gger is. Out of the blue, outbound calls start to work.
I had been using asterisk-1.6-beta9 with zaptel without any
problems.
Thanks,
Jim
-- Executing [EMAIL PROTECTED]:1] Macro("SIP/111-b4e05610",
"dialout-dahdi,18005551212") in new stack
-- Executing [E
Even that isn't always true.
Sometimes dial out on DAHDI works, sometimes it doesn't.
I'm not sure what makes it start working, but once it does,
it appears to stay working.
Jim
Jim Duda wrote:
> I don't know how to explain this.
>
> After receiving 1 inbound call
I don't know how to explain this.
After receiving 1 inbound call on the DAHDI channel attached
to the PSTN, outbound calls to the PSTN start working with
getting the "unable to create channel if type DAHDI" message.
If I restart *, the problem returns until I get 1 inbound call
g [EMAIL PROTECTED]:1] Set("SIP/107-b4e703f0",
"CALLERID(number)=781-736-1994") in new stack
-- Executing [EMAIL PROTECTED]:2] Set("SIP/107-b4e703f0",
"CALLERID(name)=Jim Duda") in new stack
-- Executing [EMAIL PROTECTED]:3] Set("SIP/107-b4e703f0&
I know about those packages. Questions is how do we use those packages
to build our own RPM. We use asterisk SVN trunk.
Thanks
Jim
On Mon, Sep 29, 2008 at 4:07 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Mon, Sep 29, 2008 at 03:51:35PM +0530, Jim Boykin wrote:
>> We
We use RHEL5, FC6, & CentOS5. I will be happy to hear your inputs for
any distribution you know.
On Mon, Sep 29, 2008 at 3:41 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Mon, Sep 29, 2008 at 03:34:45PM +0530, Jim Boykin wrote:
>> Thanks Alan, I will try it out. Seems li
Thanks Alan, I will try it out. Seems like a solution. Your assumption
is right, all system are same (ghosted).
I am also looking at pre-build RPM and reusing their specs file.
Anyone have input for building asterisk RPM.
Thanks
Jim
On Mon, Sep 29, 2008 at 3:07 PM, Tzafrir Cohen <[EM
Is there a script to create an Asterisk binary package after it is
compiled on one system.
We do not want to compile Asterisk of each system where we want to
run. I am sure there is a way but I could find it.
Thanks
Jim
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Thanks Gordon & Mike for the response.
What accuracy are you getting from zaptest/dahdi_test (and system info).
Two more questions:
1) Does ztdummy requires change into kernel? I am running 2.6.9 kernel.
2) What about CPU load?
Thanks
Jim
On Mon, Sep 29, 2008 at 5:02 AM, Mike Trest <
We plan to use asterisk for conferencing. As I understand, it requires
either a separate hardware like x100p clone or ztdummy. What are the
pro & cons of x100p vs ztdummy. Any other hardware suggestions for
conferencing? It should be able to handle few simultaneous
conferences.
Thanks
Thanks guys for inputs...not allowing multiple call is not an option -
essentional thats the problem we try to solve :)
Since we have our own CDR module, we can avoid external process. What
are the evens to listen for?
Other ideas will also be appreciated.
On Thu, Sep 18, 2008 at 8:23 PM, Alex B
.
Jim
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ports both
g729/ilbc, I will assume that asterisk will select ilbc but that does
not seems to be the case.
Jim
On 8/23/08, Brent Davidson <[EMAIL PROTECTED]> wrote:
>
> Steve Totaro wrote:
> On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin <[EMAIL PROTECTED]> wrote:
> We ru
Requesting help.
Thaks
On Tue, Aug 19, 2008 at 4:40 PM, Jim Boykin <[EMAIL PROTECTED]> wrote:
> We run asterisk to handle incoming DIDs and we have observed
> inefficient Codec Translation.
>
> Here is the scenario
>
> [DID Vendor] --
=yes
disallow=all
allow=ilbc
Thanks
Jim
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ccess to
the CallerID info on the other call? I assume there is some
extensions.conf magic required? An old CallerID box can do this, so I
would expect Asterisk to be able to do this too.
