? (since I am using SVN?) Or how do I debug and
find what was the root cause of the issue?
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Friday, December 03, 2010 9:49 AM
To: 'Asterisk
If you'll release it for python, I'll take a stab at porting it to perl.
Mike
On October 19, 2017 4:53:52 PM EDT, Jonathan H wrote:
>That's because it uses a deprecated API and endpoint.
>
>However, funny you should ask this, because I've just finished
>updati
il Record (CDR) settings
--
Logging:Enabled
Mode: Simple
Log unanswered calls: No
Log congestion: No
* Registered Backends
---
cdr-custom
Adaptive ODBC
Any ideas would be apprec
il Record (CDR) settings
--
Logging:Enabled
Mode: Simple
Log unanswered calls: No
Log congestion: No
* Registered Backends
---
cdr-custom
Adaptive ODBC
Any ideas would be appreciat
Hi all,
I have a user who would like to stream their favorite radio station from
iHeart radio for their music on hold.
It this TECHNICALLY possible? If so, any pointers would be appreciated.
Is this LEGAL in the US?
Thanks in advance,
Mike
st
\#=participant_count
===
However, my user isn't hearing anything. MoH does work otherwise.
What am I missing?
Thanks in advance,
Mike.
--
_
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Well, it SEEMS to be working now. I don't know what I did, and frankly,
don't have time to back track to find out.
Thanks for your time.
Mike.
On Thu, May 24, 2018 at 4:33 AM, Doug Lytle wrote:
> On 05/23/2018 05:23 PM, Mike Diehl wrote:
>
>
> However, my user isn&#x
g wrong? Or is this approach simply doomed?
Any thoughts would be welcome.
--
Mike Diehl
--
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Check out the new Asterisk community
work correctly.
Any ideas where to look to fix this?
Thanks in advance.
--
Mike Diehl
--
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Check out the new Asterisk community forum at: https://communi
My comments below:
On Wednesday, March 20, 2019 12:19:08 AM Antony Stone wrote:
> On Tuesday 19 March 2019 at 21:36:53, Mike Diehl wrote:
> > Hi all,
> >
> > I have a user who is reporting one-way audio, but only when a call is made
> > to or from particular PS
completely?
Anyway, my user tested later that day and they are still having problems
Any other ideas?
Mike.
On Friday, March 22, 2019 08:32:39 AM Stefan Viljoen wrote:
> Hi Mike
>
> In rtp.conf, what are the port ranges you specify?
>
> I had almost exactly the same problem
d?
Thanks
--
Mike Diehl
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Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
On Thursday, May 16, 2019 05:12:17 PM Joshua C. Colp wrote:
> On Thu, May 16, 2019, at 5:00 PM, Mike Diehl wrote:
> > Hi all,
> >
> >
> > I've got a program that connects via AMI and acts upon the voicemail
> > message waiting event.
> >
> >
&g
as to where I should start looking?
Thanks in advance,
--
Mike Diehl
--
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Check out the new Asterisk community forum at: https://community.asterisk.o
s IAX instead of SIP?
Steve
In the U.S. you might look at
http://www.addaline.com
A sister company of the company I work for. I've successfully tested
Asterisk with them, and they give you your sip password.
--
Mike Holloway
Sr. Network Engineer
[EMAIL PROTECTED]
972.323.6598
http://www.
switching and power and provide a UPS so the phone system works
when the power goes out.
[Apologies, I'm new to this whole concept of IP phones and *.]
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises (812) 476-28
l device before it sends 48
volts down the wire. This will surely fry some non PoE ethernet
devices.
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises (812) 476-2881 fax
2626 Kotter Ave, Unit D [EMAIL PROTECTED]
Evansville, IN
is
direction or whether it makes sense.
For those of you who have done it, how much time did it take you
to get the system running smoothly?
PS: In case it matters, we're extremely Linux capable (we use it
for our file serving, networking, and we built our own custom ERP
on perl and mySQ
e grows over X
size, call other extensions, etc.
I am very intrigued by the flexibility Asterisk offers, but I
need to know that I can reliably just "make calls" at first.
