Hello all,
I am planning to setup a MeetMe conference functionality on
Asterisk-1.4.13without having a Zaptel card. All users will be
calling through SIP only.
AFAIK, the said application needs a timer which makes use of the ztdummy
module. I have basically two (2) problems I am encountering
Updated to 138 responses:
http://voipusersconference.org/poll/
Anyone care to try to make a report of this that look like something?
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On 12/3/07, Steve Davies [EMAIL PROTECTED] wrote:
*BUMP* Does anyone have a workaround for the above? When a call is
attended-transferred to a queue, MOH is stopped. Perhaps someone can
tell me if it is fixed in 1.4.x ?
Sorry, replying to my own thread - Even if there is no fix, if someone
Hi
This is not the same thread. Right now I´m asking about REMOTE status.
I need from my asterisk to keep track of a remote (another sip server)
extension status. It´s not needed to know if it´s busy or DND, I think
for me would be enough to know that the user is connected at the remote
sip
Hi People!
Is there an underground asterisk command reference
manual that the Gurus here share amongst themselves
only? :-)
The reason I ask is that sometimes I see mention of an
asterisk command and I scramble for my asterisk book
(pdf) to look it up but can't find it in there. For
example, I
On 11/27/07, Steve Davies [EMAIL PROTECTED] wrote:
Hi,
I will confess immediately that this is only tested on 1.2.24, and I
would be interested to know if it happens on 1.4, but I cannot find a
bug-tracker entry which represents this issue.
Consider a PSTN call which comes into asterisk,
Dear Expert,
I am stuck when trying to register SPA-3102 on AsteriskNow ..
could any body please advise .. where can I find the article for doing this? ..
I googled but got nothing..
Regards
bie
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Am using perl AGI to invoke the dial command thus:
$AGI-exec('Dial',$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002));
What I expected that this will do is:
1. call the number using the string $numtodial2 - works OK
2. Set call limit to $maxcall and play a message $msgtime milliseconds
before
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Russell Bryant
Sent: 01 December 2007 00:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SLA: Handling of errors in outgoing call
Steve Langstaff
On Sun, Dec 02, 2007 at 01:06:50PM +1300, Matt Riddell wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
equis software wrote:
Hi, I'm testing Asterisk 1.4.15 with the -g option.
When it crash didn´t generate core file in the /tmp folder.
What is happening??
Check the directory
Am using perl AGI to invoke the dial command thus:
$AGI-exec('Dial',$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002));
What I expected that this will do is:
1. call the number using the string $numtodial2 - works OK
2. Set call limit to $maxcall and play a message $msgtime milliseconds
before
On Mon, Dec 03, 2007 at 04:11:50PM +0800, GNUbie wrote:
Hello all,
I am planning to setup a MeetMe conference functionality on
Asterisk-1.4.13without having a Zaptel card. All users will be
calling through SIP only.
AFAIK, the said application needs a timer which makes use of the ztdummy
Hello list,
I try to setup an asterisk-server with different SIP-Peers to PSTN.
The Peer are working and configured in sip.conf:
[peer1]
type=peer
host=10.10.10.1
[peer2]
type=peer
host=10.10.10.2
Now dialout is no problem. Extensions.conf says:
exten =
Dear Matthew,
Would you mind giving somehints about these module parameters to me. I just add
two manager commands to chan_zap in order to let the user adjust the gains at
runtime but dont know anything about the variables you mentioned and like to
know ;)
Regards.
---
M. Shokuie Nia,
SENA
Hello Tzafrir,
On Dec 3, 2007 7:24 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
You need to modprobe it. Or insmod zaptel first.
# modprobe ztdummy
FATAL: Module ztdummy not found.
# modprobe zaptel
FATAL: Module zaptel not found.
# insmod /lib/modules/2.6.18-5-xen-amd64/misc/zaptel.ko
On Mon, Dec 03, 2007 at 08:34:40PM +0800, GNUbie wrote:
Hello Tzafrir,
On Dec 3, 2007 7:24 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
You need to modprobe it. Or insmod zaptel first.
# modprobe ztdummy
FATAL: Module ztdummy not found.
