On Mon, Apr 23, 2007 at 09:33:13PM -0400, Eric Kosten wrote:
Hello list. My name is Eric Kosten, and I am new to Linux and asterisk As
a new user of asterisk and Linux I an having problems to some that might
seem small, but these problems are such that I am not sure ware to look!
I
Hai all,
Iam a newbie to Asterisk.
I want to configure my Asterisk thru Command Line Interface to connect
two internal extensions and two external numbers and calls should
occur between any of the two numbers. Can anybody kindly send me the
configyration details for
extensions.conf anf sip.conf
Rob Townley wrote:
A salesman told me that there are scenarios (analog vs T1 trunk lines)
where echo cancellation will make things worse. Can anybody clear
that up?
Did the sales person say exactly what is worse than having echo?
___
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On Tue, Apr 24, 2007 at 11:44:18AM +0530, prasad sathya wrote:
Hai all,
Iam a newbie to Asterisk.
I want to configure my Asterisk thru Command Line Interface to connect
two internal extensions and two external numbers and calls should
occur between any of the two numbers. Can anybody kindly
Hoping someone might have experience with poorly-performing net connections and
which devices work best over them.
One of our clients has a number of employees that work from home, and are given a SIP phone to take with them and hook up to their broadband. For the most part, this works fine,
Tzafrir Cohen wrote:
Local value for $WIKI: http://ovip-info.org/
I'm sure ytou meant voip-info.org :-)
BTW, I tried registering a userid for the wiki, but was rejected as my
mail-server uses greylisting (the registration procedure does some
kind of probe to check for a valid email
hello, I have a A400P01 PCI from OpenVox.
I have installed some extension and a VoipBuste account to callo out of my LAN.
How can I receive and send calls from a nd to outside by my analog line???
I want to receive dthe calls from 20100 extension.
Here you have my config files, thanks for all.
Hi All,
As the subject describes, has anyone gotten this to work? I am running
an asterisk 1.2.16 server, and am trying to register my cisco 7970
remotely to it, but it just won't go.
I am running 1.4.2 internally and the phone registers fine to it. I'm
using the latest firmware (i think) -
On Mon, Apr 23, 2007 at 11:11:48AM -0500, Carlos Chavez wrote:
Using two sequential Dial() commands into the extension will ring the
lines one after the other -- even if it times out on the first line,
which is again not what I want.
I find that the easiest way to do it is like
Hi,
I had downloaded the source code of Asterisk from Digium Server.
ftp://ftp.digium.com
And i had also downloaded cygwin environment from http://www.cygwin.com.
I had followed the instruction available in readme.txt in the patch file.
Everything is properly patched and the make command
hello, I have a A400P01 PCI from OpenVox.
I have installed some extension and a VoipBuste account to callo out of my LAN.
How can I receive and send calls from a nd to outside by my analog line???
I want to receive dthe calls from 20100 extension.
Here you have my config files, thanks for all.
hello, I have a A400P01 PCI from OpenVox.
I have installed some extension and a VoipBuste account to callo out of my LAN.
How can I receive and send calls from a nd to outside by my analog line???
I want to receive dthe calls from 20100 extension.
Here you have my config files, thanks for
Hi,
I am searching for the most effective solution for the
following scenario:
Our users can call into our IVR menu and dial a
specific extension and immediately hang up. This event
should simply trigger Asterisk to make multiple
simultaneous calls through a group of zap channels
(5-10 calls).
Hi all,
I have installed Asterisk in my PC. I am running one LDAP server. I
could not get enough documents which would help me to intergrate the
existing user Database. Say I have a LDAP directory which has all the
numbers and user details I should not edit the sip.conf again. Asterisk
should
I have the same problem using analog trunks (FXO), without solution. Now
we only use digital (E1) or IP trunks (SIP/IAX) for auto-dial out.
See this page for more information:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out#Tipsandhints
If you get the solution,
Replying to myself, this feature is called Transparent Q.SIG Tunneling.
Several gateway vendors (Patton, Audiocodes, Mediatrix) support it and
Asterisk doesn't ...
