BJ Weschke schrieb:
...
I have no modifiers in my dial command.
...
One reason might be is if you are passing parameters in app_dial (eg.
Hi,
sorry, I did use the wrong expression. No, there
is no parameter like tT in the Dial command.
I think, I've made everything according to the docs.
A
Hey. Maybe you can give me a hand with configuring my Linux box to send out
emails? I've installed sendmail as per *several websites* and it's installed
and running. I've gone into the voicemail.conf file and specified to allow
attachments, etc. And, yes, I restarted Asterisk. Technically, I r
Dear Moises,
Thank you for your reply! I am not sure how to use the testcall tool
to debug, but here we go with what I tried. The Main Thread message does
never stop showing on the screen, I bet this is not the expected
behavior tho. Maybe this can be of any help:
asterisk-test:/usr/local
Hi Josué,I just got the confirmation about integrating TE110P with TMS2 of Hipath 3750. Your help will be much appreciated.The configuration is as follow:PSTN -> HIPATH 3750 (14 analog trunk lines) -> TMS2 -> TE110P -> Asterisk
All extensions of Hipath 3750 are analog (120 extensions)I know that i
Firstly, thanks for the information, but I dont seem to get this SNAP work. I found out that the disadvantage of this is most computer dont come with mozilla, therefore for some non-IT literal is quite troublesome for them.
Hmm..hopefully someone can provide me some info on click n call features.
Hello Victor,
> Hi,
> I'm still a newbie, but try to help you,
>
THX ;-))
> And voicemail.conf part is:
> [general]
> format=wav49
> maxmessage=180
> minmessage=2
> maxsilence=2
> silencethreshold=150
> maxlogins=3
> [EMAIL PROTECTED]
> skipms=3000
>
> [victor]
> victor => 1234, Victor Moreno, [
Very nice phones. There is no problem when conected to Asterisk (for
about 6 months now)
any body know this phone? support NAT? and standart codecs of asterisk ?
-FD
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/etc/asterisk/logger.conf
-FD
Hi list!
I have a Centos 4.3 box running Asterisk 1.2.9.1 with FreePBX 2.0.1
I noticed that this setup is keeping a full asterisk log which, after
1 month in production, has already grown to 1300 Mb in size. This is
the log location : /var/log/asterisk/full
Wh
Any one knows how to make a call conference using a voip
gateway connected to asterisk.
In mean what should I press (extension) to have another
line and make the conference .
regards
*
No employee or agent is authorized to conclu
Any one knows how to make a call conference using a voip
gateway connected to asterisk.
In mean what should I press (extension) to
have another line and make the conference .
regards
*
No employee or agent is authorized to con
Hi
I have
next working sheme
Hicom
350 - (Diun2 card)with DSS1- Asterisk with Quiad E1
This
is work fine
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
NguyenSent: Tuesday, June 13, 2006 10:32 AMTo:
Asterisk Users Mailing List - Non-C
Hello,
I have found that the timeout 't' takes to much time to be executed,
around 10 seconds.
Is there a place to configure this timeout ?
thanks
--
Victor Moreno
CISL SPAIN, S.L.
Parque Tecnológico de Andalucía
Edif. Bic Euronova
Avda. Juan López Peñalver, 21
29590 Campanillas (Málaga)
Fax +
Hi all,
Eyebeam has a sip-chat function and it would be nice if I would be
able to use it. But the problem is that I can't really find
information about it.
I can just try to send a message and on the Asterisk console a
message like this appears:
Jun 13 10:05:25 WARNING[6512]: chan_sip
When a caller in the queue is connected to an agent, the call is placed
to the extension and context specified using Agentcallbacklogin. This
allows for me to add extra things to the diaplan *before* calling the agent.
Now, I want to be able to use a device, rather than agents. So I can use
ad
Hi,
I'm using Asterisk 1.2.1 and I noticed that the voicemail suddenly exits
if I press ANY key on the phone while the first or the second voice
messages (es: vm-no.gsm or vm-youhave.gsm ) are played. I googled around
but found nothing.
How can I solve this problem?
TIA
Giorgio Incantal
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+timeout
-FD
Hello,
I have found that the timeout 't' takes to much time to be executed,
around 10 seconds.
Is there a place to configure this timeout ?
thanks
___
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I want to start a community based voip network projcet and am thinkimg of
using VOCAL and asterisk gateways. my question is, has anyone bench
marked asterisk vs VOCAL? is it a wise idea to use VOCAL + Asterisk or
Asterisk all the way.am expecting 1000 -> 5000 users..
your thoughts
These days you don't have to worry much about your write cache unless
you're running application where once single byte changed will affect
whole file.
