You will want to use the D(digitstopluseindtmf) option on your dial cmd.
That is a capital D for the option!
ex.
Dial(SIP/2100,D(1000))
-Jon
Stephen wrote:
Hi All,
Can Asterisk dial extension which resides in the PABX?
(eg. 2000) Sip Phone - Asterisk -- ATA (FXS) --
(CO
On Mon, Aug 15, 2005 at 07:54:50AM +0200, Ronald Voermans wrote:
Okay,
First of all, thank you for your input. I didn't know that I could use 1
* for multiple companies (wish I knew it earlier, because installing
vserver and installing * on a vserver took me a lot of time :) ).
Hi all,
I have a sirrix bri card, connected to 2 ISDN lines.I used to get a lot
of slips on it, so i put the 'master' setting to 'yes'. But every couple
of hours the machine completly hangs, and i have to reboot it. I only
get 1 or 2 slips,I dont know whats wrong?
yusuf
If I install 1 * server, with multiple companies/dialplans, how do I
make 1 company dial the other company with a full telephonenumber (i.e.
10 digits)?
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Tzafrir Cohen
Verzonden: maandag 15 augustus 2005
Hi,
I'm unable to load wct4xxp module for TE406P card.
I've compiled 2.6.13-rc6, got latest CVS (as of today) of zaptel, but
when I try to load the module I get this:
kobject_register failed for Unified t4xxp/t2xxp driver (-13)
[kobject_register+53/73] kobject_register+0x35/0x49
Hi List
Does anyone have the configs to connect 2 * servers so that clients from the
one * server can make pstn (ZAP) calls out from the other * server ?
Thanks in advance
Sean
___
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Sean Rima wrote:
Using the install instructions for [EMAIL PROTECTED], I setup a FWD account,
this I
tested using X-Lite and it works okay,
Nowever I cannot make calls to fwd using Asterisk, my log showes:
Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected:
Registration Refused
I have 2 OEM X100P. The one from www.broad-tel.com works fine.However,
the other one has echo. Both use MD3200 chips. Any one knows why it is
so??
On 8/13/05, Madhawa Jayanath [EMAIL PROTECTED] wrote:
Carlos Trallero wrote:
Hello,
I have asterisk running on Fedora Core 3 with a x100p
Hi!
When I want to monitor the OSS/dsp channel through the Asterisk management
interface, I get a permission denied error:
Action: Monitor
Monitor: OSS/dsp
File: 1124096949
Mix: 1
Response: Error
Message: Permission denied
My permissions for /var/spool/asterisk look like this:
pound:~# ls -la
On 8/14/05, Bjorn Ove Kristiansen [EMAIL PROTECTED] wrote:
Is there any way of getting to know which IP address Cisco uses to contact
TFTP?
Why you're making things hard for yourself for no good reason? Unlock
the config, put a static entry for ip that belongs to the segment its
sitting on
I've now setup SIP for:
- internal softphones
- registering with external providers (like FWD) for making calls
- receiving calls from theese providers
For the latter step, it was necessary to forward ports from my NAT
to the asterisk server: 5060 + range of ports mentioned in rtp.conf.
I was
John Fawcett wrote:
Sean Rima wrote:
Using the install instructions for [EMAIL PROTECTED], I setup a FWD account,
this I
tested using X-Lite and it works okay,
Nowever I cannot make calls to fwd using Asterisk, my log showes:
Aug 14 21:06:09 NOTICE[1324]: Registration of '689482'
Hello,
i have some problem with the Asterisk and need a little bit help.
I have an dialplan like this:
[first]
.
..
...
exten = s,6,Read(Secret,,25)
exten = s,7,NoOp(**${Secret}**)
exten =
Hi,
Ronald Voermans wrote:
If I install 1 * server, with multiple companies/dialplans, how do I
make 1 company dial the other company with a full telephonenumber (i.e.
10 digits)?
This is very much dependant on how your dialplan works. We use
normalisation for each account so the system
Chris Gamble schrieb:
I have 2 TDM04b cards currently running in an asterisk at home box that I am
ready to replace with the CVS version of asterisk. What I am looking for is
thoughts / recommendations. I want to move this to a small form factor (
shuttle ) machine and was wandering what
Of course you do. How would the * system know whom the call is for?
Mark
craz sead wrote:
Do i have to create voice mail one by one (per ext at
the meridian) at the * box ?