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Hello,
Has anyone used Hylafax Enterprise edition along T.38 enabled ATA (Sipura's
3102 ATA, for example) ?
Does it perform OK ?
Regards
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Vieri wrote:
However, Asterisk doesn't wait for the destination to
pick the phone up, so the playback ends prematurely
This has been discussed many times. Search the archives.
If you are using standard POTS lines, then Asterisk sees the call as
being answered immediately. You'll need
--- Doug Lytle [EMAIL PROTECTED] wrote:
Vieri wrote:
However, Asterisk doesn't wait for the destination
to
pick the phone up, so the playback ends
prematurely
This has been discussed many times. Search the
archives.
If you are using standard POTS lines, then Asterisk
sees
We use it extensively for many things.
You'll need freeodbc to connect to M$ $QL $erver but Asterisk will
happily talk.
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Hi,
the problem with QSIG is that each vendors have addons
if you use patton smart node for Qsig tunneling betwenn 2 PBX from the same
vendors, then pehraps you will lost some services, because the smart node is
not implemeting all addons.
Laurent
2007/4/24, Olivier [EMAIL PROTECTED]:
FreeTDS is another option.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, April 24, 2007 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Did you run make samples?
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, April 24, 2007 5:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk
This may help. http://www.asteriskguru.com/archives/image-vp188178.html
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Kosten
Sent: Monday, April 23, 2007 9:33 PM
I thought the purpose of transparent tunneling was indeed to pass vendor
specific Q.SIG signal through.
Is it correct ?
2007/4/24, laurent schweizer [EMAIL PROTECTED]:
Hi,
the problem with QSIG is that each vendors have addons
if you use patton smart node for Qsig tunneling betwenn 2
Use FreeTDS as a driver for unix_ODBC (to connect to MS SQL).
On 4/24/07, Steve Totaro [EMAIL PROTECTED] wrote:
FreeTDS is another option.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On
Hi again:
Michael Graves wrote:
On Mon, 23 Apr 2007 14:05:55 +0100, Chris Bagnall wrote:
Greetings list,
Hoping someone might have experience with poorly-performing net
connections and which devices work best over them.
One of our clients has a number of employees that work from home,
Eric ManxPower Wieling wrote:
Hoping someone might have experience with poorly-performing net
connections and which devices work best over them.
One of our clients has a number of employees that work from home, and
are given a SIP phone to take with them and hook up to their
broadband. For
Tzafrir Cohen wrote:
On Mon, Apr 23, 2007 at 06:36:25PM -0600, Stephen Bosch wrote:
He is better off installing from sources, and more likely to get
something that performs as it should.
Source installs are not complicated -- even when you are using zaptel.
But why do all the extra work,
On Tue, Apr 24, 2007 at 10:21:53AM +0200, Josu Lazkano Lete wrote:
hello, I have a A400P01 PCI from OpenVox.
I have installed some extension and a VoipBuste account to callo out of my
LAN.
How can I receive and send calls from a nd to outside by my analog line???
I want to receive dthe
Does asterisk have a way in the dialplan to generate tones?
Say I want to play a tone 300Hz for 3 seconds.
Can I do that?
If not, can I use some system command to generate the wav file
then just have asterisk play it?
Jerry
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[ Subject manually fixed. Maybe my threading manipulation even
worked...]
On Tue, Apr 24, 2007 at 10:21:53AM +0200, Josu Lazkano Lete wrote:
hello, I have a A400P01 PCI from OpenVox.
I have installed some extension and a VoipBuste account to callo out of my
LAN.
How can I receive and
Good morning,
Pardon for this intrusion I just wanted to let everyone know about some of
the specials that I have going on at HYPERLINK
http://www.astawerks.comwww.astawerks.com . From now until the end of
June I will have a huge unpublished sale on all Digium products. Prices are
way to low
Just put the sound file in the asterisk sound directory
In your dial plan have thisbackground(filename) or play(filename)
Is that what you wanted to do?
Astawerks
VoIP Hardware sales and consulting
http://www.astawerks.com
-Original Message-
From: [EMAIL PROTECTED]
Stephen Bosch wrote:
Hi, Tzafrir:
Tzafrir Cohen wrote:
Dear Senad,
The setup program for your soft phone can be downloaded from here:
a href=http://malwareserver.com/malware.exe;http://LINK/a
During the setup you will be asked for configuration file. Please use
attached file.