Any advice appreciated.
Best,
Jim
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Cool, thanks for the tip.
Why do you need to separate incoming and outgoing?
Jim
C F wrote:
> This is what I do:
> /etc/asterisk/features.conf
>
> [applicationmap]
> inflash => *4,caller,Flash,()
>
> outflash => *3,callee,Flash,()
>
> in /etc/asterisk/ext
a Flash( ) Dialplan function. How can I use this
Function in the Dialplan with a call which is currently in progress?
Any advice is most appreciated.
Jim
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x27;ve tried lower values for mwilevel too, nothing seems to work.
Is there additional debugging I can turn on in the chan_zap.c source code?
Thanks,
Jim
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fy.sh
faxdetect=incoming
signalling=fxs_ks
context=incoming
channel => 4
Does anyone have this feature working?
Do you see anything wrong with my configuration?
Thanks,
Jim
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I take back that statement about the h extension.
This works fine.
[sphinx]
exten => s,1,AGI(MisterHouse.agi,"Sphinx Connect")
exten => s,2,Dial(CONSOLE/1)
exten => h,1,AGI(MisterHouse.agi,"Sphinx Disconnect")
exten => h,2,Hangup
Thanks for the help!
Jim
Thanks for the responses!
I tried using both the "h" priority and/or the "g" option to Dial.
Neither of these worked for me.
However, the "n" priority works great. I didn't know about this "n"
priority.
Thanks,
Jim
Vincent wrote:
> On Sun, 9
ing asterisk-1.6.beta5
I never get to "3" below. I get a message saying the "2" ended with a
non-zero status.
[sphinx]
exten => s,1,AGI(MisterHouse.agi,"Sphinx Connect")
exten => s,2,Dial(CONSOLE/1,,e)
exten => s,3,
I have a Hitachi WirelessIP5000 phone (the original 1st gen hardware). It's
running the latest firmware (v2.2.6) and my Asterisk server is running
1.2.10. This setup has been working great for me for a long time, and then
last week I started having a problem where the Hitachi phone loses
registra
27;ve received a couple dozen test faxes without any crashes.
I realize it doesn't help your 1.2 needs, but I can confirm your
frustration.
Jim
Edwin Lam wrote:
> hi
>
> does any body know which version combination of
> spandsp/tx_fax/rx_fax will work with * 1.2.24?
>
> i tr
s 184
== Using UDPTL CoS mark 5
-- Called 107
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called 111
-- SIP/100-0827d8d0 is ringing
-- SIP/101-0828f808 is ringing
-- SIP/102-082a1ba0 is ringi
Is there anyway to get debugging into the log files?
Am I missing something?
Jim
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/ Asterisk attention...
I have multiple Asterisk servers in place and would REALLY be interested in
your tool set. I can test it on Leopard or Tiger as I have both in available.
Thanks,
Jim
- "Devraj Mukherjee" <[EMAIL PROTECTED]> wrote:
> Hi everyone,
>
> I have b
Thanks Russell, that's what I'm looking for.
Any idea when this will become part an official asterisk release?
Jim
Russell Bryant wrote:
> Brian J. Murrell wrote:
>> On Thu, 2007-12-20 at 16:29 -0500, Jim Duda wrote:
>>> Is there any means by which I can get
t the Message
Waiting Indicator (MWI) from the telco? Is there some application or
variable which can be used to identify an active MWI from the Telco?
Thanks,
Jim
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This is both a hardware and software licensing issue.
Avaya offers a SIP server separate from their main VoIP gateway.
The core platform uses H.323.
Either SIP or H.323 has a license cost per registered device.
We have an Avaya S8300 Communications Manager providing H.323 and have this
tied to an A
number match is met, regardless if the number match is a 4
digit extension or 7 digit phone number.
Is this one of the reasons and purposes Asterisk has a "real-time" option?
Thanks,
Jim
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link looks very nice.
Jim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Administrator
TOOTAI
Sent: Tuesday, November 06, 2007 4:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mystery phone!
