Hi Sherwood ,
well , i think you did not understand my question , i want real time billing
like as i mentioned that if i want to dial 5 number with different call rate
how can i access same
balance into those 5 people, if all are connected how can i periodically
update billing , as you suggested
On Thu, 2010-10-21 at 01:15 -0400, Zeeshan Zakaria wrote:
Yes, one server will do it all. It will not be in a data center but at
customer premisis, so doesn't have to be 1U.
In that case, how about a dell-server?
And if it is not in a data center, take care of an UPS for both the
server and
On Thu, Oct 21, 2010 at 1:30 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:
Hi Sherwood ,
well , i think you did not understand my question , i want real time
billing
like as i mentioned that if i want to dial 5 number with different call
rate how can i access same
balance into those
Hi bakko,
just as a test, try to add calltokenoptional=0.0.0.0/0.0.0.0 to your
iax.conf.
Giorgio Incantalupo
bakko wrote:
Hello,
I'm trying to conect two 1.6.2.13 Asterisk server with IAX.
This is my configuration:
Asterisk A:
iax.conf
register = coiax:pa...@69.164.207.166
[smiax]
Have you tried playing with joinempty and leavewhenemèpty to avoid
people being connected to a queue with all agents in use?
l.
2010/10/20 GBR Icasiano, Ryan A. raicasi...@globalbridgeresources.com
Hi,
Is there a way to know if a member of a queue is currently engaged on a
call? Or if a
Hi,
Here is the scenario:
1. 1st phone calls and asterisk dials to extension no.
2. Extension answers 1st caller(which makes it busy).
2. 2nd phone calls and asterisk dials to extension no.
3. 2nd phone hears a BUSY tone, but have to wait for the timeout to expire(in
DIAL cmd) before proceeding
thanks mate,
for useful and good information provided by you, i am not asking you that
please write down your all LOGIC and explain everything to me, as per your
explanation i can see it will deduct amount for only 1 call but what
actually i am searching for is if user made 5 concurrent calls and
Hi,
I didn't use that feature since i only added the phones not treated as
agents(it will just ring the members, depending on the scenario chosen, instead
of ringing the queue itself until an agent answers).
The queue status is correct, although it could not tell if all members in the
queue
Here I am expecting to be configured following scenario:
User calls : it will play a sound will ask for input DTMF, then call will be
given to particular extension for any DTMF entered.
But its not working as expected.
I have attached the dial plan file.
1.vdp
Description: Binary data
--
On Thu, Oct 21, 2010 at 3:23 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:
thanks mate,
for useful and good information provided by you, i am not asking you that
please write down your all LOGIC and explain everything to me, as per your
explanation i can see it will deduct amount for
On Thu, Oct 21, 2010 at 02:46:16PM +0530, Jigar Joshi wrote:
Here I am expecting to be configured following scenario:
User calls : it will play a sound will ask for input DTMF, then call will be
given to particular extension for any DTMF entered.
But its not working as expected.
I have
On 21 Oct 2010, at 10:16, Jigar Joshi wrote:
I have attached the dial plan file.
In what format?
S
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If you look at it the way you want it.. you usually tell your customer the
available funds and minutes in their account right?
How will you explain politely that you have dropped their calls for lack
of balance because someone else used their account?
If you don't tell them their balance and call
Tarek,
I'm not sure why it would be our problem is someone came into your
office and started making long distance calls over a trunk I was
providing your company I'm pretty sure that if I had tried that
with some of my carriers in the past they would have laughed until
they cried...
Oh, and
actually my mail was not meant to be disrespectful. it was an inquiry. i have a
billing system and had a few of those thoughts regarding real time billing. my
issue was explaining to a customer that his call disconnected an hour earlier
because someone else used his account.. I'm doing retail
No transcoding? OK, this will work...
http://www.supermicro.com/products/system/1U/5015/SYS-5015A-PHF.cfm?typ=E
~
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
I think I'll prefer Dell over supermicro, as another customer I worked for
always complained about supermicro. I also once used supermicro and I had no
luck with it.
But which model of Dell is good for this requirement? I don't want to get
over powerful server than required for this setup.