# modprobe zaptel
FATAL: Module zaptel not
On Mon, 3 Dec 2007 16:34:39 +0700, Newbie [EMAIL PROTECTED]
wrote:
I am stuck when trying to register SPA-3102 on AsteriskNow ..
I don't use AsteriskNow, but I did use the SPA-3102. The idea is that
you must create an account for it in sip.conf, and configure the 3102
to connect to the * server
On Sun, Dec 02, 2007 at 03:46:32PM -0500, Bryan M. Johns wrote:
Make certain that selinux, iptables and ip6tables are disabled and off.
Hmmm... Is selinux necesserily an issue with Asterisk?
If so: anybody with a ready-made selinux policy for his own Asterisk
installation that could be shared
But with older versions of Asterisk It didn´t happend.
Which libraries?
Postgres Libraries or Asterisk librarries?
On Dec 1, 2007 10:10 PM, Tilghman Lesher [EMAIL PROTECTED]
wrote:
On Saturday 01 December 2007 09:43:41 equis software wrote:
In Asterisk 1.4.15 if I try to configure
Is not a configuration problem because if I run an older version of Asterisk
(ie 1.4.12) it generates this core in /tmp
With 1.4.15 I did # updatedb and then #locate core* and I didn´t found.
I found why Asterisk 1.4.15 crash, is because of this error:
asterisk: symbol lookup error:
Hello Tzafrir,
On Dec 3, 2007 8:50 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
What is the output of:
uname -r
2.6.18-5-xen-amd64
ls /boot/System.map-`uname -r`
ls: /boot/System.map-2.6.18-5-xen-amd64: No such file or directory
# ls -a /boot/
. ..
This is because the Asterisk-1.4.13
On Mon, 2007-12-03 at 09:19 +, Jeng Yu wrote:
Is there an underground asterisk command reference
manual that the Gurus here share amongst themselves
only? :-)
The reason I ask is that sometimes I see mention of an
asterisk command and I scramble for my asterisk book
(pdf) to look it up
www.voip-info.org
On Dec 3, 2007, at 4:19 AM, Jeng Yu wrote:
Hi People!
Is there an underground asterisk command reference
manual that the Gurus here share amongst themselves
only? :-)
The reason I ask is that sometimes I see mention of an
asterisk command and I scramble for my asterisk
Rajeev Natarajan wrote:
Am using perl AGI to invoke the dial command thus:
$AGI-exec('Dial',$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002));
The problem is that you have one too many pipes ('|') in your Dial string.
Change it to this:
On Mon, Dec 03, 2007 at 09:45:23PM +0800, GNUbie wrote:
Hello Tzafrir,
On Dec 3, 2007 8:50 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
What is the output of:
uname -r
2.6.18-5-xen-amd64
ls /boot/System.map-`uname -r`
ls: /boot/System.map-2.6.18-5-xen-amd64: No such file
Hi,
I have successfully configured two OpenBSD ( 4.2 4.0 ) Servers to do
IXA2 peering on two remote Sites.
Now asterisk users on Site1 can talk to users on Site2.
I just would like to know the following details.
1)
Currently I have allowed all in coming traffic from Site1 Public IP
Address on
Siju George wrote:
What are the security ramifications of peering two Asterisk servers on
remote locations and sending the VOIP traffice through the internet
using IAX2 ? Can this traffic be sniffed and the Voice be captured and
heard by any third party?
Yes.
If so is ther a way to prevent
Dear members of the list,
I have difficulties to obtain sync with a Digium TE420 PCI Express
For four entries E1, In my case I am with only 3 E1s available to
configure. The
telephony operator is BrasilTelecom and Signaling is R2 Digital.
The information I have is that are E1s with 32
Philipp Kempgen wrote:
Siju George wrote:
What are the security ramifications of peering two Asterisk servers on
remote locations and sending the VOIP traffice through the internet
using IAX2 ? Can this traffic be sniffed and the Voice be captured and
heard by any third party?