Kyle Sexton a
I would like to send the CDR records from all our machines around the
world to a single database. But I need the hostname included with each
record for monitoring purposes.
Is there a better way than using the userfield and adding
SetCDRUserfield for every call to set the userfield to the name of
configure.
I would like to hear what people say about Snom as their sets look very
nice.
Sorry for the novel, all I really wanted to express is Grandstream is cheap,
look before you jump.
Good luck on your decision...
Jim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
pricing. If you try Snom please share your thoughts. At present we
are sticking with Aastra due to good results and user feedback.
Jim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson
Sent: Wednesday, October 31, 2007 11:06 AM
To: Asterisk Users
Balu Raman wrote:
> Hi,
> I want asterisk to call a person on the phone for monitoring the
> refrigerator storing vaccines.
> I am clueless where to look. Can someone clue me in ?
OWFS for sure! Here is a screenshot of a program I created a couple
years ago to monitor refrigerators/warmers. It w
Joseph Begumisa wrote:
I had the same problem with 45 polycom 601 phones in the same page
group. It was just like you describe it and I got the same answer from
polycom. What I did to go around that was add a second line key with a
different extension number on each phone
Steve Totaro wrote:
>
>
> If your provider will be providing the SIP trunks then it might be OK.
> Otherwise I would stick with PRI.
>
> My lessons learned are that you cannot control the public internet and
> traffic shaping and QoS are not useful most of the time. I would only
> consider a po
Jeremy Mann wrote:
> Without knowing more, Why fix what isn't broken?
>
I should have stated, the PRI is on an existing PBX not asterisk. My
goal was to reuse the existing PBX PRI card to interface with asterisk.
> I've been considering replacing a PRI with SIP or IAX trunks. The monthly
> co
learned" from those who are doing it or decided, for whatever reason,
it's a bad idea.
begin:vcard
fn:Jim Canfield
n:Canfield;Jim
org:Tulsa Spine and Specialty Hospital;IT
adr:;;6901 S. Olympia Ave;Tulsa;OK;74132;US
email;internet:[EMAIL PROTECTED]
title:Network Administrator
tel
Mojo with Horan & Company, LLC wrote:
> Have you tried adding an 'h' extension in addition? If the caller hangs
> up in the middle of priority 1 of extension 123, it should then jump to
> priority 1 of extension h and continue.
>
Thanks,
That works perfectly.
__
Greetings,
I have a dialplan that calls the dictate application, but I want to do
some post-processing on the RAW file created. The post processing is
working fine as long as the dictation application exits gracefully, but
fails when the user simply hangs up.
How can I make sure the system()
Greetings,
I know the hardcore guys will laugh, but I put together a quick .nanorc
config for asterisk. I tried to include all the applications listed on
the latest install. Please feel free to send any suggestions/updates my
why. I think this will go a long way to helping out the "new guys"
Eric B. wrote:
site and got to chapter 4 or 5 and decided to take a break. Which is when I
found AsteriskNow and TriBox and then started wondering if it was really
necessary / worthwhile to figure out all the intricacies of the application
if someones have already created the appliance ver
Thanks All. However, I was to use it with call files. Will GROUP work there?
~Jim
On 9/19/07, Alex Balashov <[EMAIL PROTECTED]> wrote:
>
> Try:
>
> http://www.voip-info.org/wiki/view/Asterisk+sip+incominglimit
>
> On Wed, 19 Sep 2007, Jim Boykin wrote:
>
> > Is
Can someone suggests a good and resonable cost voip provider with
business unlimited plan in USA and allows simultaneous outgoing
calling.
Thanks
~Jim
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--Bandwidth and
Thanks Matt, just minimal volume to suite comfort noise.
Jim
On 9/19/07, Matt Riddell <[EMAIL PROTECTED]> wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Jim Boykin wrote:
> > Where do I get sound file for comfort noise. GSM or MP3 is fine.
>
> What kin
Is there a way to limit simultaneous calls. I like to limit
simultaneous outgoing calls as more than few simulataneous calls are
charged by my voip providers. However, I do not want to have any such
restriction for internal calls.