--- Mark Phillips [EMAIL PROTECTED] wrote:
Easily doable. I've done it twice now. Problem is
that your users will
Got the same thing, thanks Jeff.
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, August 14, 2005 12:00 PM
To: asterisk-users@lists.digium.com
Subject: Asterisk-Users Digest, Vol 13, Issue 98
Send
On Monday 15 August 2005 11:11, Christoph Eicke wrote:
Hi!
When I want to monitor the OSS/dsp channel through the Asterisk management
interface, I get a permission denied error:
Action: Monitor
Monitor: OSS/dsp
File: 1124096949
Mix: 1
Response: Error
Message: Permission denied
it's
All security issues aside here, this is very easy.
Create an IAX trunk between the 2 servers. Put the Zap side trunk into
the same context that is allowed to dial the Zap line.
On the non-Zap server create a dialplan rule that forwards all calls to
the PSTN over the the Zap host.
Why do we
Are these cards on the same line or are they on different lines.
Assuming different lines, what happens when you swap the lines over?
Does the echo follow the line?
Also, It seems that these cards can be modified to do a number of things
either in firmware or in hardware. Are they
You could make your FWD sonfigs even more secure by switching to IAX
(you have to register with them for it) and then you can use RSA keys
(already in your * distro) to prevent faking of connections.
Check with the FWD site. Ther's a howto on there.
I use this method and I like it alot.
To make sure no configuration issue, I only had one of them working at
a time. After test on one of them, I swapped out with the other and
rebooted. Both were detected as same. However, echo only happened on
one but not the other. I know nothing about electronics. Layout of
both cards looks almost
Here's their press release with the improvements for firmware v2:
http://www.digium.com/index.php?menu=press/pr_2gen_firm
MATT---
On 8/15/05, Adam Goryachev [EMAIL PROTECTED] wrote:
On Fri, 2005-08-12 at 08:29 -0500, Kevin P. Fleming wrote:
Matt Florell wrote:
We do 2nd gen firmware
We've deployed IP 300's, 500's, and 501's at customers and they work very
well.
On Thu, Aug 11, 2005 at 11:52:35AM -0700, Ing. Marlo R. Beltran G wrote:
I am about to buy ip pbx asterisk system but what ip phones do you
recommend? Are polycom ip all functional with the ip pbx system???
--
Hi all!
I'm getting an error when I try to start asterisk with chan_misdn.
I patched my kernel (2.6.12.4), and compiled the whole stuff (kernel,
mISDNuser, asterisk, chan_misdn). I got mISDN from
http://isdn.jolly.de/download/v3.0/
I'm using a CVS Snapshot of asterisk, which was checked out
I'm not sure I understand what you mean...
I want to have internal extensions (100, 101, 102, etc.) and some full
phone-numbers (10 digits). How do I implement this in *?
Ronald
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Florian Overkamp
Verzonden:
Based on research that I did some time ago, there are multiple versions
of the MD3200 chipset. One targeted for use in US telephone systems, and
another targeted for non-US systems (that have different impedence matching
requirements). Sounds like you have one of each.
I
hi,
i have this topology
pstn+(e1)asterisk1-asterisk2-sip client
asterisk1,asterisk2 allow (g729,alaw)
sip client prefer g729, then alaw
can you someone describe codec negotiation when call for sip client arrive
from pstn? (can i set g729 for calls from pstn? )
thanks
Brian Capouch wrote:
I'm trying to figure out how to do some things like round-robin server
balancing and the like using Realtime, and it seems like the right way
to do it would be either via pre-processing the SQL requests coming in,
or using stored procedures in the database that would
I am using the correct version of pwlib(1.5.2) and
openh323(1.12.2) for the stable asterisk build. Both packages configure and
compile with no problems. However
when compiling chan_h323 from the asterisksource/channels/h323 directory I get
this error.
Chan-h323.h:31: warning;
Ronald Voermans wrote:
I'm not sure I understand what you mean...
I want to have internal extensions (100, 101, 102, etc.) and some full
phone-numbers (10 digits). How do I implement this in *?
Ronald
Right. We have 3 contexts. 1 is for all incomming traffic from PRI or
other carrier. 1
you can find some resources at the end of this page:
http://www.voip-info.org/tiki-index.php?page=ISDN
hope it helps.
best regards
On 8/14/05, Vedran Dakic [EMAIL PROTECTED] wrote:
Hello there.