That would be really difficult to do, to keep track of all three channel
events while they are originating and to hangup the failed ones .
Easy solution in asterisk for this would be to originate using Local channel
and in dialplan use Dial command to make call to all the operators using ''
Maybe you should post this portion for your dialplan. I have done the same
thing several times and never had this timeout issue.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-21 4:08 AM, GBR Icasiano, Ryan A.
raicasi...@globalbridgeresources.com wrote:
Hi,
Here is the scenario:
1. 1st
Hi,
Which asterisk version are you using. try setting call-limit value in
sip.conf and see if it makes any difference.
On Thu, Oct 21, 2010 at 1:29 PM, GBR Icasiano, Ryan A.
raicasi...@globalbridgeresources.com wrote:
Hi,
Here is the scenario:
1. 1st phone calls and asterisk dials to
Hi,
I am not using 1.6 but in 1.2 or 1.4 there is no straight forward way to do
this. The workaround i use it to pull the caller into conference and play
what ever I want using a agi script connected to the same conference room.
On Wed, Oct 20, 2010 at 4:35 PM, Gustavo Garcia Bernardo
There is always one (or more) bad product from every manufacturer that
leaves a bad taste in your mouth. Always keep you mind open and
search before you hit the order button. For supply chain I like
Supermicro. I don't like all their products but I know I can get the
right part with one order
We have an employee who works from home. We sent her a SIP phone to work as an
extension off our Asterisk 1.6 system, but her DSL service is so bad she was
dropping calls all the time. It's not just a tuning or QoS issue. Her service
is simply unreliable.
She had a POTS line installed and I
Hello friends,
I'm trying to make a simple call from asterisk CLI, but is quite confuse
i followed the information here:
http://www.voip-info.org/wiki/view/Asterisk+CLI+dial
and changed my extensions.conf like this:
alsa.conf
[general]
autoanswer=no
context=consolecontext
extension=100
By
I just put in an HP DL360 G6 for a client spec with a Sangoma 4x PRI, a Sangoma
4x FXS and about 150 devices. Running live now on 1x PRI approx 20-calls and
60 phones, load is at zero.
We went with the base machine Xeon 5500 + 4GB RAM, 2x PS, 2x HD (raid
mirror)... about 3K$ but Im certain we
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias
Sent: Thursday, October 21, 2010 7:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] dialing from asterisk console?
Hello
Try:
dial 1...@consolecontext
-Original Message-
From: Danny Dias ing.diasda...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 21 Oct 2010 14:41:47
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users
Hello,
Looking the asterisk 1.8 API documentation
(http://www.asterisk.org/astdocs/api/index.html), I see a lot of new
fields for sip register uris:
register =
[peer?][transport://]us...@domain][:secret[:authuse...@host[:port][/extension][~expiry]
But the *peer* is not explained
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins
Sent: Thursday, October 21, 2010 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Busy detection in
Yhank you very much Giorgio,
now work with the general option:
calltokenoptional=0.0.0.0/0.0.0.0
Regards
- Bakko
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On 10/20/2010 04:07 PM, VoIP Question wrote:
Hello again,
If I set a peer to use G.711 only, they try to process a sent fax in
G.711, but Asterisk doesn't like it:
WARNING[4903]: res_fax.c:1709 sendfax_t38_init: Audio FAX not allowed on
channel 'SIP/Main-000a' and T.38 negotiation
Hi Bruce,
can you show agent login/logoff diaplan?
Maybe there is a solution but i have to know how yours agents login/logoff.
Regards
- Bakko
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On 10/20/2010 11:35 AM, VoIP Question wrote:
On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
This was fixed in Asterisk 1.6.2.12 and later releases, so if you were
running the current version, you wouldn't have experienced
On 10/20/2010 09:35 AM, VoIP Question wrote:
Thank you Kevin,
We'll upgrade our server to 1.6.2.12 and try again.
Another question: Is there (expect for the admin guide that we didn't
succeed to understand the example in) an example somewhere for
ReceiveFax full extensions.conf diaplan?