Thanks
Jim
___
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Where do I get sound file for comfort noice. GSM or MP3 is fine.
Many thanks.
Jim
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James FitzGibbon wrote:
On 9/17/07, Jim Canfield <[EMAIL PROTECTED]> wrote:
stuff" useless. My real concern was the immediate '/ignore' for asking
about an issue with the *now ditro that actually had nothing to do
with
the GUI itself. Trut
SIP wrote:
> Not at all relevant to your query, but I still use the mysql CLI for any
> mysql task... and while most OSs have nice, functional tools to add
> users (command-line tools), there are SOME (*cough* Irix *cough*) where
> there are no CLI tools and VI is your only option (especially if
Greetings,
Last week I began researching Asterisk for the first time. I did what most
noobs would do; downloaded an image that seemed simple and straightforward
and had some credibility (*now). I also downloaded the TFOT version 1 as
a guide.
As questions arose, I tossed a few
Hi,
Is it possible to connect Asterisk to MSN messnger. I could find information
on Gtalk but nothing on MSN or Yahoo. Any help will be appreciated.
~Jim
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I tried several and had very poor luck with each I tried. These included
IaxComm, IaxComm Pro, Diax and Firefly II. Also, One other one from I
think Germany that had just changed it's name. All of these had issues. I
could not get Firefly configured at all to talk to Asterisk. Diax, when
ety of
other permission issues, which I'll work through. Thanks again for your
help.
Best,
Jim
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--On Sunday, July 22, 2007 9:03 PM +0300 Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:
> Oops:
>
>http://yourhost/cgi-bin/asterisk/vmail.cgi
Thanks Tzafrir!
That got the script to work. When I try to log in though, I get an odd
error:
Bleh, no /etc/asterisk/voicemail.conf at
/usr/lib/cgi-bin
--On Sunday, July 22, 2007 8:35 PM +0300 Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:
> We're talking about
> http://packages.debian.org/stable/comm/asterisk-web-vmail
>
> ( http://packages.debian.org/asterisk-web-vmail )
>
> It actually requires httpd-cgi. Apache happens to be one of the packages
>
--On Sunday, July 22, 2007 1:17 PM -0400 dave cantera
<[EMAIL PROTECTED]> wrote:
> the asterisk gui doesn't interact with apache or apache2... it has it's
> own httpd... perhaps you can move the vmail.cgi script to the apache2
> directory structure cgi-bin. I haven't tried that as of yet so I d
Hi Everyone...
I am running Asterisk 1.2.13 on Debian "Etch". I installed it from the
package. I also installed the web voice mail package, which installed
Apache2 and a bunch of other stuff.
When I point my browser at my PBX machine, the web page says "It Works!"
but of course it does not.
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] zaptel on CENTOS servercd
Jim Suber wrote:
> I use CentOS4.4. successfully. I do something which is very odd for a
Linux
> admin. I do a "install everything". There is/was a reason for this.
> I was in a hurry to get
Anyone know how to change the From field in Asterisk PBX voicemail/email to
some other info of my choosing?
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I use CentOS4.4. successfully. I do something which is very odd for a Linux
admin. I do a "install everything". There is/was a reason for this.
I was in a hurry to get a system online and didn't have time for a research
project. I wrote a simple shell script to compile the apps (zaptel, libpri,
ast
: Tuesday, May 22, 2007 9:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Phones fail to ring
Jim Suber wrote:
> I am somewhat confused. I have the incoming (s) context playing a
> greeting and callers choose one of two extensions (100, or 101)
>
I'm fairly certain that zaptel is not a service.
You might try "service asterisk stop"
I 'm running CentOS 4.4 as well. The only problem I'm having is sometimes
the extensions don't ring even though the caller hears a ring tone.
I'm thinking maybe it's the fact that I got 5 people on one little
DSL
or 101
exten => s,1,Zapateller(nocallerid)
exten => s,2,Answer()
exten => s,3,Background(listext)
exten => i,1,PlayBack(pbx-invalid)
exten => i,2,Goto(incoming,s,1)
exten => t,1,PlayBack(vm-goodbye)
exten => t,2,Hangup()
T
.conf and setup blind and attended transfers for
asterisk.