Look at it this way, the only corruption will occur is whatever the
files were open by asterisk at the time of the crash. And only up to the
point
Any one knows how to make a call conference using a voip
gateway connected to asterisk.
In mean what should I press (extension) to
have another line and make the conference .
regards
*
No employee or agent is authorized to
Hi,
As long for HiPath 4000 callerID name doesn't work, only number
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Pavel Jezek
> Sent: Thursday, May 25, 2006 9:33 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello,
We connected Sipura 2100 to Asterisk PBX. Plugged simple
phone.
Trying to call – everything works ok. When we take out
phone from Sipura, and trying to call, Asterisk shows,
that Sipura is RINGING – without phone connected to
it. How could that be?
-- SIP/240-2b03 i
Akpome Akpoguma a écrit :
I want to start a community based voip network projcet and am thinkimg
of using VOCAL and asterisk gateways. my question is, has anyone
bench marked asterisk vs VOCAL? is it a wise idea to use VOCAL +
Asterisk or Asterisk all the way.am expecting 1000 -> 5
Any one knows how to make a call conference using a voip
gateway connected to asterisk.
In mean what should I press (extension) to
have another line and make the conference .
regards
*
No employee or agent is authorized to con
Have you sent this enough times yet?
On 13/06/06, Khaled Chehab <[EMAIL PROTECTED]> wrote:
Any one knows how to make a call conference using a voip gateway connected
to asterisk.
In mean what should I press (extension) to have another line and make the
conference .
regards
___
Hi Viktor,So where is the PSTN side on your schema? PSTN -> Asterisk -> Hicom 350? Or PSTN -> Hicom 350-> Asterisk?ThanksNguyenOn 6/13/06,
Viktor Tatianin <[EMAIL PROTECTED]> wrote:
Hi
I have
next working sheme
Hicom
350 - (Diun2 card)with DSS1- Asterisk with Quiad E1
This
is work fine
Hi,Oh, I just want to get it to work. Caller Name is something luxurious for us .NguyenOn 6/13/06, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
Hi,As long for HiPath 4000 callerID name doesn't work, only number> -Original Message-> From: [EMAIL PROTECTED]
[mailto:asterisk-users-> [EMA
Hi All!
I am facing some delay in conferencing.
Using DIAX for Voip calls ,no hardware used yet
I am using MeetMe to achieve conferencing and am having a lot of delays.
Can anyone tell me how to reduce the delay
Regards,
Amna Saleem
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In zapata.conf I have, among other things:
; Incoming only
group=0 ; Zap/g0
signalling=pri_cpe
context=from-pstn
channel => 1-10
; Outgoing (only?)
group=1 ; Zap/g1
channel => 11-15,17-21
; To/From Alcatel
group=2 ; Zap/g2
signalling=pri_net
context=from-alcatel
channel => 32-46,48-62
Then
each ip phone need to register to be able to call the conference. Firstly, you need to create user with username and password under sip.conf, Then you need to create conference room which is meetme in meetme.conf
then you need to create extension to point to the conference room in extensions.confaf
Hi Everyone,
This morning I upgraded Asterisk from 1.2.7.1 to 1.2.9.1 along with libpri to version 1.2.3 and Zaptel to 1.2.6. Then stopped and restarted all the Asterisk and Zaptel components. Service resumed as usual but after about 30minutes the console was filled with errors and the PRI Z
Hi,
I made the mistake of upgrading both my Linux box (to Ubuntu 6.06) and
Asterisk (to 1.2.9.1) at the same time. Now, when trying to compile
mpg123 - using the tried and true "make mpg123" -, the build fails with
an error
make[3]: Entering directory `/usr/src/asterisk-1.2.9.1/mpg123-0.59r'
ma
Try make on its own and read what it says. You probably want make linux
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marc
Rohlfing
Sent: 13 June 2006 12:09
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Compiling mpg123 un
On 6/13/06, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
When a caller in the queue is connected to an agent, the call is placed
to the extension and context specified using Agentcallbacklogin. This
allows for me to add extra things to the diaplan *before* calling the agent.
Now, I want to be
I have been running the same software versions together with a digium
single port PRI card in the UK and have not experienced any problems
since the upgrade.
