Hi All,
Does anyone have and can share with me an AGI script to dip thinQ for
cnam? oR perhaps dialplan curl using curlopts?
Thanks.
JR
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Chasing the Azeotrope
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et/wp-content/uploads/2007/08/DUNDi_So_Easy.pdf
Good luck!.
JR
--
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Check out the new Asterisk c
it master branch, I'm not running git master
in production but could really use this functionality. Any ideas on
how I could backport/patch UnicastRTP to another branch?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
Chasing the Azeotrope
--
__
> On 15-10-20 07:18 PM, JR Richardson wrote:
>> Hi All,
>>
>> I playing around with multicast paging, I saw a post from Josh Colp
>> about adding unicast support into chan_multicast_rtp but not finding
>> details if this is incorporated in dialplan functions or not
-director cisco router.
Can anyone point me in the right direction?
Thanks.
JR
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Chasing the Azeotrope
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New to
We have a couple of positions open, please contact me off-list if interested.
http://www.ntegratedsolutions.com/voice-engineer-dallas/
These are full time positions in Dallas, no telecommuters please.
Thanks.
JR
--
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Chasing the Azeotrope
Hi All,
Simple scenario:
7940 SIP>http://www.api-digital.com --
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voicemail.c manually from the patch file (revision 233691),
recompiled and now prepending voicemail works.
Thanks.
JR
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y running for a customer,
1.6.0.28.
Does anyone have a patch file that will apply to this version or an
app_voicemail.c file that is already patched and will compile with this
versions to fix this particular bug?
Thanks.
JR
--
JR Richardson
Engineering
g if
these will work with vanilla Asterisk system or are they hard wired for
Allwork systems only? Any feedback is appreciated.
Thanks.
JR
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Ntegrated Solutions in Dallas, TX is still looking for voice guy. This
position is for US hire only, will not sponsor H1B work visa.
http://www.ntegrated.net/careers/
Thanks.
JR
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Engineering for the Masses
Hi All,
Ntegrated Solutions is looking for a full-time Asterisk/Telecom Tech and a
.net/php developer.
http://www.ntegratedsolutions.com/careers/
Forward resume' to j...@ntegrated.com
Thanks.
JR
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Engineering for the M
well?
Any guidance is appreciated.
Thanks.
JR
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;
You may want to check out this presentation form the last Astricon, it
may be relevant:
http://www.astricon.net/2012/videos/Automated-Hacker-Mitigation.html
Cheers.
JR
--
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> JR Richardson wrote:
>>> My bad. I sent Igor to the boneyard to fetch 1.6.0.28 and it appears to me
>>> that by commenting out lines 309-312 and doing a fresh make you eliminate
>>> the extra files (or make them empty).
>>>
>> Appriciate the sug
> Just add noload=cdr_csv.so to modules.conf
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JR Richardson
> Sent: Friday, October 19, 2012 5:09 PM
> To: asterisk-users@lists
Hi All,
I would like to disable the cdr account logs but in 1.6.0 but the
'accountlogs=no' switch is not available till 1.8 as far as I can
tell. Is the any switch I can turn off int he Mkae file for the
cdr_csv.so module to disable accountcode logs?
Thanks.
JR
--
JR Richardson
E
nat: NULL
> allow: ulaw
> disallow: g729
> insecure: invite
> callerid: NULL
> rfc2833compensate: NULL
> mailbox: NULL
> session-timers: NULL
> session-expires: NULL
> session-minse: NULL
> session-refresher: NULL
>
terface, brand with my business logos, add or remove configuration
elements. I kind of like the Digium Asterisk GUI but I'm just not
real familiar with it, just test driving it a bit. What I do like
about it is the flat file manipulation, no database needed.
Any guidance is much appreciated.
to the new 1.8 call
servers and on to the carrier. I don't know why this seemed to fix
the issue, I'm not 100% convinced it really did. I reverted the
change and could not reproduce the issue, so I also suspect the
upstream carrier started working or may have changed something
co
Hi All,
I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in
routing calls to upstream carrier via SIP trunks out. I spent a lot of time
in the lab testing 1.8 which included heavily testing DTMF with no issues
that came up. It all just seemed to work fine. But then a
've never had a problem with stability. I also use the cdr_mysql as well.