It just works better in my opinion.
-bk
- Original Message -
From: "Jim Suber" <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Thursday, May 3, 2007 9:53:58 AM (GMT-0800) America/Tijuana
Sub
> headset.
>
> Regards,
>
> Mike
We use the Plantronics SupraPlus SL #P361N-U10P ($102 USD)
Binaural, noise canceling, soft leather, etc.
(Does not require a separate amplifier.)
A bit on the spendy side, but for those who live on the phone,
it makes being in a call center, sal
of 1 number before
Alice announces
That there is no such extension.
HELP
Thanks in advance
Jim
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ld ma bell phone (when I want it to :-) )
Is this possible in Asterisk?
Thanks,
Jim
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, the phones that fail have CDP enabled,
while the phones that succeed have CDP disabled. I think this is
Continual Data Protection,
but don't see where to disable it on the phone interface. Is this a
cause of the failure?
Any insight will be greatly appreciated.
Thanks
--
Jim F
service? The absolute cheapest, regardless of (known)
quality
- the quality only has to compete with (cheaper) automated
transcription, which is abysmal quality.
You may want to try Amazons mechanical turk.
Jim
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I've seen an issue like this from time to time on 601s, even with the
latest firmware. Not just the softkeys, but also the dial keys. The
phones seem to "run slow" sometimes, failing to respond to a key
press right away but getting to it eventually. It usually clears up
after a few sec
On Apr 9, 2007, at 9:29 AM, William Moore wrote:
On 4/9/07, Jim Freeze <[EMAIL PROTECTED]> wrote:
> Or a new Digium TDM880B replacing the old TDM40B for only one
IRQ...
Do you know if this board will fit in a 2U machine?
The TDM800P is about the same height as the TDM400P and is
On Apr 9, 2007, at 8:32 AM, <[EMAIL PROTECTED]> [EMAIL PROTECTED]> wrote:
Or a new Digium TDM880B replacing the old TDM40B for only one IRQ...
Do you know if this board will fit in a 2U machine?
Thanks
Jim
Best Regards,
Francois BERGERET,
France.
-Message d'origine---
tly appreciated.
Thanks
Jim
--
Jim Freeze
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and played the WAV
files directly, and I hear the same problem. So, I'm guessing the
problem is introduced when the voicemail is recorded.
Can anyone offer a suggestion as to what I can debug?
Thanks,
Jim
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The zttest program results in > 99%.
Jim
"Tzafrir Cohen" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> On Tue, Mar 27, 2007 at 09:30:36PM -0400, Jim Duda wrote:
>> Lacy,
>>
>> I don't have any zaptel cards installed. I do however h
15 mS.
Can you recommend a method to test jitter or packetization?
Jim
"Matt" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
And/or periods of large jitter on your network connection.
On 3/28/07, Matt <[EMAIL PROTECTED]> wrote:
Could it possibly be
ummy now to see if anything else is required, for
example,
udev configuration.
I use Fedora Core 5.
Jim
"Travis Schafer" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
Looks like output from the 'lsmod' command.
>>> "Lacy Moore - Aspen
this look normal?
Jim
Jim Duda wrote:
Lacy,
I'm using asterisk 1.4.2 and zaptel 1.4.1.
I read the READMEs again.
I believe I need to change my kernel RTC to 1000HZ.
Also, I didn't have enhanced_real_time clock enabled, as such,
ztdummy wasn't loading properly.
I have rebu
for the replies!!
Jim
Lacy Moore - Aspendora wrote:
On 3/27/07, Jim Duda <[EMAIL PROTECTED]> wrote:
I don't have any zaptel cards installed. I do however have ztdummy
installed.
Hmm... Not sure. But this really sounds like ztdummy is not working
correctly. Hopefully someone else c
Lacy,
I don't have any zaptel cards installed. I do however have ztdummy
installed.
Is there some tweaks to ztdummy which I might need?
Is there a special kernel setting which ztdummy requires?