What I'm trying to make is - have an asterisk server with
sccp/mgcp/skip/h.323
Does anybody have any experience with setting up Asterisk with the
provider RocketVoip?
Thanks,
// Jonas
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I've been playing around with asterisk for the past few days. One thing
which came to mind
which could be a useful addition to the meetme functionality would be
the possibility to
specify a moderator password in meetme.conf. (A moderator in the sense
that music
is heard until the moderator
Block sip on a firewall between * and the public internet, and then
create rules for your peers IP range.
This assumes you know the IP that all peers and client use; if not just
block from regions of the world you do not need to connect to/from.
We find that most hack attempts come from one well
On Monday 15 Aug 2005 15:19, Tom Tobias wrote:
I am using the correct version of pwlib(1.5.2) and openh323(1.12.2) for the
stable asterisk build. Both packages configure and compile with no
problems. However when compiling chan_h323 from the
asterisksource/channels/h323 directory I get this
Hi, everybody.
I was just running into a DynDNS Problem.
One of our phones has a fixed IP, but the * box only has a dynDNS IP and a
fixed name. So all phones running SIP are able to survive the change of
the IP address, but not our Cisco 7920 - it changes state to connecting to
CM0 and hangs
Armin Lediger wrote:
Hi, everybody.
I was just running into a DynDNS Problem.
One of our phones has a fixed IP, but the * box only has a dynDNS IP and a
fixed name. So all phones running SIP are able to survive the change of
the IP address, but not our Cisco 7920 - it changes state to
Hi all,
I have several queues defined in Asterisk, but when I put a user in two
queues, the user (can) get two calls at once. (One from each queue).
Right now, I configured some phones to only accept one call at once, but
that has the side-effect, that also direct calls won't come through.
Is
Hello, thanks for the replies so far.
In regards to the below quoted answer: If the phone locks up before SIP
firmware has booted, then there's absolutely no way to set these settings
manually from the phone itself. The situation is simply that I have no idea
which tftp settings that are already
Hello!
The issue is simply that I don't know which IP address the phone tries to
connect to. I am not very familiar with dhcpd (never put it up by hand), so
I'm not sure how the below would help me, but from what I can tell, I still
need information on which IP-address the phone is trying to find
Search google with sip pstn site:www.microsoft.com
You will find out how to configure LCS static routing to SIP Gateway,
like Asterisk
but you need patch Asterisk to support TCP.
http://bugs.digium.com/view.php?id=4903
Step1: configure LCS 2005 to let sip uri: [EMAIL PROTECTED] to route to
next
Register with two different accounts from the same phone. One for Queue
Calls, and one for direct calls.
Greetz,
Zoa.
---
Asterisk tutorials: http://www.asteriskguru.com
---
Leon de Rooij wrote:
Hi all,
I have several queues defined in Asterisk, but when I put a user in two
queues, the
On Mon, 2005-08-15 at 17:28 +0200, Bjørn Ove Kristiansen wrote:
Hello!
The issue is simply that I don't know which IP address the phone tries to
connect to. I am not very familiar with dhcpd (never put it up by hand), so
I'm not sure how the below would help me, but from what I can tell, I
A red alarm means I don't see any signal. A blue alarm means I
see a signal and something downstream (repeater etc) is saying they
don't see
a signal.
I used this as my reference:
http://www.fratec.com/FAQ/NFO/NFO_WAN_009.HTML
Slips sometimes cause an LOF condition, sometimes they
iam new to h323 ,i just download it and compile and it
gives me these messages on console :
Sep 13 00:20:30 WARNING[15443]: chan_oh323.c:4014
oh323_gk_check: Gatekeeper discovery failed.
-- Retrying gatekeeper registration.
does any body knows whats going on.
I got one from www.broad-tel.com. It works fine.
On 8/12/05, Douglas Logan [EMAIL PROTECTED] wrote:
Yes, but your results may vary. Apparently some people have problems
with clone cards (aka regular modems), dropping calls, and having
echos. (Then again some people have reported no problems at
Hi,
Hmm.. didn't think of that.. Thanks for the tip..
Though it wouldn't really work as it's not possible to set CWI (Call
Waiting Indication) seperately for each logged-in user.. (We currently
use SNOM190 and Cisco7960 phones)..
Also it would change our entire system (database, webinterface and
ONLY ON MONDAY!
Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it
On 09:38, Mon 15 Aug 05, kurt turner wrote:
ONLY ON MONDAY!