I wrote a couple of papers on asterisk_clustering_with_mysql_replication.
They are a bit dated but still relevant. I'll send over if you like.
Good luck.
JR
--
_
On Mon, Oct 3, 2011 at 5:01 PM, JR Richardson wrote:
> Hi All,
>
> Trying to upgrade some call servers, in the lab making sure all my
> applications work, ran into an issue with some manager perl scripts
> that pull and reset database info, it seems the command and result
> resp
nd(Event => 'DBGetResponse');
my $cnamreset2 = $result102[1];
##disconnect the manager connections##
$astman1->disconnect;
$astman2->disconnect;
print "Total CNAM Count for last month is $cnamtotal\n\n
on what else
to look for?
Thanks.
JR
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in the
> logs is 4-5 attemps before ban is applied. I am calling scripts that
> apply the ban to a cisco access-list, so there is script/telnet/config
> delay but it is very minimal and works very well.
>
> JR
>
> Speaking blindly as someone who has yet to fool with F2B, I&#
.
>
> So I forge one SIP packet and I get you to block the IP address of your
> SIP trunk (or your IAX trunk)?
>
> Cool!
>
> --
> Tzafrir Cohen
Good thing I ignore my own IP blocks
lnet/config
delay but it is very minimal and works very well.
JR
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] section of asterisk.conf
with no effect.
Thanks.
JR
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Hi All,
I see some patches about adding atxfer beep sound in the sip channel,
but I'm not clear on when this was implemented in what version?
I don't see the added function in chan_sip in 1.2.24 or 1.4.21 or 1.6.0.28?
Where is this code implemented, what stable release?
Thanks.
Dallas, no telecommuters please.
Thanks.
JR
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for any clarification.
JR
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New to Asterisk? Join us for a live introductory webinar every Thurs
le with Asterisk. Keep in mind, Cisco has no
incentive to make their SIP firmware work with any other platform other than
there servers so I don't really expect it to work properly if at all with
Asterisk. 7.5 is the only firmware version that I deploy on a few hundred
units and works fine.
Go
?
Thanks.
JR
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http
> Date: Thu, 24 Jun 2010 15:32:39 -0400
> From: Ben Winslow
> Subject: [asterisk-users] T.38 on a MAX/Lucent/Ascend TNT
> To: asterisk-users@lists.digium.com
> Message-ID: <4c23b2d7.9090...@pa.net>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hello folks,
>
> I've been tryin
Ditto I'm running a Supermicro Atom based dual core server and it's rock
solid!!!
These make excellent servers for Asterisk installation IMHO.
On Fri, Jun 11, 2010 at 05:42, --[ UxBoD ]-- wrote:
>
> - Original Message -
> > On Thu, 10 Jun 2010, Michelle Dupuis wrote:
> >
> > >
usy
with other matters. If anyone here would like to pick up the torch
and move this along, I can certainly provide info on how far along we
got and contact info for the parties involved.
Please contact me if you have time to work on this and are interested.
I'm sure the Proj
k2,${EXTEN},1:siptrunk1,${EXTEN},1)
[siptrunk1]
exten => _X,1,Set(GROUP()=siptrunk1calls)
exten => _X,n,Dial(SIP/${ext...@siptrunk1,60,)
[siptrunk2]
exten => _X,1,Set(GROUP()=siptrunk2calls)
exten => _X,n,
Thanks Steve, works great:
exten => _X.,1,Set(uniqueidcut=${CUT(CDR(uniqueid),.,2)})
exten => _X.,n,Set(result=${MATH(${uniqueidcut}%2)})
exten => _X.,n,GotoIf($[${result} > 0 ]?siptrunk1,1:siptrunk2,1)
Thanks.
JR
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--
or guidance.
Thanks.
JR
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lution was to downgrade
to another version or switch to 1.2 or 1.6 depending on what features
I need for the system.
Sorry I couldn't be of any help, but I feel your frustration.
