Scott Higginbotham wrote:
I'm interesting in testing this.
OFF LIST PLEASE, FOLKS!!
The list has enough traffic without the 10,000 me too mails that are
likely to follow if nobody points out that it's bad netiquette.
B.
--
This message has been scanned for viruses and
dangerous content
On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said:
On 11 Oct 2006, at 19:35, Dean Collins wrote:
Lol - use a real PC maybe :P
Nah, that would be dull.
In some ways the mac intel is nearer to a 'normal PC'
(whatever that is) than the systems I normally run asterisk on
- a
On 16 Oct 2006, at 07:15, Martin Joseph wrote:
On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said:
On 11 Oct 2006, at 19:35, Dean Collins wrote:
Lol - use a real PC maybe :P
Nah, that would be dull.
In some ways the mac intel is nearer to a 'normal PC'
(whatever that is) than
Hi,I have Sipura SPA-841 with two lines. And I have some little problems with it: 1) How to turn off alerting tone in Sipura, cause when I'm trying to call , I hear two alerting tones (I also have audiocodes product and I don't hear two alerting tones, just tone)?
2)The second problem: How to
Hi all,I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines.6 plus 6 is 12, how come it's 24?if I have 24 PSTN lines, i'll be needing 24 FXOs.
Pls. elaborate.thanks.Lito
each module have 4 ports, we have a tdm2400 with 3 FXO modules, so that's a total of 12 FXO ports, HTH.caloyOn 10/16/06, Lito Lampitoc
[EMAIL PROTECTED] wrote:Hi all,I'm confused, in digium website, it says:
TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24
The TDM2400P supports up to six quad modules -- each quad module
supports EITHER four FXS ports OR four FXO ports...
THEREFORE
with 6 quad FXO modules one has 24 FXO ports,
with 5 quad FXO modules and 1 quad FXS module one has 20 FXO ports and
4 FXS ports...
the remainder of these examples
Matthew Fredrickson schrieb:
On Oct 12, 2006, at 1:17 PM, Johann Steinwendtner wrote:
Hello !
I 've some questions how bridging of ISDN calls is done.
Assume an asterisk system with a TE405 card equipped.
(PRI1 - PRI4)
An incoming ISDN call on PRI1 is transfered back to
PRI3. Unless there is
Nitin Gupta wrote:
Hi,
Is there any way asterisk can detect if the outgoing call is being
received by a user or it has been forwarded to his voicemail.
If you wish to detect forwarding to voicemail (or another number) at the
telco level (e.g. mobile phone or fixed lines) , it may be possible
Hi,
I'm using mysql to store my cdr data. I compiled asterisk-addon module
without problems and I see nothing unusual in my cdr_mysql.conf but when
I do a reload I get this messages (never seen before):
Oct 16 09:43:16 WARNING[8576]: db.c:67 dbinit: Unable to open Asterisk
database
Oct 16
Giorgio Incantalupo wrote:
Hi,
I'm using mysql to store my cdr data. I compiled asterisk-addon module
without problems and I see nothing unusual in my cdr_mysql.conf but
when I do a reload I get this messages (never seen before):
Oct 16 09:43:16 WARNING[8576]: db.c:67 dbinit: Unable to open
Hi all,
I just configured a quescom 400 to route all gsm incoming calls to
asterisk, now i would route all outgoing asterisk calls to gsm port of
the quescom.
Anyone has any idea how implement it?
I did a configuration but i always get this error
-- Got SIP response 503 Service Unavailable
Anyone have a sane rc script for FOP on CentOS/RHEL systems?
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To UNSUBSCRIBE or update options visit:
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On 2006-10-15 23:50:34 -0700, Tim Panton [EMAIL PROTECTED] said:
On 16 Oct 2006, at 07:15, Martin Joseph wrote:
On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said:
On 11 Oct 2006, at 19:35, Dean Collins wrote:
Lol - use a real PC maybe :P
Nah, that would be dull.