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises
line up with the others, someone has
plugged in an RJ11 into it.
Sounds like a way to have flakey RJ45 jacks all over the place,
and ethernet does use pin 1!
This must be a FAQ somewhere...
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises (
Third question: Would you want it? Why?
Fourth question: How much $$$?
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises (812) 476-2881 fax
2626 Kotter Ave, Unit D [EMAIL PROTECTED]
Evansville, IN 47715
On Wed, 20 Aug 2003, John Todd wrote:
> At 3:20 PM -0500 8/20/03, Mike Ciholas wrote:
>
> >Are there VoIP dialtone providers? That is, could I use only
> >my internet connection for voice calls and not have a separate
> >T1/POTS bank for that?
>
> >First
On Wed, 20 Aug 2003, Brian West wrote:
> I think NuFone can do what you need contact [EMAIL PROTECTED]
>
> I have inbound 800 service and outbound ld service with them..
> works great.
And for local service, you do what?
--
Mike Ciholas(812) 476-2721 v
r *everything*
else? This assumes we are having only one emergency at a time!
Now, if that is possible, how does the VoIP dial tone provider
get my inbound local and toll calls? I would want my "local"
phone number to work, of course.
--
Mike Ciholas
he money I would
save on local CO lines I can buy a *lot* of toll free minutes!
Then the VoIP dial tone provider can route my toll free number to
me over the internet. Presumably, then, there is no real limit
on the number of "lines" coming in. It isn't hard coded like the
CO li
On Wed, 20 Aug 2003, Jeremy McNamara wrote:
> NuFone doesn't restrict any number of simultaneous channels and
> we do have a wholesale platform we ~can~ offer.
How do I find out more about this?
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS
n the US
that may want to expand to ~20 lines.
4. Interfacing so many POTS lines to Asterisk. I guess that
means an FXO channel bank to T1 card? Kind of stupid to go
digital/analog/digital in the last 100 feet.
Help?
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enter
als it?
This appeals to me given the cost and legal burdens placed on
local lines. But it won't work for everyone by any means.
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises (812) 476-2881 fax
2626 Kotter Ave, Unit D
analog to digital conversion. It
would essentially be the code found is so called "soft modems"
but taking it's input from packets rather than sampling a phone
line.
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises (812)
be gentle :)
Mike
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Is there any precompiled documentation?
Many thanks Jason, I'll look into it. And someone might want to put this info on Asterisk's webpage somewhere, just a thought
On Sun, 2003-08-24 at 14:29, Jason Ross wrote:
Mike,
On Sunday, August 24, 2003, at 05:56 pm, Mike Hollis wrote:
> Is there any precompiled documenta
is, will
there be issues of latency/bandwidth in handling the 64 kbps
streams?
Thanks for everyone's help.
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises (812) 476-2881 fax
2626 Kotter Ave, Unit D [EMAIL PRO
the previous question. But ISDN is actually *easier*
in some ways, no DSP on the samples to recover the modulation.
Thanks all for the help!
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises
ancel. This might be as
simple as "voice" versus "data" call (is that info provided by
the PSTN?). Is the echo cancel needed on voice ISDN calls? I
can live with no support for voice ISDN calls (can imagine why I
would ever get one).
--
Mike Ciholas
rent serial port hardware (USB serial dongle, PCI
multiport card, etc) for console. Switch serial cable to do BIOS
settings if need be.
What BIOS do you have that is serial configurable? I'm always on
the lookout for that.
--
Mike Ciholas(812) 476-
uot;. I'm likely to be using Cisco phones if that matters.
So, can * do this, and if so, how? Can MOH be selectively
enabled/disabled by extension?
Are there other ways to solve this problem besides QOH?
Thanks for everyone's help.
--
Mike Ciholas
uestion, is the 7960 worth so much more than the 7940?
Has only 4 more buttons that I can see. Anything else under the
hood that makes it worth that?
