autodomain=yes
domain=172.16.16.6
allowexternaldomains=no
In addition, in the general endpoint template in sip.conf, I have the lines
contactdeny=0.0.0.0/0.0.0.0
contactpermit=172.16.16.0/255.255.255.0
host=dynamic
What else am I missing?
Thanks
\RR
...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *RR
*Sent:* Thursday, March 10, 2011 7:04 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] [1.8] Unable to Register: Registration denied
because of contact ACL
Hello All
On Thu, Mar 10, 2011 at 2:12 AM, Faisal Hanif fai...@vopium.com wrote:
You can add following line to your peers configuration
permit=0.0.0.0/0.0.0.0
It will allow to use that peer’s account from any IP
Thanks. But Like I said, that's all done. Here's the Endpoint config:
On Mon, Mar 7, 2011 at 2:11 AM, Stuart Longland redhat...@gentoo.orgwrote:
http://mirror.aarnet.edu.au/pub/gentoo/distfiles/asterisk-1.8.3.tar.gz
I haven't checked that URL, but it should be correct. That, and that
mirror should be unmetered if you're on a university network.
Thanks mate,
On Mon, Mar 7, 2011 at 5:25 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 03/07/2011 03:35 PM, RR wrote:
Hello all,
mmm a bit embarrassing about not having a clue as to why we're getting
this error on make of 1.8.3
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash
On Mon, Mar 7, 2011 at 5:34 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 03/07/2011 04:31 PM, RR wrote:
On Mon, Mar 7, 2011 at 5:25 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
Please do not reply directly to posters on the mailing list unless
that
doesn't have other related stuff unselected? no clue where to start looking
Thanks
\RR
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On Mon, Mar 7, 2011 at 5:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote:
Okay, so here's the configuration I have for MySQL Realtime (Asterisk
version 1.6.2.17):
In /etc/asterisk/extconfig.conf:
sipusers = mysql,mya2billing,cc_sip_buddies
In /etc/asterisk/res_mysql.conf:
Don't know
Hello Stuart
On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.orgwrote:
On 03/08/11 08:49, RR wrote:
Any idea where this is coming from? seems like something is selected
that doesn't have other related stuff unselected? no clue where to start
looking
No SPARC expert
On Mon, Mar 7, 2011 at 6:31 PM, Stuart Longland redhat...@gentoo.orgwrote:
On 03/08/11 09:21, RR wrote:
Hello Stuart
On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.org
mailto:redhat...@gentoo.org wrote:
Even if it doesn't help fix the problem, you probably
On Mon, Mar 7, 2011 at 5:45 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 03/07/2011 04:41 PM, RR wrote:
Someone with SPARC experience will have to chime in then... for some
reason the configure script has determined that your compiler
provides atomic instructions
On Mon, Mar 7, 2011 at 6:48 PM, RR ranjt...@gmail.com wrote:
On Mon, Mar 7, 2011 at 6:31 PM, Stuart Longland redhat...@gentoo.orgwrote:
On 03/08/11 09:21, RR wrote:
Hello Stuart
On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.org
mailto:redhat...@gentoo.org wrote
Hello All,
wondering if anyone knows of any reliable mirrors to download asterisk from
in Australia or somewhere close to it than having to download stuff all the
way from the US?
Cheers,
\R
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. I just
wanted to know if someone's used it and and what their experience has been
in both, TTS and ASR.