In some
I know it won't help much but we use now a bristuffed Asterisk along with Snom 320 phones.It works now most of the time but we had to patch Asterisk keep calls from being cut (5% of calls were hit by that - symptom is voice cut in the middle of call).
Now we still have calls being hanged while
Hi Benjamin,
I checked in every place and it seems all right. The strangest thing is
that _Asterisk is writing CDR records inside the right table_that's
why I do not understand this message. I would expect Asterisk not to
fill in the DB.
Can I ignore the warning?
TIA
Giorgio
On Sun, Oct 15, 2006 at 11:15:55PM -0700, Martin Joseph wrote:
On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said:
On 11 Oct 2006, at 19:35, Dean Collins wrote:
Lol - use a real PC maybe :P
Nah, that would be dull.
In some ways the mac intel is nearer to a 'normal PC'
Hi all,
I just configured a quescom 400 to route all gsm incoming calls to
asterisk, now i would route all outgoing asterisk calls to gsm port of
the quescom.
Anyone has any idea how implement it?
I did a configuration but i always get this error
On the quescom, under the objects section,
I love TrixBox, with the custom config
files you can tweak pretty much with TrixBox too, I have at least done some. Plan
to do a plain Asterisk install later, but for now I learn a lot about the
config files just with TrixBox. Some things might be a bit harder with TrixBox
due to some of
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
is it possible on asterisk to reject unauthenticated calls or not registered
phones to call?
You can send them to [default] context that has only extensions like this:
exten = i,1,Hangup
exten = s,1,Hangup
--
Tomislav Parčina
Lama
Mike Garey wrote:
I've been noticing that my group of Polycom IP 501 phones seems to
randomly reset themselves nearly every night (I guess it usually
happens at night, since I've never seen it happen while I've been at
work during the day)..
When I say reset, I mean, the hands free volume
hi
i've got this problem:
queue A (ringall strategy)
- sip/200
- sip/201
- sip/202
suppose that sip/200 is busy and a call is received, 201 and 202 start
ringing. After some seconds 200 becomes free but 201 and 202 are still
ringing and 200 not!
where am i wrong?
i need that when 200 becomes
Hi,I am interested in test and work with your Reception appliation. Looking forward to your response. Thank you.Regards,Chaandra.Peter Lindquist [EMAIL PROTECTED] wrote: Sure thing, count me inPaul Hales wrote: We are currently writing a reception console for Asterisk - if anyone is interested in
Hi asterisk-users,
We are using Asterisk 1.2.12.1, and are trying to use the Page
application. It seems to work but after approx 4-5 seconds the call is
hung up.
The dialplan code look like this:
exten = _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2})
exten = _*2XX,n,GotoIf($[
hi
i've got a production system running asterisk-1.2.4/ with
zaptel-1.2.4/ using a beronet Beronet BN8S0 and TE205P .
at the moment (fortunately) i'm not experiencing any kind of
particular problem, do you suggest me to upgrade asterisk?
and zaptel?
and misdn?
thanks
Paul,
I would love to test it out in a busy environment. I am sure I can
provide quite alot of feedback from a real receptionist
Thanks,
Steve Totaro
Crazy Boy wrote:
Hi,
I am interested in test and work with your Reception appliation.
Looking forward to your response. Thank you.
Mr. Jones wrote:
I'm having problems with conference calls (3-way) when I have my codec
forced to g729 in sip.conf.
I'm using Grandstream 2000s.
If enable both g711 and g729 then 3 way calling and transfers work.
I'm not sure why this would matter?
Here's the error:
Oct 13 13:54:45
How do u call the quescom? With Dial() command?
exten = s,1,Dial(SIP/172.30.1.199:1123/${ARG2},Tt)
Did u set any port, or just call the ip address witout 1123 port ?