On Tue, 2006-06-13 at 12:00, Chris Teesdale wrote:
> Hi Everyone,
>
> This morning I upgraded Asterisk from 1.2.7.1 to 1.2.9.1 along with
>
> "EP" == Erick Perez <[EMAIL PROTECTED]> writes:
EP> BJ, when you say it is more than adequate, what do you do to
EP> calculate? there *must* be a way to at least tell if the
EP> motherboardboard/cpu will achieve results. I just don't want to
EP> install it and then after a 5th user going to
Hallo all,
Last night I have successfully setup Asterisk Realtime with mysql. but I
have one problem regarding the "Ex-Girlfrind"-Functionality.
The example: I have a fax running on a specific extension (300) and I
want that one to call out via ISDN, but it simply gets IGNORED.
I have tried
Why still use mpg123?
Start using format_mp3 from asterisk-addons and your * will play mp3 by itself...
K
On 6/13/06, Marc Rohlfing <[EMAIL PROTECTED]> wrote:
Hi,I made the mistake of upgrading both my Linux box (to Ubuntu 6.06) andAsterisk (to
1.2.9.1) at the same time. Now, when trying to com
I use PSTN -> Hicom 350-> Asterisk
Asterisk I use for voice mail, ivr and
gateway for voice over ip
I try connect Asterisk to PSTN with
EDSS1 signaling it work fine
at PSTN side statioon type
5ESS
What problem you have
?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:
Ahhh, that would explain it. I setup my firewall (eg. Shorewall) to
allow incoming TCP connections to port 5060. I've changed it to UDP
port 5060 and it works great! (well, Asterisk says "Forbidden", but
that's just a simple config problem I'm sure)
Which other ports do I need to forward/open
Hi Nguyen.
The Asterisk with configured 50 SIP Phones is integrated with HiPath 3750, through a board TMS2. For access the PSTN I created a context in asterisk so that asterisk has access HiPath 3750 and uses the LCR's that I configured in the HiPath. The CDR's do asterisk is registered no billi
Try this:
http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules
Peter
On 13/06/06, John Klimek <[EMAIL PROTECTED]> wrote:
Ahhh, that would explain it. I setup my firewall (eg. Shorewall) to
allow incoming TCP connections to port 5060. I've changed it to UDP
port 5060 and it wo
Erick Perez wrote:
I just don't want to install it and then after a 5th user going to
call someone the asterisk begin to crash due to lack of resuources.
Check the wiki for SIP load generation tools you can use to test your
setup on any number of calls you like.
On Tue, Jun 13, 2006 at 09:53:36AM +0200, Filip Dr?gowski wrote:
> Very nice phones. There is no problem when conected to Asterisk (for
> about 6 months now)
> >any body know this phone? support NAT? and standart codecs of asterisk ?
thank you all!!
> >
>
> -FD
> _
Attilla De Groot wrote:
Hi all,
Eyebeam has a sip-chat function and it would be nice if I would be
able to use it. But the problem is that I can't really find
information about it.
I can just try to send a message and on the Asterisk console a
message like this appears:
Jun 13 10:05:
Andrew Kohlsmith wrote:
On Monday 12 June 2006 17:55, shadowym wrote:
Believe me, you can drive yourself insane trying to come up with some
magical formula that JUST works because it usually won't happen that way.
Software echo cancellers are simply not good enough for many situations.
Mike Fedyk wrote:
Or any polycom phone that has speakerphone like the IP501 and IP430.
Time Bandit wrote:
Can you, or anyone else comment on the speakerphone ability of the
GVX-3000
? We run the GXP-2000's and for the most part are happy with them,
but for
upper management we're looking at
Joao. The more important thing about testcall is that it allows you to
isolate the problem (you dont need asterisk) .
I would recommend you to read this document I wrote:
http://phpmexic.u33.0web-hosting.com/wordpress/misc/mfcr2-asterisk-unicall.pdf
Here is described a bit more detailed. Is in s
Hi Guys,
I'm starting to work on Asterisk, trying to see if it will fit our
needs, but so far it seems it doesn't support TBCT(Two B Channel Transfer).
I've found a couple of links that was talking about TBCT, and someone
had posted a bounty for that feature, but no news since 2003.
I've also
Hello,
I've receive no response, no idea??
Bets regards,
Olivier S;
Tzafrir Cohen a écrit :
Jut as usual with Zaptel: Zap/NNN (e.g: Zap/1 , Zap/2) for individual
channels. And gNNN and similar work just the same.
OK, in extensions.conf, i put the contexts PSTN and INTERNAL as:
[PSTN] ;
Derek Lee-Wo wrote:
I just got a TDM400P with 2 FXO cards. I got it all configured and I
can place and receive calls.
