question.
Regards,
Ron
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| | | |
-
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-
I plan on doing it via DNS SRV only, but if a user register on asterisk
1 how can users at asterisk 4 reach that user. Thank You
Regards,
Ron
]
| | | |
| mysql cluster|
Thanks
Regards,
Ron
Grey Man wrote:
On Fri, Feb 29, 2008 at 2:01 AM, Ron [EMAIL
on
g729 format, so that asterisk can choose which file to play depending on
what codec is being used by the user.
TIA
Regards
Ron
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the users know. when they upload an MP3 i
would like to convert it and save it on different formats (e.g. g729 and
g711). thanks
regards
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server
2 via IAX?
hope my question is clear enough, thanks in advanced
Ron
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hi sir
yes you're correct, voice call works from 100 to 200-202 but not video call.
on my iax i simply added:
videosupport=yes
allow=h264
allow=h263
TIA
Ron
Danny Nicholas wrote:
To clarify, you have users 100, 101, and 102 on server 1 and 200, 201, and
202 on server 2. 100 can VC 101
they are only using codecs iLBC and
g711a. is that the reason why i cant hear the ivr? where do i start
checking it?
thank you
regards,
Ron
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Hi,
please see attached file for sip debug. i still cant hear anything. the
ivr after timeout will ring extension 300, but i already unplug that, so
it's now CHAN UNAVAILABLE.
Regards,
Ron
Steve Totaro wrote:
Post your SIP debug and verbose 3 for the call that does not have IVR
asterisk? and same scenario for 200 to 202 or vice versa.
what if 100 call 200 or 201? or 200 calls 100 or 100? will rtp still go
thru asterisk?
thank you
regards,
Ron
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Hi,
How could i trace what codec my voip provider is using?
I still can't make the IVR for that certain provider, i still can't hear
the sound, but i can see it connecting. where else should i look?
Regards,
Ron
Steve Totaro wrote:
On Thu, Feb 21, 2008 at 12:40 PM, Ron [EMAIL PROTECTED
. what does the error mean? TIA
Regards
Ron
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can create and overwrite files. any ideas? TIA
Regards
Ron
Ron wrote:
Hi All,
I'm trying to test asterisk voicemail on recording my own unavailable
message, busy message or temporary message. I was looking at the console
and saw this message:
app_voicemail store_file Memory map failed
php /var/www/voicemail.php on the shell, i can
receive the e-mail.
how would i know the asterisk is actually executing that command? also
i'm using unixodbc for my voicemail
odbcstorage=mydb
odbctable=mytable
not sure if that has an effect or not. TIA
Regards
Ron
Hi Tilghman,
yup my scripts starts with that line, is there anyway to check on the
logs if asterisk voicemail app is executing that command? thanks
Regards
Ron
Tilghman Lesher wrote:
On Wednesday 16 September 2009 11:35:31 Ron wrote:
Hi All,
I'm trying to use a php script to send voicemail
Hi Danny,
if the voicemail function is called then the AGI, wont the vooicemail
function already send an e-mail before going to the AGI? Thanks!
Regards
Ron
Danny Nicholas wrote:
The ODBC isn't having an effect, otherwise you couldn't run it stand-alone.
Voicemail.conf states that changing
thank you tilghman...that did the trick.. thanks again!
Tilghman Lesher wrote:
On Wednesday 16 September 2009 12:38:59 Ron wrote:
yup my scripts starts with that line, is there anyway to check on the
logs if asterisk voicemail app is executing that command? thanks
Okay, next sanity check
, thanks in advanced.
Regards
Ron
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the INVITE packet received on
the server. Where should i start tracing the delay? thanks!
Regards
Ron
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realtime mysql. second asterisk is to check
call limits going out to pstn (sorry this may confused you,but i'm
running multiple asterisk registrar server and send calls out to pstn
via another asterisk server to control channel limiting).
thank you
regards
Ron
Danny Nicholas wrote:
This may
of ports on specific users
so that the routers always keeps the port mapped to port of the ip phone .
TIA
Regards,
Ron
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i have also tried setting qualify='yes' but cpu usage spiked to 100%.