Thanks in advance
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di [EMAIL PROTECTED]
On 16 Oct 2006, at 09:09, Martin Joseph wrote:
On 2006-10-15 23:50:34 -0700, Tim Panton [EMAIL PROTECTED] said:
On 16 Oct 2006, at 07:15, Martin Joseph wrote:
On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said:
On 11 Oct 2006, at 19:35, Dean Collins wrote:
Lol - use a real PC
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Does anyone know what triggers the 7970 to update its config? I
was able to get it to update to SIP, but the config I used initially
won't go away. I am making small changes to the SEPxxx.cnf.xml file and
rebooting the phone, the
Hello,
I would like to create some form of reporting of call quality. Is
there a way to collect quality of RTP data (for SIP calls) to gather
some statistics (packet loss, ...). I would like to know when calls
are of lower quality and if I should blame ISP, operator or look for
some problems
I use myphonecompany.com. They have DID's for $5.00
a month and they 'let you' use 2 channels for per did (you can use more but they
dont like it if you abuse it). I had a client that needed 4 concurent channels
so they told him to just purchase 2 did's. So if you need 8 concurent incoming
at the moment (fortunately) i'm not experiencing any kind of
particular problem, do you suggest me to upgrade asterisk?
#1 sysadmin rule:
If it's not broken, just don't fix it.
___
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On Mon, 2006-10-16 at 13:13 +0200, Simone Ruffilli wrote:
at the moment (fortunately) i'm not experiencing any kind of
particular problem, do you suggest me to upgrade asterisk?
#1 sysadmin rule:
If it's not broken, just don't fix it.
That will get you into trouble when it _does_ break.
I
Hi Brian, hi list,
Brian Candler wrote:
On Fri, Oct 13, 2006 at 01:35:04AM +0200, Norbert Zawodsky wrote:
I've set canreinvite=no on the channel to the SIP provider and it
immediately worked. O.k., I'm happy about that but I want to
*understand* what's going on here.
.
My setup
I turned on sip debugging and noted folowing differences in the output
(1st='8904676', 2nd='890467610'):
1st: INVITE sip:s at 81.223.241.115 SIP/2.0
2nd: INVITE sip:01890467610 at 81.223.241.115 SIP/2.0
1st: To: sip:8904676 at p1.voip.inode.at
2nd: To: sip:890467610 at p1.voip.inode.at
This is really doing my head in!
For some reason, my asterisk box can't playback beep.wav.
I have this extension defined in my internal context:
'10001' =1. Answer() [pbx_config]
2. Wait(2)
Hi all
I share my Asterisk Server with a few friends. It is connected to PSTN, and
various SIP Providers.
I offer Free Calls to my friends, but myself I would like to be able to make
calls to non free destinations via my PSTN Line.
Now I do this in my dialplan:
---
When I try to upgrade 7970 phone to sip 8.0.4SR1, Im getting this error
all time:
Read request for file .loads. Mode octet [16/10 15:14:12.187]
File .loads : error 2 in system call CreateFile The system cannot find
the file specified. [16/10 15:14:12.187]
But I found this inside
Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686
running Linux on 2006-06-17
When I used monitor, I seem to get most calls cut off if they run
very long. Sometimes two minutes, sometimes 5 or 15.. Seems random. Any
ideas what might kill the recording process? I'm
Hello Paul
Yes, I very interesting
Viktor Tatianin
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Monday, October 16, 2006 7:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users]
I see, thank you very much for all your answers. Btw, the interface looks different than the ordinary rj45, so how are you going to plug in the rj45 plug to it?On 10/16/06,
George Pajari [EMAIL PROTECTED] wrote:
The TDM2400P supports up to six quad modules -- each quad modulesupports EITHER four
Matthew Fredrickson schrieb:
On Oct 12, 2006, at 1:17 PM, Johann Steinwendtner wrote:
Hello !
I 've some questions how bridging of ISDN calls is done.
Assume an asterisk system with a TE405 card equipped.