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises (812) 476-2881 fax
262
On 1 Sep 2003, Klaus-Peter Junghanns wrote:
> here is the URL for the netconsole patches:
> http://www.kernel.org/pub/linux/people/mingo/netconsole-patches
No work for me, instead:
http://people.redhat.com/mingo/netconsole-patches/
Is that what you meant?
--
Mike C
st
CVS and compile it". If you are in this line of work and really
know your way around * and the equipment listed above, please
send me a note with your areas of expertise, experience, and
rates.
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterpris
that certain NICs with certain phones
just do this and you have to find another brand/model NIC to make
this happy.
Disclaimer: everything in the email could be wrong.
--
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises (812) 476-2881 fax
2626
e
that? Is there more to it than digital to digital copy? Perhaps
echo canceling?
Can we also store sound files in ulaw? I know that takes more
space, but perhaps it is less CPU work to move the bits around
than to codec them.
--
Mike Ciholas(812) 476-2721 voice
C
On Tue, 9 Sep 2003, Eric Wieling wrote:
> It would have to do some kind of trascoding,
Forgive my ignorance, but why? PSTN is delivering 8 bit 8 KHz
ulaw samples. G711 is delivering 8 bit 8 KHz ulaw samples over
SIP. Aren't the two data streams identical down to the bit
level?
Has anyone successfully run asterisk with a VIA processor ?
I have tried unsucessfully, do I have to run make with any specific switches
?
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Does anyone sell a preinstalled asterisk server ?
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mysql
[Mar 11 21:30:25] WARNING[26144]: app_voicemail.c:2277 messagecount: Failed to
obtain database object for 'mysql'!
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'SIP/9826-ac087500' in macro 'stdexten'
== Spawn extension (macro-
Hi,
Asterisk 1.4.17
Sangoma a102DE
I'm having some issues sending CallerID Name to a Dialogic based phone app.
According to the pri debug (asterisk2a-pri-debug.txt in [3]) you can see
that it is sending the CallerID Name "Mike - Budgetone - reachme.com" to the
Dialogic car
Agreed, Callweaver and Freeswitch are both better for conferencing
(especially if you don't have zaptel hardware).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, March 12, 2008 1:28 PM
To: Asterisk Users Mailing List - Non-Co
I believe that is/was one of the goals of the phonecall project.
-Original Message-
Does it implement the ability to run more than 1 PBX in asterisk ? (Virtual
PBX)
To be clear:
more then 1 company using the same physical asterisk
___
-- Ba
Michiel van Baak wrote:
> On 15:32, Wed 12 Mar 08, Joshua Wilson wrote:
>
>> I don't believe it supports multi-tenant as of yet. It could be requested I
>> am sure.
>>
>
> I created a new VM and installed it.
> Guess what, no multi tenant support.
>
> Too bad all them good GUI tools never c
You'll need to post more info. Version and a scenario of what was happening
at the time would be a good start...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rilawich Ango
Sent: Wednesday, March 12, 2008 6:32 PM
To: Asterisk Users Mailing List - Non-Co
flash (maybe something in AJAX?)
>
> thanks
>
> ---
> Marek Cervenka
Shameless plug:
http://www.linuxjournal.com/article/9159
--
Mike Diehl
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asteris
Thanks for the help. I still had a misconfiguration in my res_odbc.conf, but I
figured it out and it appears my voicemail storage is working. I haven't had a
chance to get to the phone on the extension I'm using for it.
--
Mike Hammett
Intelligent Computing Solutions
http
qualify=yes
[8157879826]
type=friend
;accountcode=2
context=ics
secret=
username=9826
fromuser=8157879826
insecure=very
host=63.175.151.3 ;voip.essex1.com
fromdomain=63.175.151.3 ;voip.essex1.com
;canreinvite=no
;disallow=all
;allow=ulaw
--
Mike Hammett
Intelligent Comp
=8159092443
insecure=very
host=63.175.151.3 ;voip.essex1.com
fromdomain=63.175.151.3 ;voip.essex1.com
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent: Thursday, March 13
What do I have to do so the
outside world accepts emails from my Asterisk box? It is behind a NAT.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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line, or various programs
# that require network functionality will fail.
127.0.0.1 aiur.ics-il.net Aiurlocalhost.localdomain localhost
::1 localhost6.localdomain6 localhost6
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message
I am the ISP. ;-)
I'll have to look into that smarthost deal as there is no reverse DNS at
this time (my upstream's server times out).