Well it used to work - calls between my aaln's that is. I moved from debain
to redhat (same conf. files for asterisk) and this is what I get.. looks like
several errors. errors I never got before. Also asterisk isn't observing
Hi,
asterisk will negotiate codecs for both parties independently (use sip
show peer peer and look for codec order entry), so, if you have
prefered codec g729 for your sip phone/peer, asterisk will use them
(regardles of codec setting for other party - if codecs does not match,
asterisk will
no translator path = no codec...
Have you called Digium to get your G729 (format 256)codecs released and
re-registered them to the new(ly configured) box???
At the cli, type show g729... does it give an error or show g729
information?
regards,
Derek
- Original Message -
From: Michiel
Pavel Jezek wrote:
Hi,
asterisk will negotiate codecs for both parties independently (use sip
show peer peer and look for codec order entry), so, if you have
prefered codec g729 for your sip phone/peer, asterisk will use them
(regardles of codec setting for other party - if codecs does not
Hello,
I have two FXS port on my TDM card.
channel 4 is attached with a telco line that I use frequently. And channel 3
have another telco line. but I dont publish that number to my friends.
If I receive a call through channel 4, how can I handover that call to
channel 3 ..so that I can keep
Innocent Evil wrote:
Hello,
I have two FXS port on my TDM card.
channel 4 is attached with a telco line that I use frequently. And channel 3
have another telco line. but I dont publish that number to my friends.
If I receive a call through channel 4, how can I handover that call to
channel 3
Hello All,
Has anyone configured bri to answer for only one msn? In essence, when
the primary is busy I want to have channel 2 ring.
I am using an eicon diva server bri
I know I saw it in the windows interface, but don't see it in the linux
setup.
Regards,
Greg
As far as I remember, you can't really do that (because the telco isn't
switching the call), what you'll want to do is have a hunt group set up
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Innocent Evil
-Sent: Monday, August 15, 2005 2:17 PM
Chris Wade wrote:
Innocent Evil wrote:
Hello,
I have two FXS port on my TDM card.
channel 4 is attached with a telco line that I use frequently. And
channel 3
have another telco line. but I dont publish that number to my friends.
If I receive a call through channel 4, how can I handover
Innocent Evil wrote:
Hello,
I have two FXS port on my TDM card.
channel 4 is attached with a telco line that I use frequently. And channel 3
have another telco line. but I dont publish that number to my friends.
If I receive a call through channel 4, how can I handover that call to
channel 3
Sorry for the typo.
Do I need to ask my telco, if I want to use Asterisk as a PBX in a
home/small biz/large biz and I want one hunting number.
Thanks,
-Original Message-
From: [EMAIL PROTECTED]
Sent: Mon, 15 Aug 2005 13:20:17 -0500
To: asterisk-users@lists.digium.com
Subject: Re:
On Mon, 2005-08-15 at 13:20 -0500, Chris Wade wrote:
First, FXS = handset / FXO = telco line.
Ditto this.
Maybe something like fax-callback; call-in, hangup, Asterisk dials back
on the other channel using the CID received - a purely physical
solution. Otherwise, have the telco setup a rotary
As far as I remember, you can't really do that (because the telco isn't
switching the call), what you'll want to do is have a hunt group set up
Yesss... this is exactly I am looking for.
How can I do that?
Thanks,
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL
Yes, you need to ask the telco to autoforward your chan 4 num to chan 3
(called hunt grouping), there may be a fee. Also not sure if that's
available for a standard residential line (or just POTS in general). You
don't need to tell them why, just tell 'em you want it. No need to confuse
'em.
I have two FXS port on my TDM card.
channel 4 is attached with a telco line that I use frequently. And channel 3
have another telco line. but I dont publish that number to my friends.
If I receive a call through channel 4, how can I handover that call to
channel 3 ..so that I can keep
I am getting this whenever I start asterisk.
Aug 15 15:03:58 WARNING[11754] chan_oss.c: Read error on sound device:
Resource temporarily unavailable
This is my sound card:
Multimedia audio controller: Fortemedia, Inc Xwave QS3000A
I am not sure... what I am doing wrong.
Please help.
I think this email got mixed with other emails thks.
Hi all,
In this moment I have the opportunity to install asterisk in Poweredge 1400
Dell Server (PIII, 2 GB of RAM). I wonder if any of you have any experience
running asterisk (+ Digium cards) on this kind of hardware, any comment
I am clear with this issue.