Here is the login for English:
;English-temp LOGIN
exten = 800,1,Answer()
exten = 800,n,AddQueueMember(500|Local/${CALLERID(num)}...@from-internal/n)
exten = 800,n,Set(DEVSTATE(Custom:agenten)=INUSE)
exten = 800,n,Playback(agent-loginok)
exten = 800,n,Hangup()
;English Logout
;All Queues Logout
Hi,
I setup an asterisk system (version 1.8.0-rc5). While using a SIP only
environment I discovered a problem using blind transfer. The phones are
SNOM or Aastra and are using the SIP REFER Method.
The following is working:
User A calls user B, B accepts the call, user A than transfers to user C
Just done a clean install of rc5 on a totally new machine and found the
following:-
/etc/init.d/asterisk start
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
Dave Cotton
--
We have a couple of positions open, please respond to the posting if
qualified and interested.
http://www.ntegrated.net/resources/job-opportunities/field-service-install-technician
http://www.ntegrated.net/resources/job-opportunities/network-engineering-voice
These are full time positions in
On Thu, Oct 21, 2010 at 10:40 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
What OS are you running?
--
Paul Belanger | dCAP
Polybeacon |
On Thu, Oct 21, 2010 at 11:05 AM, Paul Belanger
paul.belan...@polybeacon.com wrote:
What OS are you running?
If I had to guess SUSE?
--
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Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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On 21 Oct 2010, at 15:56, JR Richardson wrote:
These are full time positions in Dallas, no telecommuters please.
A very vast majority of people on here are not in Dallas (and indeed probably a
majority in the US). So stop filling their mailboxes with this crap.
Incase you hadn't noticed
On 21/10/10 17:05, Paul Belanger wrote:
On Thu, Oct 21, 2010 at 10:40 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
What OS are you running?
On 21/10/10 16:40, Dave Cotton wrote:
More interesting is that after make samples I have no iax2 available.
Adding more info :-
[Oct 21 17:14:04] WARNING[17255]: loader.c:387 load_dynamic_module:
Error loading module 'res_crypto':
/usr/lib/asterisk/modules/res_crypto.so: cannot open shared
On Thu, Oct 21, 2010 at 11:12 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
Suse 11.3 X86_64
Try this patch for the init.d issue:
http://asterisk.pastebin.ca/1969072
after you've applied it rerun:
$ make configs
As for issue 2, I suspect you don't have res_crypto.so built. Can
you
His post may have been of interest to some outside of DFW, and I appreciated
your post less than his. But, enjoy.
C
==
A very vast majority of people on here are not in Dallas (and indeed
probably a majority in the US). So stop filling their mailboxes with this
On Oct 21, 2010, at 11:11 AM, Steve Howes wrote:
On 21 Oct 2010, at 15:56, JR Richardson wrote:
These are full time positions in Dallas, no telecommuters please.
A very vast majority of people on here are not in Dallas (and indeed probably
a majority in the US). So stop filling their
On Thu, Oct 21, 2010 at 11:15 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
Adding more info :-
Ya, so that is the issue. chan_iax2 uses res_crypto, and you likely
are missing libssl-dev (openssl) on our box.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
On 21/10/10 17:19, Paul Belanger wrote:
On Thu, Oct 21, 2010 at 11:15 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
Adding more info :-
Ya, so that is the issue. chan_iax2 uses res_crypto, and you likely
are missing libssl-dev (openssl) on our box.
Yes and ./configure and make
Hi
I wonder if anyone could give some light on SIP NAT.
I've having a friken headache with SIP NAT 1 way audio.
Client - NAT - NAT - Server
Client can hear users from server side
but server cant hear client.
Ive tried every possible settings
externip set
localip set
NAT= yes / route
Hi,
Given the recent increase in SIP brute force attacks, I've had a little idea.
The standard scripts that block after X attempts work well to prevent you
actually being compromised, but once you've been 'found' then the attempts seem
to keep coming for quite some time. Older versions of
Always start here... http://www.spamhaus.org/drop/
If the AS is stolen, you can block the network and never have to worry
about it...
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux
On Thu, 21 Oct 2010, Steve Howes wrote:
Hi,
Given the recent increase in SIP brute force attacks, I've had a little
idea.