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Erik Anderson" &l
Comments or alternative suggestions?
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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nice
too (tried that, didn't work -- only got the first subexpression).
;extract dialed number
exten => s,n,Set(dialed_num=$[ "${ARG1}" =~ "(.*)\\*" ])
;extract user specified callerid
exten => s,n,Set(callerid_num_custo
would
get confused if it didn't check the contact header for clairification since
a call is also coming from that source IP address when proxied through
openser.
Maybe I'm approaching this from the wrong direction, anyone have any ideas?
Mike
[privider1a]
type=peer
host=67.x.x.x
insecure=invite,
That's from asterisk-addons, you can ignore that error.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mark morreny
Sent: Tuesday, March 25, 2008 10:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Have problem w
the UK. Not sure on modern exchanges how long it would take for
the
line to clear.
Mike
Thanks,
> Steve
>
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> To UNSUBSCRIB
LID for this out-dialed call?
The SIPCALLID seems the incoming call's SIPCALLID.
Thanks.
Mike
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Mike
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Has anyone had any luck with Attrafax? I'm looking to use it as the T.38
gateway (PRI in, T.38 out).
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or
phone? Or, is it totally locked to your
network provider?
Any possible way of hacking it to work as some kind of voip client to work
on one's own implementation of UMA, if such
a thing even exists? :)
Thanks
Mike
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having similar problems?
Thanks,
Mike Clark
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jbenable=yes
jbforce=no
jbimpl=fixed
jbmaxsize=300
Is there something I haven't tried that will make this work or will I have
to change my dialplan so it doesn't use local channels?
Thanks,
Mike
PS, here are some pages that I have used as sources of information:
No mention of /j
is or point me in the right direction? I'm now not sure what I'm
missing or where to go looking for it?
Thanks guys and gals.
Mike
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e... I am waiting until out of
hours tonight >6pm GMT to test to see if these versions on libpri,
zaptel and asterisk fix the issues; and I will update the list to
reflect either my success or failure :/
Thanks guys
Mike
On 5/4/08, Mike Hardman <[EMAIL PROTECTED]> wrote:
> Ok Guys, I
y obvious now looking back :-/
I'm happy to say that the guys from redfone were incredibly helpful
every step of the way, without Jose explanations and tips I would
probably still be scratching my head...
Hope this helps another poor soul in my situation out in the future.
Mike
On Wed, May
Anyone have recommendations for wireless headsets that work well with
Polycom phones and Asterisk?
Thanks,
Mike Clark
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ata regarding which mobile phone the call
was originally intended for...
Is this a pure pipe dream? does PRI carry call diversion information?
Thanks
Mike
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asterisk-users mailin
I'll get right on hunting about RDNIS, thank you VERY much! :)
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*bump*
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 20, 2007 12:27 PM
Subject: [asterisk-users] e911
One of my providers has a different SIP
Then I could just make "downstream-phones" my current outbound context and
everything would do what I'm after. I got what you're saying.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Dave Mille
dtime/localtime.c -> stdtime/localtime.o
stdtime/localtime.c: In function `localsub':
stdtime/localtime.c:1136: error: structure has no member named `tm_gmtoff'
gmake[1]: *** [stdtime/localtime.o] Error 1
gmake: *** [main] Error 2
Thanks,
Mike Clark
_
No. All lines/extensions are registered to the base phone and the
handsets access the lines via the base unit. You can have multiple
simultaneous calls.
Jeremy Mann wrote:
>
> Are the cordless phones on the 480i CT from Aastra registered
> independently in Asterisk? Such that if I have 5 of the
cter, '_', to make wildcard
matching work. So your extension should be _*7XXX
Mike Clark
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on problem or issue that I am missing? I've tried Google, but have
had no success.