Thanks everybody for answering me.
-Original Message-
From: [EMAIL PROTECTED]
Sent: Mon, 15 Aug 2005 10:16:34 -0800
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Switch between FXS ports
Hello,
I have two FXS port on my TDM card.
The Call Forward On Busy does cost YOU money each time you forward a call.
Call hunting group is different from Call forwarding.
In a hunt group you have 2 or more phone lines grouped together. When a
call for a number associated with the group comes into the telco switch,
the switch checks
Alejandro...
Go search the archive... There are tons of posts regarding Dell equipment
Here is how to do so if you do not know...
Go to www.google.com
Enter the following...
site:lists.digium.com Dell Poweredge
Thanks,
W
-Original Message-
From: [EMAIL PROTECTED]
On Fri, Aug 12, 2005 at 09:03:42PM -0500, Tim Connolly wrote:
I'm having lots of stability problems with my 411's. I'm not blaming the 411
yet, just seems odd that I ran for months on a TE110P with a peak of 10-15
calls, and now my box kernel panics each time it hits the same load.
Granted,
Hello everyone,
Does anyone have experience with echo calibration for TDM card with
rxgain and txgain (Zapata.conf) and ztmonitor (CVS Head)?
I have found very few information about it and what I have found makes
me confused. I have a phone number provided by my TelCo(1004 hz at 0db)
and from
The Call Forward On Busy does cost YOU money each time you forward a call.
No, that is telco dependent. Most US telco's do not charge for that
as long as the forwarded number is a local number. If its not, then
LD charges apply. But in some US cities, you are correct that an
additional change
I have a user who has a fax machine plugged into an ATA.
They are able to SEND faxes just fine. Faxes go through wonderfully.
However, when someone tries to send them a fax, their fax machine
never receives it. And eventually the sending machine just errors
out. Any thoughts?
On 8/15/05, Ken Dresdell [EMAIL PROTECTED] wrote:
Hello everyone,
Does anyone have experience with echo calibration for TDM card with
rxgain and txgain (Zapata.conf) and ztmonitor (CVS Head)?
I have found very few information about it and what I have found makes
me confused. I have a
On Mon, 15 Aug 2005 [EMAIL PROTECTED] wrote:
Hello All,
Has anyone configured bri to answer for only one msn? In essence, when
the primary is busy I want to have channel 2 ring.
I am using an eicon diva server bri
I know I saw it in the windows interface, but don't see it in the linux
Now that I've looked back over my work for the past few days I realize that
I was trying to play with the txgain/rxgain to adjust the levels and hope to
smooth out the line noise. Well, any integer other than zero for either of
those values causes BLUE alarms and all the channels to reset in
On Mon, August 15, 2005 3:50 pm, Rich Adamson said:
That's exactly what I do with our business line. Call Forward on Busy is
a common description for that telco service. (I simply forward that next
call to an unlisted/unpublished number which also terminates in Asterisk.)
In my very limited
Sean Rima wrote:
Using the install instructions for [EMAIL PROTECTED], I setup a FWD account,
this I
tested using X-Lite and it works okay,
Nowever I cannot make calls to fwd using Asterisk, my log showes:
Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected:
Registration
I have newly installed a RedHat 4.0 EL rebuild. The install was done
without the ISDN card present.
After disabling kudzu and haldaemon I inserted the card.
Stil that *($^%$($^!! kudzu shit modified my config and is loading
hisax, crc_ccit and isn modules.
Even worse, they do not appear
Typically a hardware firewall is specialized and uses ASICs. Because
the solution utilizes specialized chips tailored to the task, this is
considered a hardware based solution. Of course software is involved
but it too is specialized and is even proprietary in nature.
A software firewall, be it
We are using the following to record conversations.
exten = _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP})
exten = _1XXX.,2,Monitor(wav,${CALLFILENAME},m)
exten = _1XXX.,3,Dial(IAX2/4506:[EMAIL PROTECTED]/${EXTEN:1})
exten = _1XXX.,4,Congestion
exten = _1XXX.,104,Congestion
Do you mean this occurs when traffic is passed over an IPSec tunnel or
that it occurs anytime a tunnel is use on a machine that also is passing
VoIP traffic (outside the tunnel)?
I assume you must mean over the tunnel but I am curious...