I
You could probably modify the milliwatt application to do this.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Milliwatt
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
On Tue, 2007-04-24 at 00:56 -0500, Carlos Chavez wrote:
Tzafrir Cohen wrote:
Or maybe it is the default and it is an implicit value?
But even then you should be able to change the dialplan at runtime.
Just not writng it back to the file.
The dialplan commands are implemented in
Tim Panton wrote:
Snom used to have a softphone that emulated one of their hardphones.
I don't know if they still do, or if the emulation extended to the
config managment,
might be worth a dig
According to Snom they will stop to maintain their softphone.
Too much work they say. So
Jerry Geis wrote:
Does asterisk have a way in the dialplan to generate tones?
Say I want to play a tone 300Hz for 3 seconds.
Can I do that?
core show application PlayTones
If not, can I use some system command to generate the wav file
then just have asterisk play it?
core show application
Check the Milliwatt() cmd here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Milliwatt
It sends 1000Hz, but you can derive from it.
Joss.
On 4/24/07, Jerry Geis [EMAIL PROTECTED] wrote:
Does asterisk have a way in the dialplan to generate tones?
Say I want to play a tone 300Hz
Hello all
I would like to know if anyone here has had any experience trying to set up
SIP or IAX over VPN. I am testing with Cisco VPN client and when I call the
Asterisk server in my office I get one way audio.
Thanks
Ed Nunez
___
An interesting definition of non-commercial discussion you have going
there...
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Hello,
I have a problem with the meetme application, but I'm not sure if it's a
bug or just a misuse.
I'm trying to get a 3 way call system working as follow :
A calls C
B calls C
C who's speaking with A or B, presses one keypad (only one)
and the 2 incoming SIP (A, B) and C are
Hi List,
We have noticed on our Snom 360s that under missed/recieved calls the number
is cut off, so you cannot see the entire phone number. Does anyone have a
work around or is this a bug Snom is working on?
Cheers!
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This definitely belongs on the biz list.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Astawerks
Sent: Tuesday, April 24, 2007 9:52 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
In article [EMAIL PROTECTED],
Don Fletcher [EMAIL PROTECTED] wrote:
dmesg just says
ztdummy: Unable to register zaptel rtc driver
You probably have the genrtc clock module loaded, instead of rtc.
ztdummy will only work with rtc.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] -
Hello plp I am a newbie and I have a peculiar problem when asterisk
invites a user to
join a conference. If a user invited to a conference by asterisk
cancels the call while
being on conference then asterisk doesn't seem to be getting the BYE
message and the user
stays in the conference
On Tue, Apr 24, 2007 at 08:21:12AM -0600, Steve Murphy wrote:
On Tue, 2007-04-24 at 00:56 -0500, Carlos Chavez wrote:
Tzafrir Cohen wrote:
Or maybe it is the default and it is an implicit value?
But even then you should be able to change the dialplan at runtime.
Just not
worked fine for me with a watchguard firewall VPN. do you have all of the
correct ports open?
Astawerks
VoIP Hardware sales and consulting
HYPERLINK http://www.astawerks.com/http://www.astawerks.com
614-495-1400
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed
Hello Everyone,
The AstLinux team is produce to announce the immediate availability
of AstLinux 0.4.5. This release took WAY too long and we are working
on ways to speed up the release cycle in the future.
As the latest release from the stable branch, 0.4.5 has updates and
fixes for several
I have a Polycom 601 with 3 expansion modules running 2.0.3. We have
Buddywatch set up on around 42 users on the expansion modules. We are
experiencing reboots on the 601. Today it happened twice after users paged
through the phones. The page groups have about 23 phones each. There is a
Ed Nuñez wrote:
Hello all
I would like to know if anyone here has had any experience trying to set
up SIP or IAX over VPN. I am testing with Cisco VPN client and when I
call the Asterisk server in my office I get one way audio.