I seem to be getting static on the call, mainly when I speak. E.g, if
I call someone, I can hear them just fine, but they would hear static.
Not a lot...more like a constant ba
Are
there any generic install guidelines for compiling the Zaptel drivers on FC5?
This is my first install of Asterisk (and my first FC5 system) and I'm having a
great deal of trouble getting it to cooperate. make clean and make are
definitely not playing nice, telling me that "You don't app
I think here are 2 mixed subject One Substitution of the GXP 3000 Video Phone Phone with a great speaker phone. For the second Subject a think Polycom are the greatest.
On 6/13/06, Steve Underwood <[EMAIL PROTECTED]> wrote:
Mike Fedyk wrote:> Or any polycom phone that has speakerphone like
a2billing does both prepaid & postpaid accounts. Each account has a calling card but you can add a sip or iax friend to that card. So, you give a card $20, and that card also has a sip user attached... the sip user will also have $20.
RTFM a little closer ;-)
bp
On 6/12/06, Daniel Salama <[EM
J.J. Feminella wrote:
Are there any generic install guidelines for compiling the Zaptel
drivers on FC5? This is my first install of Asterisk (and my first FC5
system) and I'm having a great deal of trouble getting it to
cooperate. make clean and make are definitely not playing nice,
telling me
Hello,
I look at voip-info for a simple billing application .
I wish to calculate price to pay according to the
datas stored in cdr table (unixodbc/mysql).
what do you advise me ?
Harry
__
Do You Yahoo!?
En finir avec le spam? Yahoo! Mail vous o
Both Festival and Cepstral app commands take a string but you could use
external programming to pass a file or in this modified example from "The
Future of Telephony" use a filename.
Lines indented are all on one line.
(copy text to file)
echo "All circuits are busy. Please try your call l
Hello
Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2
is much more sensitive against high CPU loads?
Also, does Asterisk support and use multiprocessor architectures, such as Xeon?
Regards
Jon
--
No virus found in this outgoing message.
Checked by AVG Free Ed
Hi,Well, I'm working with a2billing http://www.asterisk2billing.org/, without problems.RegardsOn 6/13/06,
[EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Hello,I look at voip-info for a simple billing application .I wish to calculate price to pay according to thedatas stored in cdr table (unixodbc/my
dual support yes.. however i read a few articles on the fuct that single with double the ram is better..something about the bus or sshare between both processors i think.i would go AMD opteron, but that me.
or sunOn 6/13/06, Jon Schøpzinsky <[EMAIL PROTECTED]> wrote:
HelloIs it correct that IAX2 us
Hi all,
thanks to the all of you. This list is very interesting also for a newby like
me.
My problem: I just setup my first full working asterisk installation with this
config:
1. n.1 GXP-2000
2. n.4 Budgetone 102
3. n.1 TDM400p (3 FXS, 1 FXO)
Everything seems to work fine, but the sidetone... it
I use IAX2 quite a bit and I haven't really noticed any difference between
IAX2 and SIP. CPU usage in Asterisk is aggravated by transcoding, changing
one audio format to another, and SIP or IAX2 is simply the protocol used to
carry the audio. Any function of Asterisk will be affected by high system
Unless it's changed recently, Asterik doesn't support the SIP 'MESSAGE' command.
Doug.
> -Original Message-
> From: Attilla De Groot [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, June 13, 2006 2:41 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] A
Turn down your microphone TX gains on the phones. On my TDM400 with Vista
350's I had to crank the mic value way down. This is not specific to FXS
phones, on my Snom 200's sidetone is so bad, that an appropriate setting for
mic gain is '2' (out of 8)
hth
-Original Message-
From: Marco Saj
Hi,
I'm running SER with Asterisk, and I've configured VoicemailMain like this:
exten => 201,1,VoicemailMain(@default)
exten => 201,2,Hangup()
Although, after any user enter his voicemailmain mailbox, when the phone
is hung up, the call still continues running in Asterisk, because I can
see i
Try "rpm -qa kernel-devel" or "rpm -qa kernel-smp-devel" to verify
they're installed.
If not do a "yum install kernel-devel or kernel-smp-devel" depending on
which you have.
Chad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J.J.
FeminellaSent: June 13, 2006 9:51 AMTo:
hi all,
i'am new in the
asterisk business and i have to solve following experimental
problem:
**
user 1 (calls
number 12345) ->pstn line --> *FXO
1 FXO
2*>pstn line ---> user 2 (wit
Well thanks all for your responses. My original intention was to
address the mistic know-how about machine calculations, and I still
feel the shadows remain.