Ron wrote:
Hi All,
I been having issues on my users behind NAT, even if i hard set a
specific port on the phone, there are some network that NAT's it out to
a different port, in turn, some time later the phone could
hi sir,
yes i am using Linksys SPA's i set NAT Mapping enable and NAT Keep-live
to Yes. still sometimes the phone cannot be reach even though it is
registered.
regards
ron
SIP wrote:
Does the phone have some sort of NAT Keepalive setting? Often, the only
way to keep that port open
Hi,
Does using a different codec affect the volume of the voice?
i was testing g711 and g729, voice seems to be softer on g729 compared
to g711. sorry not really familiar on how codecs work.
regards
Ron
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i don't think that is high, 64x is at microseconds the 32x is at
milliseconds.
On 2/8/2010 8:54 PM, Christopher Brown wrote:
Hi Users,
I was wondering if someone of you have the same thing on CentOS 64x?
asterisk01*CLI core show translation
Translation times between formats (in
ast_carefulwrite: write() returned error:
Broken pipe
Has anyone encountered this? TIA.
Regards
Ron
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Hi,
Is there a way to check on AMI if a user is currently engage on the
phone? i would like to display on my portal whether a user is calling or
not.
thank you
regards
Ron
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Thank you. i think i would go for this solution.
On 8/5/10 4:53 PM, Gareth Blades wrote:
Ron wrote:
Hi,
Is there a way to check on AMI if a user is currently engage on the
phone? i would like to display on my portal whether a user is calling or
not.
thank you
regards
Ron
You could
hi,
just taking a wild guess here, are the extensions set to be in the same
pickupgroup?
regards
ron
On 8/15/10 7:01 AM, Cassius Smith wrote:
Hi all,
There are a lot of posts around the web about my question; unfortunately
I have not been able to get any of the solutions to work. I'm using
troubleshooting this issues?
TIA.
Regards
Ron
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oh so sorry about this. i just thought maybe someone had experienced the
same. sorry again.
regards
Ron
On 8/24/10 10:28 PM, David Backeberg wrote:
On Tue, Aug 24, 2010 at 9:05 AM, Ronnha...@gmail.com wrote:
hi all,
i recently subscribe for an isdn and terminate it on a 3825 router.
i
on linksys devices
e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no
client has complain about it.
hope anyone can help. thank you.
regards
Ron
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and asterisk crashed.
where do i enable the UDP session timeout? on the linksys devices or the
asterisk?
TIA
Regards
Ron
On 9/28/10 6:14 PM, Daniel Tryba wrote:
On Tue, Sep 28, 2010 at 09:28:24AM +0800, Ron wrote:
got this problem that IP phones could not re-register to my server. even
if device
seems to happen on linksys devices
e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no
client has complain about it.
hope anyone can help. thank you.
regards
Ron
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and each phone, i'm just wondering why it only happens on
linksys phones, using yealink and grandstream it's ok.
Thanks again.
Regards
Ron
On 9/29/10 7:39 AM, Danny Dias wrote:
Hello Ron..
The answer that i see here is only a trying to a Register...means the
REGISTRATION procedures are taking
I have setup a asterisk with freepbx, a TE122 and i have an ISDN.
My problem now is that callers are experiencing echo. checked on dmesg i
saw this:
# dmesg -c
dahdi: Disabled echo canceller NLP because of CED rx detected on channel 2
i searched google but found no soolution and i have no idea
Thank you Shaun, will try that. will that help on the echo issues users
are encountering during calls?
On 10/20/10 10:28 PM, Shaun Ruffell wrote:
On 10/20/2010 03:20 AM, Ron wrote:
I have setup a asterisk with freepbx, a TE122 and i have an ISDN.
