LaCoursiere wrote:
On Wed, 23 Jun 2010, Tarek Sawah wrote:
you can start by simply telling us what is the purpose of your server..
and does it have long
Dear Doug and people,
In order to install an E1 Digium card in Asterisk, I've read it's
necessary to have DAHDI. But in my Asterisk installation I have
ZAPTEL.
Is it the same to have zaptel or dahdi in order to put to work my E1
Digium on my server in a plug and play way ??? because I don't
Hi all,
Anyone know why this happens?
Mem:524288k total, 508120k used,16168k free,0k buffers
Swap:0k total,0k used,0k free,0k cached
PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND
1 root 15 0 2152 664 576 S
Zaptel and dahdi is the same thing, except the later one is weirdly named to
make it harder to pronounce. Don't worry to upgrade to dahdi. But it is not
plug and play and you'll need to configure /etc/zaptel.conf and
/etc/asterisk/zapata.conf according to your requirement.
Zeeshan A Zakaria
--
Zaptel and DAHDI are basically the same thing. How different they are
depends on your branch of asterisk, your technology and where in the world
you are.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Reachable from here.
( US -Comcast )
John Novack
Dog is my Co-pilot
Gordon Henderson wrote:
On Wed, 23 Jun 2010, Jeff LaCoursiere wrote:
On Wed, 23 Jun 2010, Steve Edwards wrote:
On Wed, 23 Jun 2010, Jeff LaCoursiere wrote:
Some !...@$#@@# in the Czech Republic used
Le 23/06/2010 21:28, Gordon Henderson a écrit :
[...]
I'd like to have a look, but can't - I think there may be issues with your
registrar for your domain - from where I am, there are no glue records for
the nameservers, therefore I can't look it up... Looks like it was last
edited just over
On Wednesday 23 June 2010 15:45:05 Miguel Molina wrote:
Hi all,
Anyone know why this happens?
Mem:524288k total, 508120k used,16168k free,0k buffers
Swap:0k total,0k used,0k free,0k cached
PID USER PR NI VIRT RES SHR S %CPU %MEM
On Wed, 23 Jun 2010, Gordon Henderson wrote:
Ouch. 82.0.0.0/8 is on my block list, available at:
http://www.sedwards.com/class-a-block-list
If you don't need to receive packets from far away places, it's a
great start.
I'd like to have a look, but can't - I think there may be issues
I'm still trying to figure that out. Our SIP usernames are seven digit
phone numbers, so not really difficult to guess, but the passwords are 7
char alpha-numeric strings, auto generated. We don't at present restrict
people to their addresses, as some are dynamic.
If they're randomly
I'm still trying to figure that out. Our SIP usernames are seven digit
phone numbers, so not really difficult to guess, but the passwords are 7
char alpha-numeric strings, auto generated. We don't at present restrict
people to their addresses, as some are dynamic.
If the extension in
List,
Over the last few months we have managed to bring the total number of
issue on the tracker from 610+ to 537 (as of writing). While this is
good news, we still have a number of open issues that require testers
to help move them along. Below, I have posted the oldest 50 issues
that are in
于 2010年06月24日 04:45, Miguel Molina 写道:
Hi all,
Anyone know why this happens?
Mem: 524288k total, 508120k used, 16168k free, 0k buffers
Swap: 0k total, 0k used, 0k free, 0k cached
PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND
1 root 15 0 2152 664 576 S 0.0 0.1 0:49.26 init
7398
i've got the parking lot set up in asterisk 1.6.2.8.
when a caller calls into the pbx and connects to an extension, the
answering extension can place the person into the parking lot by dialing
#72 during the call via the parkcall parameter in features.conf. this
works just fine.
however,
Hi all,
Just as a heads up o the list the IP above was trying to register with random
names to some of our servers and were flooding them with registration requests.
Dovid
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