Great! thanks
On Dec 3, 2007 8:31 PM, Mark Michelson [EMAIL PROTECTED] wrote:
Rajeev Natarajan wrote:
Am using perl AGI to invoke the dial command thus:
$AGI-exec('Dial',$numtodial2|30|L($maxcall:$msgtime)|M(conn^1002));
The problem is that you have one too many pipes ('|') in your Dial
Roger C. Beraldi Martins wrote:
Dear members of the list,
I have difficulties to obtain sync with a Digium TE420 PCI Express
For four entries E1, In my case I am with only 3 E1s available to
configure. The
telephony operator is BrasilTelecom and Signaling is R2 Digital.
*snipped
The
Hi,
Has anyone a hint for click2call from Thunderbird Address Book ?
I know Snapanumber but would be interested tio discover alternatives.
Regards
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To
From the command line do sudo /usr/sbin/asterisk -r
then at that command line type show application $application
Where $application is the application you are interested in knowing info
about.
Jeng Yu wrote:
Hi People!
Is there an underground asterisk command reference
manual that the Gurus
Siju George wrote:
Hi,
I have successfully configured two OpenBSD ( 4.2 4.0 ) Servers to do
IXA2 peering on two remote Sites.
Now asterisk users on Site1 can talk to users on Site2.
I just would like to know the following details.
1)
Currently I have allowed all in coming traffic
You can also check:
http://www.the-asterisk-book.com/
This online book has a good reference of applications and functions of
Asterisk.
Best regards, Tomás.
On Dec 3, 2007 2:11 PM, Anthony Francis [EMAIL PROTECTED] wrote:
From the command line do sudo /usr/sbin/asterisk -r
then at that
I admit I haven't seen an attractive-enough reason to switch from
straight extensions.conf to AEL for the dialplan.
I use a little AGI written in PHP for when I DO need fancier
capabilities, mainly because of the ease of reusability of
my intranet's existing PHP scripts.
Vincent wrote:
On
Richard,
so I sould use 'unused' for de 4th span, but I don't find information
about how do this
configuration. I think it's something like this:
span=4,0,0,unused
that's it ? What do you think ?
If it's dosn't work I will check for de cable buildings.
Thank you for your help !
2007/12/3,
Hi,
I'm very new to asterisk and managed to set one up in debian, I
installed via apt-get the asterisk and asterisk-bristuff packages. I
downloaded the bristuff source as well.
I managed to get as far as loading the following modules:
zaptel199144 4
Well do you have a packet filter between the asterisk box and your phone? Is
the phone or the asterisk behind a NAT? Do you have an asymmetric route for
traffic in your network? Does the media take the same path inbound and
outbound between the asterisk and the phone? If you take a packet capture
For the HFC-4S (4 bri channels) you need to qozap driver not zaphfc
Regards
Stelios S. Koroneos
Digital OPSiS - Embedded Intelligence
http://www.digital-opsis.com
Quoting Elijah [EMAIL PROTECTED]:
Hi,
I'm very new to asterisk and managed to set one up in debian, I
installed via apt-get
Your problem seems to be that the card is in T1 mode and you need it to
be in E1 mode. Check the jumpers on the card and change them to the E1
position. Or you can send the module a parameter to put the card in E1
mode.
On Mon, 2007-12-03 at 13:14 -0200, Roger C. Beraldi Martins wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Good evening, I have something strange,
I have unread message in my voicemail box but the SIP NOTIFY that are
received by my telephone are like:
whereas there is voice messages inside!
Any idea how to solve that? Thanks
PS: I'm using asterisk 1.4.13
Hi,
I' still fighting the problem, that I can talk from one SIP phone to
another, but I can't hear the output of the playback or similar
applications:
exten = 202,1,ANSWER()
exten = 202,2,PLAYBACK(tt-monkeys)
exten = 202,3,HANGUP()
When I dial 202, asterisk show the
I'm sure this has been discussed many times, but I have a question about
hoteling.