(PRI1 - PRI4)
An incoming ISDN call on PRI1 is transfered back to
PRI3. Unless there is
Viktor Tatianin wrote:
Hello Paul
Yes, I very interesting
Hi
We have MS Windows based operator consol/ panel available :)
http://www.bicomsystems.com/products/C/P/319/154_2571/#
Senad
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Hi Lito,
you need a particular cable to connect TDM2400 (which has 1 big port) to
a patch panel. Try Google on internet for a retailer.
Giorgio Incantalupo
Lito Lampitoc wrote:
I see, thank you very much for all your answers. Btw, the interface
looks different than the ordinary rj45, so
Can anybody that is currently using a Tellabs 2572 E.C. with a PRI/ISDN
with success, please let me know how they have the card (Wiring and
Settings) setup. I still have random local echo on our PRI.
Thanks,
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase
Brian Candler wrote:
On Fri, Oct 13, 2006 at 07:00:54PM -0500, Eric ManxPower Wieling wrote:
* Phones = stations, regardless of where they are
Asterisk = SIP Server, Phone = SIP Client
* Trunks = trunks to other SIP servers, bilateral
Asterisk and the other server is peer to peer
*
Hello Users,I doing on Voicemail in Asterisk For my RealTime, By using the ODBC connectivity For Voicemessages.in Made the Change in res_odbc.conf,odbc.ini, odbcinst.ini and voicemail.conf
When I start My Asterisk server it give me Some Warning,When I googled , a proper Docummentation is not
Secure multi-tenant partitioning capabilities?
What is your distribution intentions, commercial or GPL?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Sunday, October 15, 2006 10:33 PM
To: Asterisk Users Mailing List
Hi all,
We are getting ready to release our Call Control Gateway application
which allows for both remote phone control and PC to phone integration
between LCS and an Asterisk PBX. The gateway is scheduled to be released
in the beginning of Nov. Currently we are looking for Beta Testers that
Thanks!But i've solved my problem only using g(#) gain argument from voicemail application! For me was enough.Voicemail([EMAIL PROTECTED],b,g(10)) ; where 10 is the gain in dBthks guys for all your replies
On 10/16/06, kjcsb [EMAIL PROTECTED] wrote:
The
problem is:Right now, and i'm
Error syntax:is Voicemail([EMAIL PROTECTED],bg(10)) ; for busy announce and 10dB record gainOn 10/16/06, Marco Mouta
[EMAIL PROTECTED] wrote:Thanks!But i've solved my problem only using g(#) gain argument from voicemail application! For me was enough.
Voicemail([EMAIL PROTECTED],b,g(10)) ; where
The cable is an Amphenol Cable. This may help some.On Oct 16, 2006, at 8:34 AM, Giorgio Incantalupo wrote:Hi Lito,you need a particular cable to connect TDM2400 (which has 1 big port) to a patch panel. Try Google on internet for a retailer.Giorgio IncantalupoLito Lampitoc wrote: I see, thank you
On Mon, Oct 16, 2006 at 02:08:05PM +0200, Benoit Panizzon wrote:
[myself]
; National Destinations
exten = _0z.,1,Dial(SIP/someisp/${EXTEN});
exten = _0z.,n,Dial(Zap/g1/${EXTEN});
; International Destinations
exten = _00z.,1,Dial(SIP/someisp/${EXTEN});
exten =
Richard G. Cavanna
Information Technology Manager
SyChip Inc.
P - 972.202.8840
F - 972.633.0327
You can buy a pre made breakout box or go directly to a patch panel.
I have used this one form VoIP supply with success
http://www.voipsupply.com/product_info.php?products_id=1164searchid=111
839
Hi,
every second I get on the console:
Remote UNIX connection
Remote UNIX disconnected
which gives no problem but makes console unusable.
Is there anybody who has encountered the same problem? How did you solve it?
TIA
Giorgio Incantalupo
___
I have a bata site we can use to test your software.