Thanks
\RR
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On Tue, Jan 25, 2011 at 6:59 AM, Andrew Latham lath...@gmail.com wrote:
Thanks Dave. Sounds like a man who's not had his hand soaking in ivory
liquid and been through the toils and tortures of various upgrades over
the
years. Very insightful though. Goof thing this discussion ensued as I
On Mon, Jan 24, 2011 at 3:11 AM, Stelios Koroneos
skoron...@digital-opsis.com wrote:
On Mon, 2011-01-24 at 01:09 -0500, RR wrote:
On Mon, Jan 24, 2011 at 12:24 AM, Paul Belanger
pabelan...@digium.com wrote:
On 11-01-23 10:24 PM, RR wrote:
email from Kevin Flemming
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West ro...@firedrake.orgwrote:
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
In the meantime, does anyone have a nice way to update a stable/stock
lenny
installation with the updated glibc as well as the latest kernel
At this point
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com wrote:
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West
ro...@firedrake.orgwrote:
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
In the meantime, does anyone have a nice way to update a stable/stock
lenny
installation
On Mon, Jan 24, 2011 at 12:09 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/24/2011 07:29 AM, RR wrote:
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com
mailto:ranjt...@gmail.com wrote:
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West
ro...@firedrake.org mailto:ro
On Mon, Jan 24, 2011 at 7:07 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/24/2011 12:46 PM, RR wrote:
On Mon, Jan 24, 2011 at 12:09 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
On 01/24/2011 07:29 AM, RR wrote:
On Mon, Jan 24, 2011 at 4
On Mon, Jan 24, 2011 at 8:57 PM, Dave Platt dpl...@radagast.org wrote:
I know this is an {*} list but does anyone know if simply adding the
Squeeze
repository to my sources.lst and running an 'aptitude
upgrade/safe-upgrade/full-upgrade will just upgrade Lenny - Squeeze
without me having
the linux kernel (currently at 2.6.37) from scratch
and get access to the TimerFD source? Should I even bother with it for
app_confBridge or does pthread work well enough?
Thanks
\RR
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On Sun, Jan 23, 2011 at 10:16 PM, Paul Belanger pabelan...@digium.comwrote:
On 11-01-23 10:01 PM, RR wrote:
I'm sure this has been talked about and based on some searching of
archives,
I'd discovered that to be able to use timerfd, one needs to have a kernel
version =2.6.27? Is this true
On Mon, Jan 24, 2011 at 12:24 AM, Paul Belanger pabelan...@digium.comwrote:
On 11-01-23 10:24 PM, RR wrote:
email from Kevin Flemming talking about =2.6.27 so thought I'd ask esp.
coz
I have 2.6.26-2 yet I don't think I have timerfd on my machine...and I
see,
the following
If you read
On Mon, Jan 24, 2011 at 12:24 AM, Paul Belanger pabelan...@digium.comwrote:
On 11-01-23 10:24 PM, RR wrote:
email from Kevin Flemming talking about =2.6.27 so thought I'd ask esp.
coz
I have 2.6.26-2 yet I don't think I have timerfd on my machine...and I
see,
the following
If you read
itself but then I got distracted with something else and now it's closed and
I'll have to repoen it to add any more notes. Just FYI.
\RR
On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher tles...@digium.com wrote:
On Wednesday 08 December 2010 14:21:57 RR wrote:
Hi Guys,
Any one want to take a stab
BTW, the issue was created yesterday, but I didn't think there was a need to
post it here but nevertheless for posterity, the Issue ID is: 18442
Thanks
\RR
On Wed, Dec 8, 2010 at 6:57 PM, RR ranjt...@gmail.com wrote:
On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher tles...@digium.comwrote
On Thu, Dec 9, 2010 at 10:02 AM, Bruce McAlister
bruce.mcalis...@blueface.ie wrote:
Hi RR,
I’ve not tried compiling 1.8.1-rc1 on Solaris yet and I’ve not come across
this issue as of yet. I did build 1.8.0-rc5 on Solaris 10 without any build
error’s though. I’m not sure if the code has
in OpenSolaris resides in /usr/include/net as
opposed to maybe /usr/include/linux.
Any ideas?
Thanks
RR
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...this is the excerpt
from the netsock.c file:
*#if defined (SOLARIS)
#include sys/sockio.h
#elif defined(HAVE_GETIFADDRS)
#include ifaddrs.h
#endif
*
I would've have thought this would have taken care of the issue by making
sure 'make' handles this correctly but I guess not. Anyone? Please?
Thanks
\RR
On Wed, Dec 8, 2010 at 3:40 PM, Paul Belanger pabelan...@digium.com wrote:
On 10-12-08 03:21 PM, RR wrote:
Any one want to take a stab at helping with this please?? All I have
found
so far is that the netsock.c file has code that references to taking note
when it's being built
On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher tles...@digium.com wrote:
On Wednesday 08 December 2010 14:21:57 RR wrote:
Hi Guys,
Any one want to take a stab at helping with this please?? All I have
found so far is that the netsock.c file has code that references to
taking note when
Hi Bruce,
Thanks again for your generous response, please see a few comments inline
On Sat, Dec 4, 2010 at 6:27 AM, Bruce McAlister bruce.mcalis...@blueface.ie
wrote:
Hi RR,
Replies inline below
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun
and is it better to stick with 1.6x or just go to 1.8 as there's no
upgrades or backward compatability requirements for me.