My understanding would be this:
A phone sitting on a desk. A user hits 9000 and it asks what extension
you'd like to become. You type 1001 and then it asks for your
password. You type 1234, and it says you're
On Monday 03 December 2007 18:58:22 Stelios Koroneos wrote:
For the HFC-4S (4 bri channels) you need to qozap driver not zaphfc
--
I was using bristuff on a Junghanns quadBRI up to a few weeks ago, trying to
upgrade the bristuff that had been installed 2 years was giving so many
problems I
Rob Schall wrote:
A phone sitting on a desk. A user hits 9000 and it asks what extension
you'd like to become. You type 1001 and then it asks for your
password. You type 1234, and it says you're logged in. You now are
accepting calls at your phone and you're getting mwi on that phone for
#span=4,0,0,cas,hdb3
^ uncomment that
and
cas=1-15:1101
cas=17-31:1101
cas=32-46:1101
cas=48-62:1101
cas=63-77:1101
cas=79-93:1101
unused=94-124 #whatever your ending is
Roger C. Beraldi Martins wrote:
Richard,
so I sould use 'unused' for de 4th span, but I don't find information
On Tue, Dec 04, 2007 at 01:25:53AM +0800, Elijah wrote:
Hi,
I'm very new to asterisk and managed to set one up in debian, I
installed via apt-get the asterisk and asterisk-bristuff packages. I
downloaded the bristuff source as well.
I managed to get as far as loading the following
Richard,
Thanks understood I will use this configurations for de last span. But
I think the Carlos Chávez are right about this. I realy forgot to put
jumpers to set E1 mode in TE420 card, if it's come with the jumpers
open (and I believe this) probably this is the problem.
I don't know what
Sorry, not sure I understand the question. What is the problem here?
On Mon, 3 Dec 2007, Marc LEURENT wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Good evening, I have something strange,
I have unread message in my voicemail box but the SIP NOTIFY that are
received by my
My quick guess would be that it's a timing issue. You didn't mention
whether you are using a Zaptel device or ztdummy.
I know this sounds like I'm being a smart***, but I'm not... try this...
rub the mouthpiece of the file while the sound file is playing and see if
you hear any of the file. If
John Constalgie wrote:
My updated SEPMAC file for this hard phone is at
http://cid-ff3ef0764138e401.skydrive.live.com/self.aspx/Public/SEP001E4A5F1270.cnf.xml
try set the backup, emergency, and outbound proxies to blank
under sipProxies section:
sipProxies
backupProxy/backupProxy
I have zero issues with multihomed asterisks.
One potential issue is that some people are multihoming onto the same subnet.
This will cause issues with many applications as normal routing usually sends
data OUT the lower IP address if there are two on the
same subnet.
Multihoming, as a rule
I think Lacy means rub the mouthpiece of the phone - to make sound (blowing
into it should yield the same result)
Lacy Moore wrote:
My quick guess would be that it's a timing issue. You didn't mention
whether you are using a Zaptel device or ztdummy.
I know this sounds like I'm being a
On Monday 03 December 2007 12:40, Roger C. Beraldi Martins wrote:
Thanks understood I will use this configurations for de last span. But
I think the Carlos Chávez are right about this. I realy forgot to put
jumpers to set E1 mode in TE420 card, if it's come with the jumpers
open (and I believe
So you mean have a script rewrite the MAC-phone.cfg file, correct? If I
do that, then i'll have to have the phone reboot (which i can do), but
that really isn't a virtual extension anymore..
Rob
Philipp Kempgen wrote:
Rob Schall wrote:
A phone sitting on a desk. A user hits 9000 and it
Not strictly an Asterisk question.
I've tried to install adhearsion on TWO relatively fresh CentOS 5.x systems,
and I get this...
[EMAIL PROTECTED] rubygems-0.9.5]# gem install adhearsion
Bulk updating Gem source index for: http://gems.rubyforge.org
ERROR: While executing gem ...
On Monday 03 December 2007 11:33, Tomás Laureano Peralta Tormey wrote:
You can also check:
http://www.the-asterisk-book.com/
This online book has a good reference of applications and functions of
Asterisk.
If the author is paying attention, in the example for ODBC_USER_DATABASE,
the write
On Sun, 02 Dec 2007 23:56:25 +0200, Zoa [EMAIL PROTECTED] wrote:
There are many, (i'm one of the people working for zoiper):
In that case, I think it'd be useful to add a forum on the site, so
people can post when they have problems with the software :-)
Look at the iaxclient homepage,
Thanks
Actually I believe the process you are describing is the agentcallback feature.