Please contact me [EMAIL PROTECTED]
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Secure multi-tenant partitioning capabilities?
What is your distribution intentions, commercial or GPL?
-Original Message-
This:
http://en.wikipedia.org/wiki/RJ-21
and this:
http://en.wikipedia.org/wiki/66_block
will get you there.
On 10/16/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi Lito,
you need a particular cable to connect TDM2400 (which has 1 big port) to
a patch panel. Try Google on internet for a
I know this question has been asked a great deal, but
does any1 have a simple way Of getting video to work
using this particular channel...
Or at least is it possible just using the conf files, or do I
Have to have a separate decoder to encode the video
Thanks again
I am currently running 1.2.7.1 and it works just fine. I personally like to
stay 3 or 4 months behind the current release. This time it is a bit longer
because I don't feel comfortable with the stability of later releases.
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I know this question has been asked a great deal, but
does any1 have a simple way Of getting video to work using this particular
channel...
Or at least is it possible just using the conf files, or
do I Have to have a separate decoder to encode the video
Thanks again
More info...
All calls come in from a Tekelec-7000/r4.0.
The box has 2 te410p's left over from when calls came in from PRI. They
were left in for a timing source since I don't have physical access.
On Fri, 13 Oct 2006, Steve Edwards wrote:
In the past, there have been reports of problems
We deployed a PABX in China, orginally it used Netcom（网通）'s E1, the
zaptel.conf is as following:
span=1,0,0,ccs,hdb3,crc4
bchan=1-15
bchan=17-31
dchan=16
loadzone=cn
defaultzone=cn
However, recently customer changed to use China Telecom（中国电信）'s E1, it
always show REC, RED/REC, RED, cycling
Is there some way I can tell?
On 10/16/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mr. Jones wrote:
I'm having problems with conference calls (3-way) when I have my codec
forced to g729 in sip.conf.
I'm using Grandstream 2000s.
If enable both g711 and g729 then 3 way calling and transfers
Hi.
I'm running some asterisk boxes on different sites,
some equipped with a couple of ZapHFC cards, others with
Junghanns quadBRI cards.
All boxes were compiled with Asterisk 1.2.10 (libpri 1.2.3 zaptel 1.2.6)
and bristuff 0.3.0 pre 1s, distribution is Fedora Core 4 with
kernel 2.6.17.3
The
On Mon, Oct 16, 2006 at 11:27:51AM -0400, C F wrote:
This:
http://en.wikipedia.org/wiki/RJ-21
and this:
http://en.wikipedia.org/wiki/66_block
will get you there.
If the TDM card's connector is actually an Amphenol 50 (which *just*
fits into a card bracket hole, IIRC) and it's actually wired
Like the one that comes with it?
[EMAIL PROTECTED] ~]$ sudo more /etc/init.d/op_panel
#!/bin/bash
#
# chkconfig: 2345 99 15
# description: Flash Operator Panel
# processname: op_server.pl
# source function library
. /etc/rc.d/init.d/functions
DAEMON=/usr/local/op_panel/op_server.pl
OPTIONS=-d
On most traditional pabx's it's possible to set layer 1 to permanent or
call. It sounds like your system is configured for permanent and your lines
to call. How you would set this on asterisk I have no idea.
fadge
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hello Chandra,
What about Teliax´s service? Is it recommended? How´s their call quality? Thanks in advance...
On 10/10/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi William,My DID is working and am receiving calls. The problem is with Teliax settings from their end. Thank you for spending your
Hello.
I had the same problem, and was able to fix it as
follows.
1. Run fxotune
2. Call your XO rep and get a milliwatt test line
number
3. set the gain in the zaptel.conf incoming with the
milliwatt test line
4. loop a call through the pbx and set the outgoing
gain.
Withthese setup
On Mon, Oct 16, 2006 at 04:47:31PM +0200, Giorgio Incantalupo wrote:
Hi,
every second I get on the console:
Remote UNIX connection
Remote UNIX disconnected
which gives no problem but makes console unusable.