JR
--
JR Richardson
Engineering for the Masses
--
___
iguration, just switching out the
ATA. I have the latest firmware on each unit. Any ideas on what could
cause this? The configuration is pretty simple so I don't think I'm missing
anything there. I'm guessing there is a built in speed limit on
c Speed, Disc I/O? Would there be a problem running 3 to 4
PRI's full of T38 to SIP Faxes on one server? Could the Attrafax
software handle that volume?
Thanks in advanced for any feedback.
JR
--
JR Richardson
Engineering for the Masses
--
'host=ip address' is not being looked up.
It works in 1.4 but not in this version. I'll do some more debugging
and try to figure out what is going on.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
--
_
t=ip address
context=provider_1_incoming
or something like this:
[from ip address]
type=trunk
context=provider_1_incoming
authentication=none
Thanks.
JR
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Engineering for the Masses
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On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson wrote:
> Hi All,
>
> I'm using Asterisk 1.4 branch and checking the status of some SIP
> Peers with the functions ${SIPPEER(101:status)} and the result is "OK
> (48 ms)". Seems to work fine.
>
> Now I would l
"OK (48 ms)" and then do some follow on
stuff if the status is OK.
I'm running into syntax errors in the Set command, I think due to the
spaces in the SIPPEER status.
Any suggestions on how to deal with the 'spaces' in the status?
Thanks
hat can be identified and resolved.
Or maybe suggest another version of 1.4 that does not have an issue
like this at these volumes?
Thanks.
JR
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f the AGI server in your dialplan?
>
Ok, I went with #4 for a bit, then resolved to #5 (pardon the pun), works fine.
Thanks.
JR
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Engineering for the Masses
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> >> On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson
>
> > wrote:
> >>> problem I'm running into is if the DNS server is not responding, the
> >>> script hangs and waits for 30 seconds before returning to the Asterisk
> >>> dialplan. ?I
arse();
# Set variables according to supplied arguments
$number = $ARGV[0];
$AGI->exec("agi","agi://agi.server.com/script.agi?user=username&number=$number");
***
Any assistance will be appreciated.
Thanks.
JR
--
> On Monday 28 December 2009 23:49:13 Tilghman Lesher wrote:
>> On Monday 28 December 2009 18:09:15 JR Richardson wrote:
>> > I turned on console debug to see the actual mysql queries and to my
>> > surprise and concern, I see every query for an extension priority
&g
ySQL RealTime: Everything is fine.
test1-6*CLI>
Any guidance on trouble shooting this will be appreciated.
Thanks.
JR
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Engineering for the Masses
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asterisk-users
>> for IFPs, and does not allow successful FAXing at any possible bit rate
>> (except for 2400 bits per second using 10 millisecond IFPs, but no FAX
>> stack would do that).
>>
I was having similar issues, trying Asterisk 1.6.1.12-rc1 resolved it.
http://www.mail-archive.co
particular area and also thank the dev team for
responding to the bug tracker, taking suggestions for improvements and
doing the coding to make Asterisk the best it can be. I can't wait
for T38 gateway. Keep up the good work.
Thanks.
JR
--
JR Richardson
Engine
ng like that.
Thanks.
JR
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rking.
I'm wondering if anyone has tried the new Cisco 2430 series IAD's and have
been successful and reliable, care to share your experience and sample
configs?
Thanks.
JR
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Engineering for the Masses
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can send it to me, gsm or ulaw?
Thanks.
JR
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s an Cisco IOS bug or
maybe a router overload? I've searched for a cisco nat bug with no
luck.
So my question is, has anyone else experienced this type of issue and
if so, is there a solution to resolve?
Thanks.
JR
--
JR Richardson
Engineering
Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
FXO port and 1 FXS port. I have a POTS line from my phone company attached
to the POTS line.
I have asked for "disconnect supervision" to be provisioned o
g that I'm doing terribly wrong,
> that would break everything and make the universe collapse into itself
> when I apply the same principle on production?
>
> I'll be happy to provide more details in case there are any doubts. I
> really appreciate your feedback, no matter
Ok on the workaround, how would I implement it? I'd like to give that a
shot.