The standard scripts that block after X attempts work well to prevent
you actually being compromised, but once you've been 'found' then the
attempts seem to keep
Hello,
I
have a problem of saturation of bandwidth because of HANGUP which sends
thousands of times per second for a single call. Furthermore, the
timestamp is still the same for this HANGUP.
Thanks
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[snipped very confusing top and bottom posting mix]
On Thu, 21 Oct 2010, Sherwood McGowan wrote:
Dhaval,
You're right, I forgot one thing. The frozen table's id column should not
be an autoincrement, it should be set by the insert statement, using the
original method I decsribed for
With CRON or as an init.d you can do many things...
http://www.spamhaus.org/faq/answers.lasso?section=DROP%20FAQ#116
~
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
*
I was thinking on the same lines, i.e. setup a server which will be
regularly updated with these bad IP addresses, and anybody looking to block
bad IPs will be able to get this list from here. For example when I get mail
from Fail2Ban (which I am getting more and more everyday now), a copy would
On Thu, 2010-10-21 at 08:15 +0800, GBR Icasiano, Ryan A. wrote:
anyone?
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A.
On 21 Oct 2010, at 16:54, Jeff LaCoursiere wrote:
I'll subscribe, that is for sure. What is the best way to dist the
blacklist? iptables include file? Or something more integrated to
asterisk... just thinking off the top of my head that a module that vetted
inbound connections against
We would be interested.
Spam is a harder problem to fight due to volume and the ability of any idiot
to set up free email accounts. But anyone blasting SIP systems is a pure
commercial crook. Tagging and strangling them should be a clear cut project.
Cary Fitch
-Original Message-
From:
Tarek,
Ouch, I'm quite sorry. I couldn't sleep when I tried to around 4:30AM
after working on a project all night. Unfortunately, I'm not quite
sure what your question was...
:( Maybe when I wake up a bit more
On Thu, Oct 21, 2010 at 5:38 AM, Tarek Sawah tareksa...@hotmail.com wrote:
Didn't work. It correctly times out after 20 seconds and continues to
voicemail, but the caller still hears the remote busy signal during those 20
seconds.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
On Thu, 21 Oct 2010, Andrew Latham wrote:
Always start here... http://www.spamhaus.org/drop/
If the AS is stolen, you can block the network and never have to worry
about it...
~
Andrew lathama Latham
lath...@gmail.com
I guess you are assuming that spam networks should be included in
On 10/21/10 12:07 PM, Steve Howes steve-li...@geekinter.net wrote:
On 21 Oct 2010, at 16:54, Jeff LaCoursiere wrote:
I'll subscribe, that is for sure. What is the best way to dist the
blacklist? iptables include file? Or something more integrated to
asterisk... just thinking off the top
On 21 Oct 2010, at 17:03, Zeeshan Zakaria wrote:
But the problem is how to make sure that only legitimate users are
contributing to this list. Contributors to this list somehow need to verify
to an admin that they are not hackers, and this the hard part.
I was thinking of having a threshold
On Thu, 21 Oct 2010, Steve Howes wrote:
On 21 Oct 2010, at 16:54, Jeff LaCoursiere wrote:
I'll subscribe, that is for sure. What is the best way to dist the
blacklist? iptables include file? Or something more integrated to
asterisk... just thinking off the top of my head that a module
On 21 Oct 2010, at 17:32, Jeff LaCoursiere wrote:
I agree in principle - some cron job pulling the list by http would
certainly be simple. But just to continue my thoughts to the brick wall,
I don't see a lookup adding latency to the call other than what should
be a very brief addition to
Always start here... http://www.spamhaus.org/drop/
If the AS is stolen, you can block the network and never have to worry
about it...
I guess you are assuming that spam networks should be included in the
blacklist by default? I'm not sure that is a good assumption. Some of my
customer
On Thu, Oct 21, 2010 at 11:25 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
Yes and ./configure and make menuselect did not signal it. :(
Did the patch at-least work for you?
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Hi to all, I am in the process of setup a new asterisk server, I think in
the HP Proliant ML350 G6 Server (aprox. 100 SIP Users), and Sangoma A102DE
Card.