Once you are logged in if the agent is configured to have voicemail and does
the light should come on.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent:
On Mon, 03 Dec 2007 08:14:32 -0900, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
I admit I haven't seen an attractive-enough reason to switch from
straight extensions.conf to AEL for the dialplan.
Thanks. I need to let admins add new items in the database (people who
called with
Tilghman Lesher wrote:
On Monday 03 December 2007 12:40, Roger C. Beraldi Martins wrote:
Thanks understood I will use this configurations for de last span. But
I think the Carlos Chávez are right about this. I realy forgot to put
jumpers to set E1 mode in TE420 card, if it's come with the
Tilghman Lesher wrote:
On Monday 03 December 2007 11:33, Tomás Laureano Peralta Tormey wrote:
You can also check:
http://www.the-asterisk-book.com/
This online book has a good reference of applications and functions of
Asterisk.
If the author is paying attention, in the example for
Rob Schall wrote:
So you mean have a script rewrite the MAC-phone.cfg file, correct?
Yes, either rewrite it or generate it dynamically by a
PHP/Perl/... script.
If I
do that, then i'll have to have the phone reboot (which i can do), but
that really isn't a virtual extension anymore..
Hi Edwin,
I did what you said for the SEP file ( updated SEP xml file :
http://cid-ff3ef0764138e401.skydrive.live.com/self.aspx/Public/SEP001E4A5F1270.cnf.xml
)
By the way, I was reading up online that I could change the qualify=yes
setting to no in sip_additional.conf to make my phone
Lacy Moore wrote:
My quick guess would be that it's a timing issue. You didn't mention
whether you are using a Zaptel device or ztdummy.
I think timing is only an issue with meetme conferences, right? I don't
believe you need a hardware or ztdummy timing source to make the
Playback command
I agree, if you never changed the jumper.
I have never noticed, does the output of ztcfg change is it set to E1?
Roger C. Beraldi Martins wrote:
Richard,
Thanks understood I will use this configurations for de last span. But
I think the Carlos Chávez are right about this. I realy forgot to
Richard Lyman wrote:
I have never noticed, does the output of ztcfg change is it set to E1?
Yes. More channels. :)
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Hi,
My quick guess would be that it's a timing issue. You didn't mention
whether you are using a Zaptel device or ztdummy.
I'm using ztdummy, and yes, I guess your're right - it seems to be a
timing problem, because I found the following messages in /var/log/messages:
Dec 3 22:51:36
Search for ztdummy, zttest and Zaptel Issue 11153 in the Dev Mailing
List. You might have a buggy kernel.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stefan
Guenther
Sent: Monday, December 03, 2007 5:00 PM
To: asterisk-users@lists.digium.com
Subject:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi All,
Has anyone been successful in making ekiga's chat functionality work
with Asterisk. This is a really neat feature and it would be awesome to
finally see it working.
- --
Alan Hanley
FSF Member 4949
No matter where you go , you're probably
Hi!
So you mean have a script rewrite the MAC-phone.cfg file, correct? If
I do that, then i'll have to have the phone reboot (which i can do),
but that really isn't a virtual extension anymore..
Then do it the other way around: Always use the same (virtual) voicemail
box for a specific
On Sat, 2007-12-01 at 10:22 -0700, Anthony Francis wrote:
[EMAIL PROTECTED] wrote:
Anyone have an idea how to implement a phone number that can only be
called once? The first time it will process normally and any
subsequent calls will be rejected.
Philipp Kempgen wrote:
Richard Lyman wrote:
I have never noticed, does the output of ztcfg change is it set to E1?
Yes. More channels. :)
Regards,
Philipp Kempgen
only if defined G
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John Constalgie wrote:
Hi Edwin,
I did what you said for the SEP file ( updated SEP xml file
:
http://cid-ff3ef0764138e401.skydrive.live.com/self.aspx/Public/SEP001E4A5F1270.cnf.xml
)
By the way, I was reading up online that I could change the
qualify=yes setting to no in
That would be VERY much appreciated Russell,
There seems to be a lack of info and the accompanying
confusion/misinformation about this.