Thanks,
Wiley
-Original Message-
From: [EMAIL
Try reinstalling sox - it is responsible for mixing the caller and
callee channels. Also, if IAX2/4506:[EMAIL PROTECTED] is your
real username and password, change them asap, you just made it available
to 1+ people and the archives ;)
Regards,
Vahan
Eric Smith wrote:
We are using the
On Mon, Aug 15, 2005 at 10:23:25PM +0200, Remco Barende wrote:
I have newly installed a RedHat 4.0 EL rebuild. The install was done
without the ISDN card present.
After disabling kudzu and haldaemon I inserted the card.
Stil that *($^%$($^!! kudzu shit modified my config and is
Hi list!
On a newly installed RHEL 4 box I'm trying to install bristuff-0.2.0-RC8n.
Everything did compile but I am running into some problems with the zaphfc
driver.
First of all when I load zaphfc *before* zaptel (yes I know I shouldn't do
that) I get a kernel panic and the box hangs. Not
Hello everyone,
I want to build an Asterisk Box where i need 8 FXS interfaces
to connect 8 phones to. The problem is, that there is only one
PCI slot available. What i have is 4 USBs 2.0 interfaces free
(if this helps).
So here's my question: how am i going to do this?
i tried to find any PCI
On Mon, 2005-08-15 at 23:32 +0200, Roland Zagler wrote:
Hello everyone,
I want to build an Asterisk Box where i need 8 FXS interfaces
to connect 8 phones to. The problem is, that there is only one
PCI slot available. What i have is 4 USBs 2.0 interfaces free
(if this helps).
So here's my
I have been doing a bit of this too lately. This was also useful.
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html
Dan
What about for PRI lines? We get echo every now and then. The docs link
above references FXO lines. We have none. But we do
Hello,
What's dnsmgr ?
Anybody could tell mr more?
cat /etc/asterisk/dnsmgr.conf
[general]
;enable=yes ; enable creation of managed
DNS lookups
; default is 'no'
;refreshinterval=1200 ; refresh managed DNS lookups
every n seconds
;
This service has been working well lately, but as of this morning is promptly
blowing off IAX connections with the dreaded 'No Authority Found' error.
Any concrete info greatly appreciated!
Dr G
___
Asterisk-Users mailing list
As of 22:45 GMT it's working for me
Jerry Glomph Black wrote:
This service has been working well lately, but as of this morning is
promptly blowing off IAX connections with the dreaded 'No Authority
Found' error.
Any concrete info greatly appreciated!
Dr G
Joseph:
Thank you for the help.
Orlando
From: Joseph [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users]
Easy and cheap.
Get two gateways AG-468 (each have 4 FXS ports) made by Atcom
http://www.voip-info.org/tiki-index.php?page=Atcom
one is about 88/ea
I have two on the way and will let you know how it works.
--
#Joseph
On Mon, 2005-08-15 at 23:32 +0200, Roland Zagler wrote:
Hello everyone,
Thanks for the hint, where have you bought them?
Roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Tuesday, August 16, 2005 12:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 8 FXS in
Here is my setup:
* is on a NAT'd subnet, but also has an externally routable IP address.
I have a Snom360 that's external to this and behind NAT.
The Snom360 can call other phones in * subnet (by their internal extension
numbers) and voice is transmitted fine; however, when I attempt to check
Hi All.
We have 2 clone x100ps and they work well but we cant
get callerID working, they should work right out of the box so if anyone in NZ
has a working callerID setup if they could send me the Zapata.conf config that
would be great
Cheers
Tristram
For me to
- Original Message -
From: Mat Stace, Colewood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, August 15, 2005 5:46 PM
Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
On Mon, 2005-08-15 at 12:40 -0700, Matt wrote:
I have a user who has a fax machine plugged into an ATA.
They are able to SEND faxes just fine. Faxes go through wonderfully.
However, when someone tries to send them a fax, their fax machine
never receives it. And eventually the sending machine
Joseph wrote:
Easy and cheap.
Get two gateways AG-468 (each have 4 FXS ports) made by Atcom
http://www.voip-info.org/tiki-index.php?page=Atcom
one is about 88/ea
I have two on the way and will let you know how it works.
I would be interested in knowing how these work as well
Sean
--
Strange things.
When I run the RxFAX command through an internally dialed extension, I
can *hear* fax tones, meaning, I presume, that the RxFAX application is
running. In fact, doing a show application confirms that. So, I'm
presuming RxFAX application is talking as it should.
However,
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