My problem now is that callers
sorry i am not familiar with sshguard, but you can also try ossec by
trend micro http://www.ossec.net/ it can auto-block an IP address using
iptables. you can also follow this howto for asterisk:
http://sysbrain.wordpress.com/2010/05/24/asterisk-ossec-part-ii/
hope this helps.
regards
Ron
it to compute cost of the call. unfortunately with this setup, after i
hangup, it does not insert the CDR yet. so my AGI get-unqiueid.php does
not find any record. have i placed my ResetCDR(w) correctly?
thank you in advanced.
regards
Ron
with
disposition ANSWERED and one with NO ANSWER.
any ideas? thanks again.
regards
Ron
On 12/22/2010 7:29 PM, Ishfaq Malik wrote:
On Wed, 2010-12-22 at 18:10 +0800, Ron wrote:
Hi All,
I've got this dialplan:
[macro-callout-intl]
exten = s,1,ResetCDR(w)
exten = s,2,Dial(IAX2/${ARG1}/018
not
engaged anymore. is this possible on asterisk? what is that feature
called? i am using asterisk 1.4 with freepbx. Thanks in advance.
Regards
Ron
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thanks for all the reply. now that i know what it's called should be
easy to find something on the net.
btw, the URL below did not load anything on my side...it seems like it's
connected somewhere but just downloading slow, but thanks for it anyway.
regards
Ron
On 1/11/2011 1:20 AM, Paul
no response again, then again until it finally gets it. but if you will
notice on the 2nd ngrep, the asterisk replied to all the INVITE's it
received before it says Ringing. Really need help on this badly, anyone
has an idea. Thank you in advance.
Regards
Ron
U 172.16.0.6:5068 - 172.16.0.1:5060
Tx 200 result=0
my php code include something:
#!/usr/bin/php-cgi -q
?php
include('phpagi/phpagi.php');
$agi=new AGI();
$param = $argv[1];
$agi - exec(Noop,$param);
.
.
.
.
?
not sure where to check next i'm stumped, hope somebody can help. thanks
in advance.
Regards
Ron
Hi ben,
actually the ${OUTBOUND} is generated by the php... but based on what
will be received on ${ARG1}. unfortunately i am not getting the value
from the argument. not sure why. thanks again.
regards
Ron
On Friday 25,February,2011 11:10 AM, Ben Klang wrote:
On Feb 24, 2011, at 5:27 PM
Hey folks
I have a new setup with a TDM400P for a pair of analog extensions and a few
SIP phones. We seem to be experiencing a bunch of Crackeling when talking
between the analog and SIP extensions.
Any ideas?
Thanks,
Ron
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On Monday 14 March 2005 16:18, Eric Wieling wrote:
Ron Joffe wrote:
Hey folks
I have a new setup with a TDM400P for a pair of analog extensions and a
few SIP phones. We seem to be experiencing a bunch of Crackeling when
talking between the analog and SIP extensions.
Any ideas?
Yes
.
3/ European Resellers do/will not sell single contracts
What route is left for guy with a few Cisco phones in Europe?
Piracy?
/RANT
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Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG
h323_request(format)
i think) and so no codec negotiation and no voice.
Howto run up g711/H323 - * - g729/H323
You probably also need to allow ulaw
Change h323.conf to:
disallow=all
allow=alaw
allow=ulaw
allow=g729
- --
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[EMAIL PROTECTED]
FWD:519961 Gossiptel
with Zap/g1 which will take the lowets available channel.
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http
asterisk to yes. Anybody know what might be
wrong?
Try starting atserisk at run level 2 or 3 (debian/ubuntu does this a
little differently)
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[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version
loadInformation7 model=IP Phone 7960P0S30201/loadInformation7
/Default
- - - - - CUT HERE - - - -
This worked for me.
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment
. copy P0S30201.bin to P0030201.bin)
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iQEVAwUBQlMBvUtP
Board },
{ 0 }
};
This should allow reboots but YMMV (well, it works on my server!).
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using
/E1 Board },
{ 0 }
};
This should allow reboots but YMMV (well, it works on my server!).
HTH
- --
Ron Wellsted
I don't suppose any one can confirm this type of fix for the TDM boards. I
can make a guess but the systems are live and I will need to schedule outage
to attempt this fix
]: chan_sip.c:2894 process_sdp: No compatible codecs!
I can provide more detailed debug if required, but anyone ran across this.
Thanks,
Ron
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Stefan Gofferje wrote:
Ron Wellsted schrieb:
The VLAN has been configured on the phone, the rest of the config comes
from the tftp server. I was unable to find any way of setting the vlan
via the tftp server.