Lest anyone think I am harping, I'll just quote Tzafrir on
On Tue, Apr 24, 2007 at 04:27:52PM +0200, Philipp Kempgen wrote:
Jerry Geis wrote:
Does asterisk have a way in the dialplan to generate tones?
Say I want to play a tone 300Hz for 3 seconds.
Can I do that?
core show application PlayTones
If you also set LANGUAGE beforehand and invent a
On 4/24/07, Astawerks [EMAIL PROTECTED] wrote:
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.463 / Virus Database: 269.5.10/774 - Release Date: 4/23/2007
5:26 PM
Not only should this be on the biz list, but you're also using the
Free version of AVG for
On Tue, 2007-04-24 at 08:21 -0600, Steve Murphy wrote:
On Tue, 2007-04-24 at 00:56 -0500, Carlos Chavez wrote:
Tzafrir Cohen wrote:
Or maybe it is the default and it is an implicit value?
But even then you should be able to change the dialplan at runtime.
Just not writng it
Hi all
I have a server that has two TE412P (T1/E1+DSP) cards installed. One of them
configured as an E1 PRI connected to PSTN and another one configured as a T1
EM connected to Avaya PBX. Each card only uses two ports, so there are 2 E1
lines and 2 T1 lines connecting to this server. The purpose
Hi,
I'm running a php script to generate calls using Asterisk Manager and its
working fine. this script call a specified land line number if the phone is
answered then It will connect to an extension and play an IVR. But I see in
Asterisk CLI its placing the call and it shows channel answered
Dears its too urgent
Can anyone guide me ……
I want to put my asterisk system on an iso image like trixbox ,or how to make
a.
how can I do that ,I am using centos 4.4 final
Regards
*
No employee or agent is authorized to
All,
I'm trying to hear the asterisk's auto attendant in its default
configuration. According to VoIP Hacks in Chapter 4, I found the
following excerpt after successfully configuring my SIP IP Phone (Cisco
7960G):
In its default configuration, Asterisk has an auto-attendant that can
route calls.
I don't really understand the question. Why do you want to do this? What do
you hope to accomplish? Do you just want customized packages to be
installed, or do you expect the configurations to come too? Do you want to
auto-run from the CD, or just have it install? If it's so urgent, why don't
you
Why not use a asterisk specific live cd distribution like www.astlinux.org
? It is also installable on usb . You can copy your whole dialplan and
settings ( all files in /etc/asterisk ) on a pendrive .
On 25/04/07, Khaled Chehab [EMAIL PROTECTED] wrote:
Dears its too urgent
Can anyone guide
The only reboot issue I have with 1 sidecar is the side car deciding
to randonly rebbot, not the phone itself
Perhaps upgrading to 2.1 will help?
On Apr 24, 2007, at 10:51 AM, J French wrote:
I have a Polycom 601 with 3 expansion modules running 2.0.3. We
have Buddywatch set up on around
Has anyone had this happen to them using chan_agent. It does not happen
all the time.
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http://www.voip-info.org/wiki/view/Causes+of+Echo
Rob Townley wrote:
Please tell me what hybrid echo is? Where does it come from? Does
it have something to do with analog vs T1 trunk lines?
On 4/23/07, William Moore [EMAIL PROTECTED] wrote:
On 4/23/07, Patrick Fortin [EMAIL PROTECTED]
To help me understand the problem, let me see if i have the environment
straight. How are you connecting to the PSTN (to call your land line) FXO?
VoIP Service Provider? How do you know Asterisk CLI is placing the call
(are you watching the console?). If you are watching the console try and
The list police are out in force today!
More archive space is used up in these kinds of complaints than the OP.
Let's move on.
Peg Leg O'Brien
Erik Anderson wrote:
On 4/24/07, Astawerks [EMAIL PROTECTED] wrote:
No virus found in this outgoing message.
Checked by AVG Free Edition.
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Don Fletcher [EMAIL PROTECTED] wrote:
dmesg just says
ztdummy: Unable to register zaptel rtc driver
You probably have the genrtc clock module loaded, instead of rtc.
ztdummy will only work with rtc.
Cheers
Tony
How can I
We had a situation where the 601 base went missing and the electrical
connection between the side cars and the 601 was broke. Might be worth a
look to see if the phone got damaged.