Once I get this going, I promise to have an updated document uploaded
somewhere or will mail to the list so someone can put it on the wiki.
Thanks,
RR
On Thu, Dec 2, 2010 at 5:05 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
Assuming Solaris is anything like Linux, the installer will just be a shell
script. Open the script in a text editor and search for the text of the
error message. It will be wrapped inside an `if` statement, just
On Wed, Dec 1, 2010 at 3:58 PM, RR ranjt...@gmail.com wrote:
Zaptel package isn't installing though ...crashes midway complaining that:
*Operating environment requirement not met.
This package requires Solaris 7 or better.
checkinstall script suspends*
huh? I'm running 5.11, which
is just to document how to compile
Asterisk/Zaptel under Solaris 10/11 so when I do get a real Solaris machine,
I have already sorted out all the issues with installing/compiling etc
Thanks
\R
On Wed, Dec 1, 2010 at 1:10 PM, Bruce McAlister bruce.mcalis...@blueface.ie
wrote:
Hi RR,
As far
anything from anyone. This is EXTREMELY
critical for me to work...can anyone kind generous gentleman please help?
Thank you so much
\RR
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on Solaris 10 and a cluster/farm of TTS
servers for TTS processing.
Thanks
RR
On Thu, Nov 11, 2010 at 10:21 PM, Luis Morales faston...@gmail.com wrote:
You try install debian in your sparc platform ?
On Thu, Nov 11, 2010 at 8:52 PM, RR ranjt...@gmail.com wrote:
Hello Group,
I have been going
configuration you have. I agree with you. I will use TTS in its own
native environment and have Asterisk talk to it using UniMRCP or something
but I need a lot of help.
Any help will be appreciated. Thanks
\RR
On Thu, Nov 11, 2010 at 11:22 PM, Luis Morales faston...@gmail.com wrote:
I use Nuance, festival
Cepstral? How expensive is Loquendo?
Thanks
RR
On Fri, Nov 12, 2010 at 1:35 AM, Luis Morales faston...@gmail.com wrote:
Well,
I use many tts products because i work with diferents telphone
systems. Now for asterisk the best way for free is Festival and noon
free is Loquendo.
I'm not have
On 7/11/07, Guillermo Salas M. [EMAIL PROTECTED] wrote:
On Wed, 2007-07-11 at 08:32 -0400, Maximo Villamayor wrote:
you can prove this www.portsip.com
You can use the older version of firefly that supports IAX2/SIP
protocols and g729 codec.
Get the sofhophone and codec from:
On 7/8/07, Dovid B [EMAIL PROTECTED] wrote:
What does the NexTone run for ?
- Original Message -
From: Andy Brezinsky [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 03, 2007 8:17 PM
Subject: Re:
Hello,
is there anyway or any plan to have the date/time stamp that's printed
in an outgoing voicemail notification email to NOT be the date/time of
the (*) machine but infact correspond to the timezone set for the
subscriber under the TZ variable?
I have the (*) machine set to UTC and when the
On 1/18/07, JR Richardson [EMAIL PROTECTED] wrote:
I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3.
All seems to work normally with realtime voicemail, reads vmbox
parameters from the db fine. When I try to change the password,
asterisk operates normally, enter new
On 1/18/07, JR Richardson [EMAIL PROTECTED] wrote:
I use the same database for the sip, iax, exten and vm, different
tables. When a sip device registers, asterisk writes to the database
with updates to the sip table ipaddress, port and regseconds, so I
don't think there is a write permissions
On 1/17/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
All,
Just came across the prompt #3 from inside the top menu of VM in latest
stable. Allison does not announce the prompt, but if you know it is there,
you can press 3 successfully follow the prompts from there to send your
message to
On 1/18/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
some body told me that you can make asterisk to register itself to another
asterisk server. i just want to know whether it really can be done or not. i
have googled a lot but no answeres.
--
Regards
Rizwan Hisham
Software Engineer
Hello Gents,
following on this discussion, anyone particularly have one view or the
other about 1.4 and the voicemail and meetme enhancements (supposedly)
it has? We're not in production yet, I've tested 1.2 up until 1.2.13
in the lab as well as 1.4b3, since none of them got a real hammering
Its
On 12/15/06, Michael Hamann [EMAIL PROTECTED] wrote:
Hello,
we are using asterisk in combination with the voicemail system. I´m just
wondering if it is possible to switch the voicemail to an I am on
holiday mode.