Then I suppose, it's a bug
asterisk then to key in
the extension of intended destination (SIP phones
number) that already connected behind asterisk ? I
utilize exten = s,1,dial,SIP/${EXTEN},but it doesn't
work. Appreciate for the helps.
Regards,
ron
__
Do you Yahoo!?
Yahoo
Hi All,
Does Asterisk has the ability to recognize the CLI
that comes in GSM format, as we know that PSTN has
different format compared to GSM. Is there any
specific configuration needed ?
Thanks,
ron
__
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Jazz up your holiday
? Please help
thanks,
ron
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All your favorites on one personal page Try My Yahoo!
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every time, so I do not think that I can put in seperate
extensions.
Thank
You,
Ron
Frederick
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Dunno how to tell if
IAX2/username/6 is IAX2/username
I was wondering if
there is some sort of wildcard character that can be used here? The number
changes every time, so I do not think that I can put in seperate
extensions.
Thank
You,
Ron
Frederick
of experimentation and
eventual success. Now, if I can only remember what I did to get it to
work... ;)
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.6 (GNU/Linux)
Comment: Using
version 1.0.4
- - - - - - 8 snipped
Change your:
load = chan_zap.so
load = res_musiconhold.so
To:
load = res_features.so
load = res_musiconhold.so
load = chan_zap.so
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD: 519961
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Have you built your kernel on that machine?
The errors suggest that while the kernel sources are installed, the
kernel has not been built.
Check on the exact procedure for your distribution.
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL
/ Maybe - 79XXG will work
3/ With a special cable/dongle (a la wikki)
I am looking at getting several 20 x 7960 (not Gs) to work with * and a
NetGear FSM7326P switch. Do I also need to get a PowerDSine converter
dongle for each phone?
TIA
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL
question...
Ron
On Fri, 27 Feb 2004, T. Chan wrote:
Hi, all
I wonder when passing calls through asterisk with H323, is there anyway to
find out what codec the calls are using, anyone can help please, thanks alot
!
TC
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Outgoing mail is certified Virus Free.
Checked by AVG anti-virus
Hi, I already have one g.729 license on *. Could anyone tell me if I want
to add a few more, can I just buy these online and follow their
installation instruction, and * will add these addtional licenses? Or this
will invalidate my current license?
thanks
ron
I know this has come up before but...
Has anyone done anything to get an IAX client built on Windows?
I thought someone had started one, but I haven't heard anything about it
since - and that was months ago?
Anyone have any idea what the status is?
--
Ron Gage - Saginaw, Michigan
I am looking
I just set up asterisk for the first time and I keep seeing the following across the
console:
NOTICE[1192484144]: File sched.c, Line 209 (sched_settime): Request to schedule in the
past
any idea what this means?
Thanks
Ron
___
Asterisk-Users
I don't know if this would help. But the latest version have two login
options now.
One is the administrator and the other one is a simple user.
If you're not seeing the other forms there probably you're logged in as
a simple user.
HTH
-Original Message-
From: [EMAIL PROTECTED]
to make (also) calls
into Asterisk.
How can I call Asterisk with any of these ISDN phones, without making an
outgoing (paid) call through the telephone switch ?
Should I need to install two Fritz! card, one connected to the ISDN/NT
line, the other to the ISDN phones ? :-))
Thanks for any hint.
Ron
anything I can do to fix this problem?
thanks
Ron
line.
Thank you
Ron
dial usually fails, both pcphoneline and FWD IPP. I have only tried with Sipura ATAs.
Have anyone use similar senerios and got this to work? Using a different ATA? Or any product similar to FWD Internet Phone Patch or pcphoneline port converter and works better?
Thanks
Ron
Hi,
This is not gonna work, is it? Is there such thing as Dial_but_not_connect_? I am trying to do the same thingbut don't know how to accomplish this. If you've or anyone here figured out, please let me know.
Thank you very much,
Ron
[EMAIL PROTECTED] wrote:
I am trying implement two-stage
a problem bridging. If the codecs are the same on both ends then there is no problem.
Is it different to call Dial from extension.conf than to call from AGI-exec('Dial'...)?