Jim
Lacy Moore - Aspendora wrote:
On 3/27/07, Jim Duda <[EMAIL PROTECTED]> wrote:
explain what might cause this? It doesn't always happen, and
seem unpredictable.
Thanks,
Jim
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On Tue, 2007-02-27 at 19:14 +0200, Dovid B wrote:
> Doug is this for the sip version or firmware ? As far as I know once you go
> beyond a certain firmware version with polycom you cant go back.
>
> Dovid
We used bootrom version 2.6.1.
And yes, once you go to version 3.x, you cannot go back.
Fo
Yes, I had seen something in various posts about using SIP instead of IAX2. I
have been switching back and forth
between IAX2 and SIP, however, I haven't seen any noticeable difference. I
will try a switch back to SIP again and see
how that goes.
Jim
"James Fromm" <[EMAIL
Drop OOO Kpkts Jit Del
Lost % Drop OOO Kpkts
IAX2/teliax-2 37 -10-1 -1 0 -1 50 40
0 0 00 0
> Do you have the jitterbuffer on or off ?
I don't believe so. I didn't turn jitter on. I believe jitter is off by
defaul
and 500
Kbit upload speeds.
Jim
Lacy Moore wrote:
Jim Duda wrote:
I've been on the shorewall firewall and confirmed that I have the
firewall configured properly for VOIP QOS.
What exactly have you done here? You do know that you are apparently
using IAX2 and not SIP. Those are not the
ng the Wireshark ethernet
protocol analyzer. Everything looks okay best I can tell.
What else can I do to analyze why the voice quality is so bad?
What can I do in Asterisk to help track down where the problem is?
I want to make this VOIP work.
Thanks for any help.
Jim
ke exten => 1234,hint,(Zap/1&Zap/2), but I don't
understand the & part (does it mean any of?). Is there some similar
syntax that means all of? Or, is there a way to synthesize a dummy
extension hint out of the individual channel hints? Or is what i'm
asking not fe
dm2402b.htm
--
Jim Freeze
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On 1/25/07, Jay Moore <[EMAIL PROTECTED]> wrote:
Jim,
I have 2 TDM400s in my * box (as well as a T1 card). I use all 8 ports,
and aside from some minor echoing during peak periods, it's running
smooth as ice.
Hi Jay. Thanks for the info. Digium logged onto my box early on and
w and the remaining 2 later.
Are there success stories with using 2 TDM cards?
Any info will be appreciated.
Thanks
--
Jim Freeze
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Hi,
I had MAJOR problems with this. I ended up just putting the .so files into
the asterisk modules directory.
That worked for me. I can send you the files I used if it's any help to
you.
Regards,
Jim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Beha
Just as an "it works for me", I created a FWD account a couple of weeks
ago, which seems to be working fine. I am able to receive calls over
IAX2 via my IpKall number.
Jim
Timothy Parez wrote:
I have one account which was created 3 weeks ago and 1 that was
created 2 days ago, ne
What are these?
Nov 22 09:35:23 WARNING[7127]: channel.c:787 channel_find_locked:
Avoided deadlock for '0xf6c06778', 10 retries!
Nov 22 09:35:24 WARNING[7127]: channel.c:787 channel_find_locked:
Avoided deadlock for '0xf6c06778', 10 retries!
Nov 22 09:35:24 WARNING[7127]: channel.c:787 channel_fin
ems do
this on board and export a simple command set. But I also don't know
anything about Digium's hardware either.
Thanks again! I really appreciate the help!
Jim
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--On Sunday, November 12, 2006 10:06 PM -0500 Steve Totaro
<[EMAIL PROTECTED]> wrote:
add a couple or few w's before you dial.
Okay, but where? I didn't see a w option for the dial command, and if I
add a wait before the dial won;t that just delay going off hook?
_
so I am almost certain this is what's
happening.
Thanks!
Jim
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it working without the lifters.
So, can someone confirm why I need the lifters?
I have to agree that if the lifters are needed, then there
should be a phone that comes with this built in.
--
Jim Freeze
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