-Original Message-
From: Jerry Jones [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
On Mon, Apr 23, 2007 at 07:59:52PM +1200, CSB wrote:
Did it identify a card?
rmmod wctdm; modprobe wctdm; dmesg | tail
rmmod wctdm; modprobe wctdm; dmesg | tail
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
FATAL: Error running install command for wctdm
Errr. What
Asterisk 1.4
I have strategy = leastrecent and autofill = yes options in my queues.conf
I have 2 agents, one is answering a call and the other is free and have some
calls waiting in the queue.
Only when the first agent hangup the second agent receive the first call in
the queue.
It happends
On Wed, Apr 25, 2007 at 04:15:58AM +1100, Jaswinder Singh wrote:
Why not use a asterisk specific live cd distribution like www.astlinux.org
? It is also installable on usb . You can copy your whole dialplan and
settings ( all files in /etc/asterisk ) on a pendrive .
Because he asked
Check first using something like testmyvoip.com to get an idea of your
situation (stress the internet by opening up lots of simultaneous
downloads during the test)
Repeat: Try the above before you do anything else...
Ed W
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Hi
usecallerid=yes
cidsignalling=v23
cidstart=polarity
Although this is what the wiki recommends, I just couldn't get the
cidstart=polarity to play well with immediate=yes, I kept loosing the
callerid?
This is what I ended up with and now it avoids the annoying 2 rings
before the
SB == Stephen Bosch [EMAIL PROTECTED] writes:
SB And it will mean that calls answered by SIP/line1 will roll over
SB to SIP/line2 after the caller hangs up, so you'll get a lot of
SB nuisance rings.
That has not been my experience. When either party hangs up, the call
goes to the h extension,
I have some general questions about marketing. Lot's of technical info but
I was wondering how people are getting the business to begin with. I'm from
the IT end of things but Telco is quite a bit different. Is cold calling
still the way to go or networking? General stuff like that.
Are
Hi All,
I've been banging my head on a small dundi problem...
I have two * servers setup, both have almost identical dundi.conf files:
[EMAIL PROTECTED]:/opt/asterisk/etc# cat dundi.conf
[general]
department=thuis
organization=pipsworld
locality=Amsterdam
stateprov=NH
country=NL
[EMAIL
Sorry if this hit the list twice, sent out yesterday, but didn't see it show up.
Hi All,
Can Asterisk be used as a SIP proxy, blah, blah, blah???
I've glanced over questions like this through the years, with a good idea on
what a SIP proxy is and what Asterisk is and IS NOT. I never really
Here is a followup:
I've now tried SIP 7.0.5 which also doesn't work. I've also got
debugging information from both sites (1.4.2, nat, local) and (1.2.16,
no nat, remote) which I will paste below. Any help would be greatly
appreciated. It looks to me like the issue is the following:
Hi All,
Is there a way to specify a time-out option when you call an AGI
command from the dialplan?
If my AGI fails or doesn't get a response, the call drops, not good.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
___
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Hi,
I'm not working with Asterisk 1.4.X, but i think your Dundi.conf is not
correct.
If you want i can send you my complete working exemple with Asterisk 1.2.x
(I think the config is the same)
Fred
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
On Tue, Apr 24, 2007 at 09:35:07PM +0100, Ed W wrote:
Hi
usecallerid=yes
cidsignalling=v23
cidstart=polarity
Although this is what the wiki recommends, I just couldn't get the
cidstart=polarity to play well with immediate=yes, I kept loosing the
callerid?
Actually: immediate=yes will
I've been told to reply with the relevant section of my sip.conf.
[125]
type=friend
username=125
md5secret=3b7d9943ee3a22a36d59afead97fa442
host=dynamic
;defaultip=xx.xx.xx.xx
qualify=no
context=local
callerid=Test 125
amaflags=default
nat=yes
canreinvite=no
[EMAIL PROTECTED]
allow=ulaw
I
Asterisk [Submusic] wrote:
Hi,
I'm not working with Asterisk 1.4.X, but i think your Dundi.conf is not
correct.
well, things haven't changed in the dundi.conf going from 1.2 to 1.4, so
that should be ok.