This means that the unavailable message is played to the caller but no
On 12/14/06, Tobias Wolf [EMAIL PROTECTED] wrote:
Actually you don't need 2 different extension, but two different
parameter-sets for the meetme-App. So, you have to implement some logic
that detects, if the calling user has to be marked or not. It's your
choice if you do this by dialplan logic
On 12/14/06, Tobias Wolf [EMAIL PROTECTED] wrote:
Hmmm, there is really not much to share. Most of the code handles
Authentication or other stuff, like informing another server that a new
user has entered an conf-room, or updating databases.
Mostly I look an the CallerId to decide if this
On 12/13/06, Savoy, Kevin - Williston, ND [EMAIL PROTECTED] wrote:
I am trying to set up a Conference room where users are put on hold
until the host arrives. I have figured out that the A option activates
marked mode and the w option is used to activate the waiting until the
marked user
On 12/11/06, David Thomas [EMAIL PROTECTED] wrote:
I only say two options for voivemail staorge when compiling 1.4, IMAP
and ODBC. Are you using one of these? Are they configured?
Does anyone know if Version 1.4 still does filesystem based storage of
voicemail or if you must use IMAP or ODBC?
On 12/11/06, Martin Joseph [EMAIL PROTECTED] wrote:
Sometimes if there is a message in a format that voicemail doesn't
like, it crashes like that. Make sure the voicemail box is empty and
try again... I have seen it crash like that with audio data it didn't
like going back to before 1.2.
HI All,
Something weird has happened to my (*) setup.
Setup:
I'm using a Realtime-Driven (*) server for voicemail which has the
knowledge of all mailbox users on the softswitch which is remote to
this (*) box. Since that's all this box is used for, all I have in the
sip.conf is the definition
RR, mate, I don't think that I have so many problems.
1.) I asked a simple question:
Is it (still not) possible to connect Asterisk directly (= without ODBC)
to mySQL for the purpose of storing voicemail data?
Now, some posts later I've got a simple answer:
No!
Oh, haha sorry about that, I
On 11/29/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
You can have your own external script to do whatever you want when vm is
left
from voicemail.conf:
; If you need to have an external program, i.e. /usr/bin/myapp
; called when a voicemail is left, delivered, or your voicemailbox
; is
On 11/29/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
You could of course edit app_voicemail.c to pass more info...
Round about line 2329:
if (!ast_strlen_zero(externnotify)) {
if (messagecount(ext_context, newvoicemails,
oldvoicemails)) {
And as I wrote before, Asterisk - mySQl connection is already up and
runnig (for CDR). So it just would have been quick and easy if Asterisk
could have used the same path for audio data.
O.K., lets invest some time in installing ODBC.
NOrbert
Norbert, mate, I don't know why you're
Hello all,
does anyone have a clever way of creating a customised email that goes
out as result of the voicemail notification. And I don't mean Editing
what you want in the emailbody, emailsubject, serveremail etc
keywords. I mean custom in the sense that it has that info but the
email is
On 11/28/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
Is the storage of actual voicemail messages in a database still limited
to ODBC? If so, why?
And is the use of mySQL and ODBC at the same time still a bad idea? If
so, why?
I want to store all of my voicemail stuff in a database
Hello all,
just curious if anyone's successfully compiled (*) with the latest
FreeTDS code/driver. The Makefile in (*) seems to only take care of
0.63 or older. I tried to muck around with it a bit into tricking to
compile for not just 0.63 but anything later than 0.62 but it seems to
crap out
On 11/6/06, Mosiuoa Tsietsi [EMAIL PROTECTED] wrote:
Hi,
After some more searching I decided to try USING unix ODBC for the
connection. I have both the unixODBC and unixODBC-devel packages on my
fedora box:
[EMAIL PROTECTED] /]# rpm -qa | grep -i unixodbc
unixODBC-2.2.11-7.1
On 11/8/06, Steve Edwards [EMAIL PROTECTED] wrote:
All calls come in from a Tekelec 7000 via SIP.
Out of a peak of 200 calls, probably around 100 are in meetme, others are
listening to recorded messages or bouncing around in the menus.