On Tue, Sep 15, 2009 at 12:23, Danny Nicholas wrote:
> The issue is that POTS as a technology does not have Answer/Hangup
> Supervision control (This is per the good folks at Digium). Your local
> Telco may or may not
, the device sends out the correct digit tone
associated with that character, like on a regular phone keypad.
That is how folks can use a Blackberry effectively with the PBX
Directory application.
Hope this helps.
JR
--
JR Richardson
Engineering for the M
m-goodbye
incomingconf136 6 Hangup
JR
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the POTS line config for the IAD.
Thanks.
JR
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y anyway so no disruption of local function when the user
is in the office.
Hope this helps.
JR
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To UNSUBSC
326,n,GotoIf($["${cf_active}" = "no" ]?:326|20)
exten => 326,n,Goto(cfaccess,${cf_cfnum},1)
So I have in mind that maybe the function is a bit more versatile than the
old app, but I don't really see it just yet.
Can
but it make a good statement about why their prices where
high for the past few years. The technology and how it is delivered has not
changed much over the past year so the "Economic Downturn" has affected them
enough to reposition their margin strategies.
JR
--
JR Richardson
ain.
We are using the same firmware on the phone that worked fine with the
Asterisk 1.2 code, Polycom 650 with 2.1.1. So I'm guessing there is
something particular with this version of Asterisk. Any guidance will be
appreciated.
Thanks.
JR
--
JR Richardson
Engineering f
there possibly a patch to addons that would relieve this issue?
Thanks.
JR
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To UNSUBSCRIBE or update op
you could check this with the etchandahalf (2.6.24) kernel?
That was going to be my next step if I couldn't get it resolved
Thanks,
Stephen
On Thu, Dec 18, 2008 at 11:57, Tzafrir Cohen wrote:
> On Thu, Dec 18, 2008 at 10:49:03AM -0500, Stephen Brown Jr wrote:
>> I'm having troub
I'm having trouble with ztdummy and I can't seem to figure it out. I
am running Zaptel 1.4.12.1 under Debian 4.0 with latest updates
applied and I have compiled Zaptel from source along with a new kernel
from Debian sources to include 1khz timer support.
The modules build fine, yet when I load the
the web page does not change, always shows 'not in use'. The page
does update with 'Last In Call' info after hangup of a call.
Any ideas?
Thanks.
JR
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,0)
kernel /boot/vmlinuz-2.6.18--686 root=/dev/sda1 ro acpi=off
initrd /boot/initrd.img-2.6.18-686
savedefault
Reboot, and that should do it.
JR
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I noticed that the vicidial site has documentation available which probably
covers the topics required. However, I also see that they want $50-$100 to
download the docs. Seems harsh.
Ron Byer Jr.
NetWeave Integrated Solutions, Inc.
+1.732.786.8830 x120
-Original Message-
From
to do is embed the username:password in the Dial
string, something like this:
exten => 1234,1,Dial(SIP/[EMAIL PROTECTED]:[EMAIL PROTECTED]|30|)
doesn't work though, can't create sip channel.
I'm not sure if this can be done?
Any guidance will be appreciated.
JR
--
-
JR R
rrors. I started
getting protocol and various page errors. I tweaked every T38
parameter that Audiocodes had with zero improvement.
So I have to say, my confidence in T38 is very low, at least where
open Internet connections are being used. I'm now going to look at
some other technology,
y there is no impact to a running system.
JR
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ll
setup fine with audio.
Thanks.
JR
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Regis
I currently have two T1 PRI lines feeding our company's legacy PBX. All
our numbers are DIDs and can pass over both PRI. Currently, if one PRI
line (or T1 interface card in the PBX) is down, communication continues
to function as normal (with the exception of the reduction of channels
available f
> Is this a one VIP to one cell number match? Or is it on VIP to multiple
> cells?
>
> On Tue, Aug 26, 2008 at 7:28 PM, JR Richardson <[EMAIL PROTECTED]>
> wrote:
> > Hi All,
> >
> > I received a request for a special application and need some guidance.