Thanks
Ron
Hi all,
If I'm just using Asterisk as PBX and calls going
through between ouside lines and inside extensions,
(not using any softphone running on the asterisk pc),
does what soundcard I use affect voice quality at all?
Do I have to get a full duplex soundcard?
Thanks
Ron
hisDOESN'T work.
There some errors about invalid argument.
If I were to do
$AGI-exec('Dial',"SIP/PHONE1"), where PHONE1 is an phone registered directory with Asterisk, then it works.
Any hints?
Thanks!
ron
Thank you. This explains it.Nathaniel Powning [EMAIL PROTECTED] wrote:
On Mon, 12 Apr 2004, Ron McMillin wrote: Is it possible to dial an OUTSIDE SIP address while inside AGI application? For example, within extension context, I could use [from-sip] exten = 7723,1,Dial(SIP/[EMAIL PROTECTED
Hi all,
Do you know if there's any free public SIP proxy server that is more reliable that FWD and Iptel?
Thanks
Ron
Thank you.Simon Brown [EMAIL PROTECTED] wrote:
You could try these:
voiptalk - www.voiptalk.org
sipgate - www.sipgate.de
Simon
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McMillinSent: Thursday, 15 April 2004 15:29To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Most
read somewhere that combining SER with Asterisk, or vice versa,
will be very good, how come?
Regards,
Ron
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Hi All,
How can I be able to define multiple SIP extensions?
Do I have to define each extensions on sip.conf?
For example, extension 2000-8000, do I have to define it one by one
on sip.conf?
[2000]
secret=2000
type=friend
username=user2000
..
..
[2001]
secret=2001
type=friend
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Extensions
Hello Ron Ramos
With the power of bash it's easy
copy this code to a file name it createsip.sh
#!/bin/bash
for ((i=2000 ; i 8000; i++ )); do
echo [$i]
echo
, is it possible to save the actual logs to a file (ie: As the
messages are bring printed, save them all to a file to be viewed later).
I utilize this command:
nohup script -f -c asterisk -vvvTn /tmp/asterisk.log
To start up my apps. This will log everything to a log file.
Ron
not looking for screen
functionality.
-F is not an option on my version of asterisk.
Ron
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.
Ron
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If I issue a restart gracefully command, the system will wait until all
channels are idle before restarting.
During the time the system is waiting for idle activity, is there a command
that can let me know it is in graceful restart wait mode ?
Thanks,
Ron
step approach:
1. Issue a restart gracefully
2. If x minutes go by and and all channels have not cleared then issue a
restart now
Each of these items might not know of the other so it would be helpful for
item two to know that asterisk is in a restart gracefully mode.
Ron
asterisk
asterisk
This sounds like a feature waiting to be coded. Something like show restart
status
Anyone have experience with setting up a bounty for a developer to make a
change like this ?
Ron
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On Wednesday 29 August 2007 08:12, equis software wrote:
Hi!
I need to use text to speech, what is the best application?
Thanks!
Cepstral
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To
lately?
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Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
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On Saturday 08 September 2007 04:52, satish patel wrote:
I have asterisk server with 2 E1 port now i want to
redendecy for my server means one of server goes down automatically second
goes in active mode is it possible and how to switch E1 to second server ??
start at 20).
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Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
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Christian wrote:
Hi,
What parameter should I use to that command?
On 2007-09-09 at 13:45 Ron Wellsted wrote:
Tzafrir Cohen wrote:
On Sun, Sep 09, 2007 at 02:32:14AM +0200, Christian wrote:
Hi all,
Have just installed v1.4.11 of Asterisk
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Christian wrote:
Hello,
On 2007-09-09 at 22:36 Ron Wellsted wrote:
Christian wrote:
Hi,
What parameter should I use to that command?
On 2007-09-09 at 13:45 Ron Wellsted wrote:
Tzafrir Cohen wrote:
On Sun, Sep 09, 2007 at 02:32:14AM
says they priorize
VoIP traffic based on VLAN ... are those methods the same ?
No, they are not.
Ron Arts
Thank you,
Luis A P Barbosa
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.
Ron
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