Sounds exactly like what people in my system would be
On 11/8/06, Steve Edwards [EMAIL PROTECTED] wrote:
I have alaw, g729, gsm, ulaw, and wav sound file sets so that should
cover the transcoding bases pretty well.
It should but if you're not allowing anything but ulaw all of those
are probably not ever being used. What you might want to have is
On 11/2/06, Eddie Johnson Jr [EMAIL PROTECTED] wrote:
Hello Matthew,
Did you test Snom or Sipura hard ip phones? I was considering Budgetone for an
office of
10 users. After reading your testimonial I will have to re-think my selection.
FWIW, after having played with 3-4 BudgeTone phones
On 11/2/06, Steve Edwards [EMAIL PROTECTED] wrote:
I'm running CentOS 4.4, Asterisk 1.2.13 on HP DL380's. My application is
mostly meetme conferences being created and closed all day long. Peak load
is around 200 SIP calls.
I was crashing 7 to 10 times a day until I booted a non-SMP kernel. I
On 10/3/06, Jay R. Ashworth [EMAIL PROTECTED] wrote:
It seems unreasonably difficult to get a list of the supported formats,
but does sox (http://sox.sourceforge.net/) do what you need?
Cheers,
-- jra
hey Jay, thanks but I am not sure what to tell sox as my output format
to be. I must admit,
Hello All,
does anyone happen to know of a good utility or CLI tool to convert
prompts into a g.726 format? I tried using the convert utility in (*)
but it doens't like G.726. I understand I can just hunt around the net
for it, but if someone knows one off-hand that I can run on linux and
even
That's all well and good, but there are some phones out there that pack
samples into RTP payloads using the AAL2 direction. This causes interop
nightmares (i.e. your phones talk G.726-32, someone elses phones talk
G.726-32, but it sounds rubbish when you attempt a conversation). I
would guess
That's all well and good, but there are some phones out there that pack
samples into RTP payloads using the AAL2 direction. This causes interop
nightmares (i.e. your phones talk G.726-32, someone elses phones talk
G.726-32, but it sounds rubbish when you attempt a conversation). I
would guess
On 9/16/06, Rich Adamson [EMAIL PROTECTED] wrote:
RR wrote:
All,
is there anyone who uses g726-32 ? If not, then does anyone know why
don't people use it?
I use g726 on iax links between systems and to teliax.com for LD calls.
Have no idea if its -32 or what though. What ships with asterisk
have a look at Wiki for asterisk + odbc storage. The database for
storing entire voicemail messages can be stored on a local or a remote
database. Then you can do whatever you want with it. You will have to
recompile asterisk by turning on ODBC storage. It's all there on the
Wiki
All,
is there anyone who uses g726-32 ? If not, then does anyone know why
don't people use it?
It's free, and provides the best compromise on quality, bandwidth and
cpu load (judging by it's specs and algorithm) and oh did I mention,
it's FREE?
So why don't people use it? Any ideas? Is it too
You can achieve the active-passive setup using Linux HA techniques
using heartbeat and the like. You can also do load-balancing with LVS.
What I am not sure about is maintainance of call and session states
between the two servers such that when one server dies, the other
server picks its IP
Hi matt,
sorry this might be a stupid question but is a bit pertinent to me,
I'd asked something similar in one of my last email regarding SMP. Do
you know if (*) is capable of making use of HT support i.e is
multi-threaded and improves performance for operations like
transcoding? Is that a
Thanks Leo, great explanation. Will do some additional research and
try out a few tests if I can find the time to setup a small load-test
sort of a scenario but it does sound from your explanation that
symmetric multi-processing is what we need to share the load and get
double or close to double
Hi Matt,
The best use I have seen is the newly converted IAX2 which can use
multithreading in version 1.4, the beta of which should be released
later this week.
The best idea would be to compile Asterisk, run some tests (show
translation recalc 60) with HT turned on, restart the box, bring it
I am currently running this with UnixODBC - FreeTDS - MSSQL Server
2K ( please don't hate me for using an 'evil empire' product amongst
the pure sanctity of open source :D). But the results are, well...So
far so good. But I can't say much because the most i've tried is 4
concurrent connections to
HI Mojo,
thanks for that. Sounds like a hidden option. It doesn't show up when
I do a tab after I type show translation on the CLI.