Thanks,
Mike
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Mike Clark wrote:
> I have installed Asterisk 1.4.15 on Solaris and got it all running
> seemingly fine. However, when I record a message or voicemail, it will
> not recognize the '#' key to stop recording. Hanging up is the only way
> to end the recording. DTMF seems to w
Hi,
just wondered if it was the same firmware on both devices?
thanks
Mike
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00-001 Rev: A
BootBlock 2.7.0 (12500_001)
BootRom: 4.0.0.0423
SIP: v.2.2.0.0047
PolyDSP Titan Mem1 FS3 v1.7.0.0057
Here is my SIP config:
[2000]
type=friend
username=2000
password=sip-access
dtmfmode=rfc2833
[EMAIL PROTECTED]
disallow=all
allow=ulaw
Any h
Doug,
That bug ID was a dead ringer. The workarounds in the bug worked
perfectly. BTW I'm on a openSuSE 10.3 system with gcc 2.4.1.
Thanks for the pointer.
Mike
On Jan 9, 2008, at 8:30 PM, Doug Lytle wrote:
> Mike Coakley wrote:
>> I'm setting up a new Asterisk system o
-w $LTMP && rm $LTMP
test -w $RTMP && rm $RTMP
test -w $OUT && rm $OUT
#remove input files if successfull
test -r $OUT.mp3 && rm $LEFT $RIGHT
# eof
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
__
Does what I have in the dialplan look right or am I way off base with being
able to use that script?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Steve Johnson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED
licit declaration of function âmemcpyâ
combine_wave.c:991: warning: incompatible implicit declaration of built-in
function âmemcpyâ
make: *** [combine_wave.o] Error 1
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Steve Johnso
000 +0200
--
File to patch:
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Patrick" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, January 15, 200
Never mind, I got it. I needed a -p0
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Patrick" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, January 15,
ed a procedure for upgrading the firmware, however I have not
had chance to do it yet.
Mike
> ___
>
> http://www.venturevoip.com (Great new VoIP end to end solution)
> http://www.venturevoip.com/news.php (Daily Asterisk News - html)
> http:/
reacts the same way as the AstDB configured lines.
Thanks,
Mike
On Jan 25, 2008, at 1:55 PM, BJ Weschke wrote:
> Mike Coakley wrote:
>> I'm trying to use the FollowMe app with Asterisk 1.4.17. I've
>> followed
>> the WIKI page on setting it up but I can't s
=> FM3/2000,30
Here is the relevant section of the macro that calls the FollowMe app:
exten => s,7,GotoIf("${DIALSTATUS}" = "NOANSWER"?:8:14)
exten => s,8,FollowMe(${STATION_EXTENSION},a)
I've tried different context in my FollowMe configuration file but
extensions table and builds
a context with the appropriate switch statement for each realtime
context. The output is included much like the text from an include
file. Of course, it still requires a reload if you add a brand new context.
regards,
Mike Clark
Yves Räber wrote:
> That
me, the difference was light night and day. I
was not using X100P cards though but the principal is the same for
the Sangoma A200 I was using.
Mike
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On 18/02/2008, srinivas Antarvedi <[EMAIL PROTECTED]> wrote:
>
> Hello all,
>
> I am struggling with sending voicemail as an attachement in Email.
>
> When i have given the email like [EMAIL PROTECTED] it is delivering
> to my gamil account perfectly(of course to spam folder).
>
> But when i given
.com (63.164.210.14)
Change setting to use "outbound Proxy"
--
Mike Hammett
Intelligent Computing Solutions
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Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Alex Balashov" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - No
ING[21149]: app_voicemail.c:1263 delete_file: Failed to
obtain database object for 'asterisk'!
[Feb 22 18:15:26] WARNING[21149]: app_voicemail.c:1400 store_file: Failed to
obtain database object for 'asterisk'!
[Feb 22 18:15:26] WARNING[21149]: app_voicemail.c:1122 retrieve_file:
It was my understanding that voicemail.conf referenced MySQL and not
asterisk.
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Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Tilghman Lesher" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-
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