Why? Because to achieve a 24 user PBX-only/One E1, I was going to
install a Dual Xeon with 2 GB of RAM and a 3ware card runing RAID-1
with
When i did this test ages ago, i found out that iax was worse than sip,
but sip was worse than trunked iax.
Joachim
olin Anderson wrote:
I use IAX2 quite a bit and I haven't really noticed any difference between
IAX2 and SIP. CPU usage in Asterisk is aggravated by transcoding, changing
one au
Maybe you are over thinking it. I have 200 users on a quad Xeon 700 with 2
PRI's and we regularly have 40-60 channels up, no problem (believe me, if
there was a problem I'd have 200 guys freaking on my head). I rarely see >
30% single-CPU usage, and that's only when Sendmail is invoked to send out
Is there a way to patch an existing Asterisk 1.2.5 version with the
follow me application?
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 25, 2006 1:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] As
taskset does not seem to exist on redhad 9 nor freebsd..;)On 6/13/06, Zoa <[EMAIL PROTECTED]> wrote:
When i did this test ages ago, i found out that iax was worse than sip,but sip was worse than trunked iax.
Joachimolin Anderson wrote:> I use IAX2 quite a bit and I haven't really noticed any differ
2002
called. They want their operating system back. :- ) >
-Original Message-From: Mike Lynchfield
[mailto:[EMAIL PROTECTED]Sent: Tuesday, June 13, 2006 9:42
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] IAX2 Vs SIP cpu
loadta
Hi,
is there a RHEL4 RPM for the Festival text-to-speech system?
Thanks
--
Domenico Viggiani
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On 6/13/06, Kevin Kiely <[EMAIL PROTECTED]> wrote:
Is there a way to patch an existing Asterisk 1.2.5 version with the
follow me application?
Not at this time, no. There are a number of API calls in the
application that are specific to the new version of Asterisk and will
not port back to the
um, "yum install festival" worked for me.
-Original Message-
From: Mimmus [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 9:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Festival RPM?
Hi,
is there a RHEL4 RPM for the Festival text-t
Mike Lynchfield wrote:
taskset does not seem to exist on redhad 9 nor freebsd..
;)
On Fedora Core 4, it is provided by the schedutils RPM.
Steve
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Go to:
http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
and search for "affinity", iirc i explain there how to do it with echo
instead of tasksel.
Zoa
Colin Anderson wrote:
2002 called. They want their operating system back. :- ) >
-Original Me
Hi
I've problem with incoming call quality to GSM gateway connected to
beronet card (BN8S0),
-> [ GSM Gateway ] ---> [ BN8S0 ] asterisk
Port connected to GSM gatway is in TE mode , gateway is in NT mode ,
When I dialin to cellphone numer , call goe
On 06/13/06 22:49 Colin Anderson said the following:
Although this may have changed in the newer 1.2.X series of Asterisk, I
believe that Asterisk does not support SMP from the perspective of
isnt asterisk multithreaded ? on a proper OS thread implementation, threads
can migrate across CPUs,
The cold hard truth is that if Asterisk cannot achieve 99.999% uptime
without becoming much more expensive that a traditional PBX then it is not a
viable alternative. Even elcheapo Key systems are rated for five nines.
That is what the telco world requires unless your just using Asterisk in
your
Hi Erick -
Now This thread tells me that my dual core pentium d (a 700$ computer)
will do the work. (the other equipment costs about 3500.00$). I do
realize that i must minimize transcoding (ulaw all the way) but you're
telling me it will work for 24 users (let's say 30 for round numbers)
all wi
Hi folks,
I'm really confused, so please help me, or at least give me some
pointers to clarify this issue.
Can I mix "Static" and "Real" realtime?
Is there a way to easily switch from one to another, say, for sip.conf?
Which are the major benefits of "Real" realtime?
Please help me!
Thanks in adv
- Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
> Now, I want to be able to use a device, rather than agents. So I can
> use
> addQueueMember and add my SIP device. However, I still want to do a
> couple of things before the device is called.
This is what the Local channel (chan_local) is f
TRX Teleocmmunications "the VoIP provider that pays you" would like to
assist those that make asterisk better. To that end we are setting up a
program where the community itself can help double the bounty for all of
the outstanding code that is wanted but not yet present.
TRX will match any bou
Don't forget to be sure your power supplies are reliable, and if necessary redundant.