But to respond to your comment, I thought that's what it was, as in
calculated based on the current load of the system but the fact is
that there is absolutely
Hi Leo,
Sorry mate, I thought I had done some research but I only found one
reference to it somewhere and it stated that if you have more than one
VM running inside VMWare server v1.0, then there are timing issues
where the clock seems to vary randomly. I figured that didn't apply to
me since I
Ben,
The family name is not sipuser, its sipusers. So try this command
realtime load sipusers name username and see if you get nothing. What about?
realtime load sipusers username username ?
To answer your question, any change in the tables holding this sip
users information comes into affect
Ben,
that's exactly how it is, the load command is only for you to see
what's being pulled from the database and to test if realtime has been
configured properly. If you see nothing, then I suspect realtime for
you isn't really working and the calls that are working are being
looked up in the
Assuming you have the tables as named int he extconfig.conf as well as
the database astDB, how about enabling the module app_realtime.so?
Also, if you're using mysql, I don't think you need res_odbc,
res_config_odbc. Instead try turning on app_realtime.so and
pbx_realtime.so and see how you go :)
I use rtcachefriends=yes and any changes I make in my database become
effective immediately along with also getting the MWI functionality.
Even though what you say makes sense. Go figure!
Ben, yeah if it shows it's loaded then it's there for sure. Sorry I
asked for it as in your module listing
Hi all, (2nd attempt)
this is probably a weird question and something I'm not doing right
but I got this bizarre thing going on here. When I boot the system
with the SMP kernel and compile (*) with the smp kernel source
(actually even if I don't compile, but as long as I boot into the SMP
Hi Zoa,
thanks for responding. Ok, now where do I find this? I'm running
2.6.9-34.0.1 kernel. I tried doing a bit of search and it seems like
that the ability to change the frequency doesn't appear till 2.6.13.
Am I looking at the right thing? Any hints?
Hi there,
sorry I wasn't sure exactly where to start so didn't know what info to
provide. Now that I know, here's the info
1) using a P4 w/HT
2) Using CentOS 4.3 with the 2.6.9-34.0.1-smp (Note, this was
installed through an rpm, but the (*) and zaptel code is being
compiled against the source
Hi all,
this is probably a weird question and something I'm not doing right
but I got this bizarre thing going on here. When I boot the system
with the SMP kernel and compile (*) with the smp kernel source
(actually even if I don't compile, but as long as I boot into the SMP
kernel), I get this
Also, keep in mind that from what I had understood, Vonage required
any endpoint/acct. to register with them every 30 secs (I'm assuming
they set the register expiry timer to 60) to ensure all endpoints keep
their firewall pinholes open. This just got proven for a fact now that
they do this i.e.
Sorry to badger everyone on the list but I never heard from even a
single person on this so felt maybe I'll repeat it, just in case, it
got unnoticed.
Any ideas if it's possible to either record greetings/names in a
different format than GSM OR be able to convert these voicemail
subscriber
in a different format OR be
able to convert these voicemail subscriber greetings in my database to
some other format?
Any ideas? suggestions?
Thanks in advance,
${RR}
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Bruce,
this might be able to help give you some hints or a place to start:
http://www.voip-info.org/wiki/view/QoS+Cisco
Hope that helps
\R
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Larry, am I missing something but you seem to be putting the externip
into the MYIP variable but reading some EXTERNIP variable through
$ENV{}. Shouldn't you be doing something like externip=${ENV{MYIP}}?
The other issue is also the use of curly brackets as opposed to
paranthesis. The snip from
Nitin,
I'm sure others have better advice but there's no best format per
se. Whatever makes asterisk and more importantly the CPU work less in
playing those prompts is probably best. from what I understand (*)
picks up the best suited format based on the capabilities of the
channel and endpoint.
Mate, I'm beginning to think that it can't be done. As in, maybe
you're not allowed to put anything into externip other a valid IP
address and the $ENV{} variable doesn't really work there. You might
want to decipher your externip by registering your server with a
dynamic dns service and then
Hello good people,
I'm sure this has been brought up previously but I basically wanted to
wait to resurrect this topic till 1.2.10 has been out for a little
while, like a cpl of mths. Now I think it has and I just wanted to
request for peope who've chosen to upgrade their systems to 1.2.10 to
Yes
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Follow the instructions here:
http://www.voip-info.org/wiki/view/Asterisk+app_conference
There's no config file where conferences are stored. You need to add
them to astdb using the 'database' CLI command like so: database put
conferences 1234 9
Look at the setting up conferences section in the
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