Is there anybody who has encountered the same problem? How did you solve it?
Have
Hi,
every second I get on the console:
Remote UNIX connection
Remote UNIX disconnected
which gives no problem but makes console unusable.
Is there anybody who has encountered the same problem? How did you solve it?
You probably have some script that use the console to query something,
like the
In my IAX config file I have:
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
delayreject=yes
disallow=all
allow=ulaw
allow=gsm
jitterbuffer=yes
forcejitterbuffer=yes
mailboxdetail=yes
dropcount=3
I concur with Conrad. As I understand it, as long as you stick with
1.2.x versions, there should be no new 'features' to worry about
implementing, only bugfixes. So I'd recommend keeping up with them, and
the 'upgrade' should go smoothly because it's generally not too much of
an upgrade.
Why is it running on port 1207?
because Asterisk is listening on port 4569 and when a connection comes
in, it as handed to another port so it can continue listening on port
4569. Otherwise you would only be handling 1 connection at a time.
Pretty basic networking stuff I think :c)
On 10/16/06, Time Bandit [EMAIL PROTECTED] wrote:
Why is it running on port 1207?
because Asterisk is listening on port 4569 and when a connection comes
in, it as handed to another port so it can continue listening on port
4569. Otherwise you would only be handling 1 connection at a time.
Ciao Giorgio,
I'm using mysql to store my cdr data. I compiled asterisk-addon
module without problems and I see nothing unusual in my
cdr_mysql.conf but when I do a reload I get this messages (never seen
before):
Oct 16 09:43:16 WARNING[8576]: db.c:67 dbinit: Unable to open
Asterisk
I don't think this is a problem because of the snow storm.
I just got off the phone with them. The sales guy I used to deal with left a
few months back and since then, its been a pain to get anything done with
them. People I have dealt with had no clue.
I called them this morning for a problem
Hi,
I have set up a fax to email capability on our Asterisk Server (which is used for voicemail, and fax2email only),
and would like to improve it !
Everytime Asterisk receives a fax, I would like it to go and search in a DB if the Extension is defined.
If it is, then it will set the Email
Hi, ( Sorry for previous post, it was incomplete :o(
I have set up a fax to email capability on our Asterisk Server (which is used for voicemail, and fax2email only),
and would like to improve it !
Everytime Asterisk receives a fax, I would like it to go and search in a DB if the Extension is
Do me a favor and try running netstat -aplntu | grep asterisk and see
what ports are actually being used. Are you connected to another ITSP? If
so then that may be the local port of that connection... just an idea, i
don't have Asterisk access right now to double check.
Ryan
On 10/16/06, Time
On Monday 16 October 2006 16:15, Matt wrote:
Thanks for the answer, but I don't buy it. There are currently 0
Whether you buy it or not is irrelevant. That is the port that this asterisk
box is seeing the other one up on. It is seeing it that way (most likely)
due to NAT between the two
On 16 Oct 2006, at 20:43, Matt wrote:
In my IAX config file I have:
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
delayreject=yes
disallow=all
allow=ulaw
allow=gsm
jitterbuffer=yes
forcejitterbuffer=yes
Ciao Jean-Marc,
Everytime Asterisk receives a fax, I would like it to go and search
in a DB if the Extension is defined.
If it is, then it will set the Email address where to send the fax.
You can use app_addon_mysql for your purposes.
See:
Dear VaibhaV,You can purchase this part from pretty much any certified Polycom reseller.For the IP 30x/50x you would want the Mfg Part Number 2200-07496-001For the IP 430/60x you would want the Mfg Part Number 2200-17492-101We among many other certified resellers sell this part.Being a reseller
Thanks for the answer, but I don't buy it. There are currently 0
calls up on that bridge, while another connection which has calls up
on it is on Port 4569.. please try again. IAX2 is suppose to run on
ONLY one port.. this is why it is so nice for use in firewall
situations.