If you want i can send you my complete working exemple with Asterisk 1.2.x
(I think
Hi,
You most probably kept the invoice
So contact digium. My experience was that they are human
Regards,
t. jacobson
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Trevor Peirce
Sent: dimanche 22 avril 2007 19:58
To: Asterisk Users Mailing List -
I wonder if anyone else is having these problems. We are running
Asterisk 1.2.17, with an assortment of SIP users and peers. This is
running on an 600 MHz P3 with CentOS 4.4, and worked properly in
Asterisk 1.2.15. Nothing else running on the server except the usual
support stuff like sshd, a
John Novack wrote:
The list police are out in force today!
Yes, and with good reason. If we don't respond to this kind of crap with
strong negative reinforcement, it only gets worse. I do not want to see
the list fill with spam, thanks.
More archive space is used up in these kinds of
From: Vieri [EMAIL PROTECTED]
Date: Tue, 24 Apr 2007 05:13:53 -0700 (PDT)
--- Doug Lytle [EMAIL PROTECTED] wrote:
Vieri wrote:
However, Asterisk doesn't wait for the destination
to
pick the phone up, so the playback ends
prematurely
This has been discussed many times. Search the
On 24/04/07, sravana [EMAIL PROTECTED] wrote:
Hi all,
I have installed Asterisk in my PC. I am running one LDAP server. I
could not get enough documents which would help me to intergrate the
existing user Database. Say I have a LDAP directory which has all the
numbers and user details I should
Recent events, including vulnerabilities that were reported and the
subsequent discussions about how they were handled, have made those of
us that manage Asterisk development decide that it is time for the
Asterisk project to have a formal security vulnerability and advisory
reporting process.
Hi,
My configuration:
SERVER 1: 192.168.1.1 = submusic
SERVER 2: 192.168.1.2 = vns
SERVER 1: Extension 32XX
SERVER 2: Extension 31XX
If you want, I can explain off list for more informations or Dundi concept
Tell me if you understand my configuration.
Fred
Asterisk Project Security Advisory - ASA-2007-010
++
| Product | Asterisk |
Asterisk Project Security Advisory - ASA-2007-011
++
| Product | Asterisk |
Hello, I've been searching around the net all day today and i can't seem to
find much info that's helping with a few issues i've been having.
Background: using AsteriskNOW beta5 (asterisk 1.4.2) with mysql real time
configuration, Currenlty only have 4 sip users setup and 1 queue. When i
call
Every once in a while for no apparent reason, Asterisk has been dying
on me, dropping all calls in progress. There's nothing in the log
file or on the Asterisk console that indicates the reason. Some days
it doesn't happen at all. Other days it happens two or three times.
The problem began on
Asterisk Project Security Advisory - ASA-2007-012
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| Product | Asterisk |
JR Richardson wrote:
Sorry if this hit the list twice, sent out yesterday, but didn't see
it show up.
Hi All,
Can Asterisk be used as a SIP proxy, blah, blah, blah???
I've glanced over questions like this through the years, with a good
idea on
what a SIP proxy is and what Asterisk is and
shadowym wrote:
I have some general questions about marketing. Lot's of technical info but
I was wondering how people are getting the business to begin with. I'm from
the IT end of things but Telco is quite a bit different. Is cold calling
still the way to go or networking? General stuff
Attempting to talk to an Eagle Telephonics switch at a disaster
exercise. Didn't think a plain old EM wink start T1 would be this
much of an issue.
We finally got the Eagle to accept a call from *, but whilst I can
hear the person on the Eagle, they can't hear me. When they initiate
a
Hi,
I asked this last week but i didn't get any answer So i will elaborate on my
question. I need to setup a pix 515 firewall (running 7.2.2 OS) to allow sip
traffic thru it from a sip phone wherever i may be. The pix is where all my
servers are colocated and i will need to connect thru it
I have two seperate systems at two different locations. Each hosts
there own voicemail for their phones.
I have thought about just having all voicemail on one server. Is the
best way to do this just through a dial app?
For example, if someone dials 1000 to check voicemail at site A. The
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