&g
and deleted, through a web page on the PBX.
So I'm thinking I need a dialplan app that has to interface with a
MySQL database that holds the list of numbers, so I can build a
webpage to add/delete the numbers.
Any ideas would be much appreciated.
Thanks.
JR
-----
JR Richa
Hi All,
I finally got the time to test t38 pass through with a TNT, * 1.4.21.1 and
Linksys 2102:
PRI><2102> ___
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Register Now: h
Hi All,
I've been testing reliability with t.38 faxing pass through with * 1.4.21.1,
Linksys ATA's 2102 x 2, Sharp UX-B800SE and Cannon ImageClass D880.
cannon> <2102 #1> <*> <2102 #2> ___
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.6 is all
about, capability, functionality, call flow to what application, library
requirements, spandsp versioning.
And when do you think we can expect to see stable solutions for each.
Thanks.
JR
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t;
Use DUNDi, perfect for this. The protocol is very light, no load on
the servers to run it, can handle hundreds of queries a second with no
load. You want to use regcontext and a few other things to make it
all work together. Here are some papers to guide you:
ftp://208.81.55.228/DUNDi_So_Easy.pd
100+ user environment with high call
volume and high chat volume. Java seems to be a bit resource hungry
with the user notifications and call pop ups. I would hate to have
the IM server walking over Asterisk and affecting call quality or PBX
stability.
Thanks.
JR
-----
JR R
his is not a max tnt problem, the tnt will pass anything you send to it,
911/411/7 digit/10digit/011 international, the question is, does your PSTN
provider accept 911 call on the trunk your passing the call to?
JR
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My test is connecting fine to local and remote databases, I'm use
Asterisk 1.6-current and addon-1.6-current from digium ftp, not trunk.
Hope this helps.
JR
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ge and the addon package did find mysql
installed. So I'm guessing debian etch is putting mysql_client in
some other place that /usr/sbin/.
What I did notice is the addon sample config file for res_mysql.conf
doesn't specify how to setup the read/write entries, clarification on
that would hel
Hi All,
I'm poking around with 1.6, tried to compile the addon package, but it
doesn't see mysql_config installed.
I have mysql-client, mysql-common and mysql-server installed. I'm
running debian etch.
Any suggestions?
Thanks.
JR
--
JR Richardson
Engineering
> 1-2ms latency end to end tops...
>
> If the ata is reinviting to the MAX TNT shouldn't fax work with T.38...
> Does anyone have any experience with this configuration ?
> Thanks,
I have been wanting to do this for months, but just can't find the
time to work on
= 3db-loss
set line-interface voip-gain-control output-pad = 3db-loss
Hope this helps.
JR
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> JR Richardson wrote:
> > I have 1.4.9.1 setup, with the compiler flags enabled for T38, and
> > have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes
> > between devices but can't seem to invoke T38 pt UDPTL. It's enabled
> > in sip.conf [gener
RNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite
after T38 session not handled yet !
sip show channels shows the call setup with ulaw.
Any guidance will be appreciated.
Thanks.
JR
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___
--
that gets called, so after a week or month,
I can see how many times a specific dilaplan action has been used.
Thanks for any advice.
JR
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Mike Hammett wrote:
> I need to setup a small mail server on a local network. It only needs
> SMTP ability as it's just so Asterisk can send out emails. The machine
> has sendmail installed. My primary mail server seems to be rejecting
> the messages. Some research says something isn't confi
ndom reboots are not uncommon on the 601 with sidecars if you're
> running it on PoE.
That makes sense but in my case the 601 w/3 sidecars did not reboot at all
and it is run from POE. The 650 just seems to perform much better.
JR
---
JR Richardson
Engineering for the Masses
_
JR Richardson
Engineering for the Masses> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of asterisk-users-
> [EMAIL PROTECTED]
> Sent: Saturday, March 01, 2008 12:00 PM
> To: asterisk-users@lists.digium.com
> Su
processor,
will this eliminate the issue?
Has anyone experienced this or have ideas for resolution or further
troubleshooting?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
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