The specs of the Proliant (HP PART 487932-001) about PCI are the next.
1 ( 1 ) x PCI Express 2.0 x16 ( x8 mode ) ,
1 ( 1 ) x PCI
On 21/10/10 19:26, Paul Belanger wrote:
On Thu, Oct 21, 2010 at 11:25 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
Yes and ./configure and make menuselect did not signal it. :(
Did the patch at-least work for you?
I'd already edited the init file so I didn't use it..
Dave Cotton
The Asterisk Development Team is proud to announce the release of Asterisk
1.8.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
Term Support (LTS) release,
I have a 100MB internal lan. aastra's are wired. asterisk box is wired
next to the switch. But look:
sip show peers
142/14210.10.10.42 D A 5060 OK (137 ms)
144/14410.10.10.44 D A 5060 OK (136 ms)
145/145
Am 21.10.2010 19:30, schrieb Ricardo Melendez:
Hi to all, I am in the process of setup a new asterisk server, I think in
the HP Proliant ML350 G6 Server (aprox. 100 SIP Users), and Sangoma A102DE
Card.
The specs of the Proliant (HP PART 487932-001) about PCI are the next.
1
Am 21.10.2010 20:03, schrieb sean darcy:
I have a 100MB internal lan. aastra's are wired. asterisk box is wired
next to the switch. But look:
sip show peers
142/14210.10.10.42 D A 5060 OK (137 ms)
144/14410.10.10.44 D
Hi all,
After a lot of trouble with a TE110p working with mfcr2 , brazil variant,
everything looks great,but I can not go out of my calls.
When I try I receive the following log:
== Using SIP RTP CoS mark 5-- Executing [33220...@local:1]
Dial(SIP/4804-001a, DAHDI/g11/33220567,,T)
On Thu, 2010-10-21 at 17:12 +0200, Dave Cotton wrote:
On 21/10/10 17:05, Paul Belanger wrote:
On Thu, Oct 21, 2010 at 10:40 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make
This did the trick! Masks the busy signal. Thanks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, October 21, 2010 1:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Hi Bruce,
with this configuration you can`t control the state of agent.
Sorry
Regards--
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Hi,
After some changes, the status now is:
== Using SIP RTP CoS mark 5-- Executing [21341...@local:1]
Dial(SIP/4804-, DAHDI/g11/21341400,,T) in new stack == Everyone is
busy/congested at this time (1:0/0/1)-- Auto fallthrough, channel
'SIP/4804-' status is
I use the following:
Exten = s,n(status-NOTIFY),System(echo '${DIALSTATUS} on
${CALLERID(num)}' at ${STRFTIME(${EPOCH},,%H%M%S)} | mail -s Call
Unsuccessful on DNIS '${ARG10}' neeraj.ch...@ocis.com.au)
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Hi folks,
How would I go about running a stored procedure call from asterisk via
func_odbc.
I'm after an example entry in func_odbc if possible for ast 1.4
Also, if someone could post an insert statement that actually works,
would be nice.
Thanks,
:)
--
Hi,
I have modified the way agents are being treated since they are using mobile
phones. Having that kind of scenario, it is not recommended to make the agent
logged in by using that scenario. Instead, they will call a certain number,
login by using the given parameters(company id, username,
Is it already Friday?
This week Counterpath has two big stories. Todd Carrothers, VP Product
Management and Mike Doyle, VP Technology will be on board to tell us
more about these two developments and to answer your questions on VUC
at 12 noon EDT.
1) Counterpath was granted a patent (#
Thanks for the input. By this configuartion you mean by the way I do Add and
Remove member from the Queue?
Can you please explain by what sort of configuration (what to use instead of
Add and Remove queue member) would get this working.
I guess I am looking for speedial/BLF on the same key ?!!!
How can I let asterisk immediately dials a trunk when off hook?
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I am not sure that can be done literally by Asterisk because most
phones/adapters give dial tone when off hook, but Asterisk doesn't know the
phone is off hook until a send button is pushed, several seconds pass after
some keys are pressed, or the # button is pressed.
However many of the adapters
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