It doesn't change
Andrew,
I totally buy YOUR explination and that is what I think is happening..
the NAT box on the far end (not ours) is changing the port.
My question is... if both machiens are set to listen on 4569, will the
fact that that router is mangeling the port cause any issues?
-- Forwarded
On 16/10/06, Simone Ruffilli [EMAIL PROTECTED] wrote:
at the moment (fortunately) i'm not experiencing any kind of
particular problem, do you suggest me to upgrade asterisk?
#1 sysadmin rule:
If it's not broken, just don't fix it.
Slightly older and wiser sysadmins consider the importance
Thanks,
it seems to be not easy to use, but ... should do what's needed !
Thanks.
On 10/16/06, Andrea Spadaccini [EMAIL PROTECTED] wrote:
Ciao Jean-Marc, Everytime Asterisk receives a fax, I would like it to go and search in a DB if the Extension is defined.
If it is, then it will set the Email
I am getting ready to image a production system. Right now I am planning on
using Centos 4.4, Asterisk 1.2.12.1, Freepbx 2.1.3. I will be using a
Sangoma A200D card.
I read of some people having problems with Asterisk 1.2.12.1 crashing. Is
this across the board or is there anyone out there
Or try WalMart. Just make ABSOLUTELY CERTAIN that you
use the correct voltage and polarity. Also make
certain that the current rating is adequate.
At 16:16 10/16/2006, Jessee J Holmes wrote:
Dear VaibhaV,
You can purchase this part from pretty much any certified Polycom reseller.
For the IP
OMG, please read more about network ports.
:c)
MM
-Original Message-
From: Time Bandit [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Mon, 16 Oct 2006 17:25:22 -0400
Delivered:
On 10/16/06, Melcon Moraes [EMAIL PROTECTED] wrote:
OMG, please read more about network ports.
Could you tell me what is wrong with my explanation ?
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Dear All,
I am trying to find a way to stop people who use phones after
business hours (a policy the company wants to implement), we have cisco
7940 and 7910 phones and sadly they don't have a phone lock password system
(on these ciscos it locks config menu changes but not the calls but the
I'm researching an asterisk implementation for a client. Originally, they
wanted a T1 (as other vendors had quoted such). Now tho, they are asking about
just doing VOIP, cause fortune 500's seem to be so successful at it.
That questionable assertion aside, I see there are a lot of outfits
Sure, in the context the phones live in, play around with the
GotoIfTime() application:
Completely pseudocoded, will not work without research:
[internal]
priority 1 : gotoiftime(8:00-17:00|mon-fri?priority 3)
priority 2 : goto 10
priority 3 : dial(out_trunk, ${EXTEN})
priority 4 : hangup
You're a little backwards. When you connect to a remote server via HTTP
protocol, for example, you ARE connected to their remote port 80. They
do not send data to YOUR port 80 though.
Moj
Time Bandit wrote:
On 10/16/06, Melcon Moraes [EMAIL PROTECTED] wrote:
OMG, please read more about
Dear Moj,
Thanks a lot fo the tip, it seems I can do that it is very flexible
and easy to use, I will try to add it to the trixbox files in a nice fashion
but that will be after I get some sleep ;-)
Thx
MAG
"Mojo with Horan Company, LLC" wrote:
Sure, in the context the phones live in, play
So I was wondering is there a way to make this happen in asterisk??
Depending on where you are located, you might want to allow emergency calls to go through. The bloodsuckers, I mean attorneys, here in the US would have a field day if something were to happen to someone at a company that did
Ok I understand all that... Just wanted to confirm that A) it was the
remote router mangeling the port and B) that it wouldn't cause an
issue (I wasn't 100% sure if it would.. since only the 4569 port is
open on the firewall).
Could this cause an issue? If only 4569 is open on the firewall, and
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