-Original Message-
From: Russell Bryant [mailto:[EMAIL PROTECTED]
Sent: Friday, November 30, 2007 4:11 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing
Hello Tzafrir,
On Dec 3, 2007 11:01 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Interesting. That explains why depmod was not run at package install
time. So the next question is: why is that file missing? Do you have
any guess?
It is because this is on the Xen domU where it only uses the
On Sep 5, 2007 3:36 PM, Kai-Uwe Jensen [EMAIL PROTECTED] wrote:
How are you playing the voice? Do you use something like app_swift
or app_cepstral? Just fixed app_swift for my own installation by
changing the framesize constant definition from 160*4 to 20,
after googling for a similar issue.
hi folks.
i have a Digium TE220 PCI-E 2 port T1/E1 controller installed
in an IBM x3400 server. i load the wct4xxp driver seems ok.
but when i execute ztcfg -vvv command. the kernel panic.
i tried zaptel 1.2.21 22. they have the same result.
following is my zaptel.conf:
loadzone=cn
I have searched for this without much luck. I want to be able to send
public-address-like notices over VoIP phones. The LinkSys SPA-941
auto-answer support comes close to working, except that if you are
currently in a call it places that call on hold without warning. I'm
willing to consider a
This is the core trace
(gdb) bt
#0 0xb7e5a231 in strcasecmp () from /lib/libc.so.6
#1 0xb7ce0a3f in local_ast_moh_start (chan=0x82496a8, mclass=0xb720f828
default, interpclass=0x0)
at res_musiconhold.c:646
#2 0x08083695 in ast_moh_start (chan=0x64, mclass=0x64 Address 0x64 out of
bounds,
Hi!
In 1.4.15 I have 3 agents, while 4 calls are waiting, 2 agents are ringing
and the third agent don´t ring.
I´m using autofill=true
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To UNSUBSCRIBE or
At 22:13 12/3/2007, Doug Meredith wrote:
Content-class: urn:content-classes:message
Content-Type: multipart/alternative;
boundary=_=_NextPart_001_01C8362C.05AB2A2D
I have searched for this without much luck. I want to be able to
send public-address-like notices over VoIP phones.
I think the newer version of the firmware fixes this problem.
Paul Hales
AsteriskIT
On Tue, 2007-12-04 at 00:13 -0400, Doug Meredith wrote:
I have searched for this without much luck. I want to be able to send
public-address-like notices over VoIP phones. The LinkSys SPA-941
auto-answer
Hi,
SUCCESS! I've been working on zaphfc for hours and didn't realize I've
been using the wrong driver all this time. Thanks!
Best regards,
Elijah Alcantara
On Mon, 2007-12-03 at 19:58 +0200, Stelios Koroneos wrote:
For the HFC-4S (4 bri channels) you need to qozap driver not zaphfc
Hi friends.
I have problems with the voicemail system, when some user forward the
message to other box all the Asterisk falls down and restart.
How do I disable the option to forward messages in voicemail (option 8 in the
menu)? and Which can be the cause for the problem if I wanna use forward
On Mon, Dec 03, 2007 at 09:56:51PM +0100, Philipp Kempgen wrote:
Tilghman Lesher wrote:
On Monday 03 December 2007 12:40, Roger C. Beraldi Martins wrote:
Thanks understood I will use this configurations for de last span. But
I think the Carlos Chávez are right about this. I realy forgot to
On a similar note... has anyone ever seen a SIP-based door intercom unit?
Functionality I'm looking for is... basically an outdoor rated weather
resistant speaker with 1 button and microphone, when the button is
pressed, it dials a specified SIP extension. Likewise, from the Asterisk
box,
If I was wanted to multi-home on the same subnet I would use Ethernet
Bonding (similar to Windows Teaming) in a failover configuration. This will
make one of the links on the LAN active and the second one as a failover in
case the first one goes down. It takes a couple seconds for the 2nd link
On Mon, Dec 03, 2007 at 10:51:43PM +0100, Philipp Kempgen wrote:
Richard Lyman wrote:
I have never noticed, does the output of ztcfg change is it set to E1?
Yes. More channels. :)
No. The channels listed in ztcfg -vv are the channels you wrote in
zaptel.conf . By the time they are
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