On 6/13/06, Colin Anderson <[EMAIL PROTECTED]> wrote:
Maybe you are over thinking it. I have 200 users on a quad Xeon 700 with 2PRI's and we regularly have 40-60 channels up, no problem (believe me, if
there was a
Hi Andrea,
yes you can, static realtime helps you out with replacing config files
one by another. The "real" realtime you mean is for the dynamic backend,
which gives you the ability to be reread with every call, so asterisk
doesnt even need a reload.
So the main difference between "static" and "
There is work by the devs to threading in the IAX and SIP channels, I
believe. I don't know if it's made it's way back to -HEAD or not, maybe kpf
can give a definitive answer. I remember reading something by Mark S earlier
this year that he had IAX threading working.
-Original Message-
Fr
Erick,
Please see message: "Paul Mahler: Asterisk Scalability" at the following
link:
http://asteriskvoip.blogspot.com/2005_06_01_asteriskvoip_archive.html
Much slower machine than yours was involved in tests:
47 Simultaneous VoiceMail messages
333 Simultaneous SIP Calls
122 Pass through cal
Hi,
I'm running SER with Asterisk, and I've configured VoicemailMain like this:
exten => 201,1,VoicemailMain(@default)
exten => 201,2,Hangup()
Although, after any user enter his voicemailmain mailbox, when the phone
is hung up, the call still continues running in Asterisk, because I can
see i
On Tue, 2006-06-13 at 23:47 +0800, Dinesh Nair wrote:
> On 06/13/06 22:49 Colin Anderson said the following:
> > Although this may have changed in the newer 1.2.X series of Asterisk, I
> > believe that Asterisk does not support SMP from the perspective of
>
> isnt asterisk multithreaded ? on a pro
shadowym wrote:
The cold hard truth is that if Asterisk cannot achieve 99.999% uptime
without becoming much more expensive that a traditional PBX then it is not a
viable alternative. Even elcheapo Key systems are rated for five nines.
Even massive redundant public exchanges struggle for 99.9
On Jun 13, 2006, at 2:39 AM, Mindaugas Kezys wrote:
Hello, We connected Sipura 2100 to Asterisk PBX. Plugged simple phone.Trying to call – everything works ok. When we take out phone from Sipura, and trying to call, Asterisk shows,that Sipura is RINGING – without phone connected to it. How coul
First, thank you for your quick and kind answer.
I cannot change the TX gain on the Grandstream phones, or atleast I don't know
how to...
Can anybody help, please?
Thanks in advance,
Marco
On Tue, 13 Jun 2006 08:54:31 -0600, Colin Anderson wrote
> Turn down your microphone TX gains on the phones
I'm trying to setup Asterisk on my Linksys WRT54G router but it only
has 4MB of storage (and only about 250KB free with OpenWRT linux
installed).
I'd like to setup voicemail, but I'm afraid I don't have enough
storage on my router. Can I somehow store this on my VOIP provider?
(eg. BroadVoice)
Don't forget to be sure your power supplies are reliable, and if necessary
redundant.
Good
point. I'm a Tier 1 server bigot, myself, although I have done rollouts in a
clone 4U with *freakin good* redundant PSU's. Nuthin beats Tier 1 redundant
goodness...
__
Hi,
I have two asterisk systems that share voicemail on an
NFS. I recently upgraded to Asterisk 1.2.9.1.
After the upgrade, the voicemail gets cut out after
about 5 seconds of recording. Any ideas on what might
be causing this?
__
Do You Yahoo!?
Anybody know anything about this. Is it just a rebranded Digium FXS card or
is this something new?
http://support.intel.com/products/desktop/adapters/600sm/index.htm
___
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Asterisk-Users mailing
Power supply is a 500w gold edition with an APC UPS/regulator.
So, It seems that no table exists to calculate demand imposed on the
equipment *prior* to purchase/quote to a customer (not good for
sales/consulting).
It is a bit strange that no one has come to this kind of table (even
digium), giv
No way.. im keeping it.. be uptime since.. let me see..uptime 13:12:56 up 587 days, 13 users, load average: 0.11, 0.02, 0.01i think i got an old sun laying around in the noc that been just compiling since 1973..
lol nah but i remember the old days when a password leak on our hosting clients woul
Ciao Michael,
[cut]
> you should although never mix up both in one particular type. What I
> mean is if you add IAX "realtime" support with something like
>
> iaxpeers => mysql,asterisk,iax_peers
> iaxusers => mysql,asterisk,iax_users
>
> you shouldn't make an additional static like here:
>
>
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