Hi all,
Anyone able to remotely reboot their password protected Sipura
SPA-3000 from command line. I am trying:
Sipura SPA-3000 from command line:
# wget http://admin:[EMAIL PROTECTED]/admin/reboot
The strange thing is it works fine when I go to
http://admin:[EMAIL PROTECTED]/admin/reboot with
On Sat, Aug 13, 2005 at 04:29:07PM +1000, Eric Bishop wrote:
Hi all,
Anyone able to remotely reboot their password protected Sipura
SPA-3000 from command line. I am trying:
Sipura SPA-3000 from command line:
# wget http://admin:[EMAIL PROTECTED]/admin/reboot
The strange thing is it
Hi
Kind of late reeply. Still it mat be useful for others.
On Wed, Aug 10, 2005 at 05:07:42PM +0100, Kevin Walsh wrote:
Joseph [EMAIL PROTECTED] wrote:
Don't forget to experiment with nice to increase priority of for
Asterisk.
By default asterisk run with priority 0 same as apache and any
On Fri, Aug 12, 2005 at 09:47:21AM -0600, Joseph wrote:
Though is the way to verify that asterisk is running with -p switch?
If Asterisk has failed to get real-time priority it should print an
appropriate error message and exit.
I've modified the startup script to start asterisk with -p;
On Thu, Aug 11, 2005 at 11:20:41PM -0500, David Williams wrote:
Trying to config the latest Asterisk/zaptel with an Digium Wildcard and a
single X100m FXO interface connected to a POTS analog line.
Build and install of both work ok - I'm using Suse 8 on a dual Pentium
box. I load the
On Fri, Aug 12, 2005 at 09:20:54PM -0600, Rich Adamson wrote:
Updated the kernel on a fc3 box from 2.6.9-1.667 to 2.6.12-1.1372_FC3
today. Now the cvs-head for zaptel won't compile (libpri and asterisk does).
The problem seems to be a symlink issue with the zaptel/Makefile looking for:
Hi,
What do you mean by saying Bearer Capability. Either Speech, or
3.1khzAudio How can I solve the problem ?
Thanks
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To
However: this means that a bug that makes Asterisk consume
100%CPU will practically stall the system. The system will
still answer pings and open sockets, because that is done
in the kernel. But nothing further will be done.
Correct. Same if, for any reason, you have a loop in your dial-plan.
Mark Johnson schrieb:
Jason wrote:
Hey all, I have set up my cisco 30vip using chan_skinny because
chan_sccp wont register. The problem I am having is, everytime a
call is sent to the phone Skinny/[EMAIL PROTECTED] it rings once, then
asterisk segfaults.
Man... Use chan_sccp from
I have connected asterisk 1.0.7 with Avaya Definity via E1 with a
TE100P Digium Card.
Inbound calls are working perfectly and I dont have any problem. But
when I try to make an outgoing call with my softphone (xlite) I am
getting the following messages.
Hungup 'Zap/13-1'
Executing
Hi
how to disable call waiting on SIP User agents
(incominglimit=1 is Deprecated , End of life already
announced
no idea how to use setgroup to achieve same
functionality)
Thanks
Start your day with Yahoo! - make it
On Sat, Aug 13, 2005 at 12:41:15AM -0700, Luki wrote:
However: this means that a bug that makes Asterisk consume
100%CPU will practically stall the system. The system will
still answer pings and open sockets, because that is done
in the kernel. But nothing further will be done.
Correct.
Hi,
I have managed to successfully receive faxes from a fax machine
attached to a Linksys PAP2, and send those faxes off via email using
rxfax (spandsp). From within the same process, I would now like to
automatically send the tiff file onwards as a fax using txfax out via
a zaptel interface. I
Hi all,
I have a cologne chip card which is connected directly to the ntba.
Outgoing and incoming calls work fine, but incoming calls from ntba have
the wrong callerid (first 0 is missing). I'm using current jolly misdn
drivers and chan_misdn-14_04_05 with asterisk stable.
Is anyone seeing this
Hi again,
next problem I have is: I want to write an application which connects
via manager api and displays the current telephone state.
I know I have the action id to identify events which belong together.
But if I have a call going inside asterisk and asterisk rings a phone
these are two
On Sat, 2005-08-13 at 11:59 +0200, Christian Peter wrote:
I know I have the action id to identify events which belong together.
But if I have a call going inside asterisk and asterisk rings a phone
these are two channels with different action ids. How can I know that
these channels belong
Gulzar Hussain wrote:
Hi
how to disable call waiting on SIP User agents
Configure it on the SIP user agent!
(incominglimit=1 is Deprecated , End of life already
announced no idea how to use setgroup to achieve same
functionality)
We will have to change that. Incominglimit has an
Hello,
I have installed SpanDSP and the apps txfax an rxfax. Unfortunately I am
having problems sending faxes. I only get cancelled transfers. I am
trying to send a fax to a ISDN card connected to a zap channel.
I am using following call file:
---
Channel:
Hi,
I plan on setting up an asterisk server to be used as an
email-2-fax/fax-2-email server, for a company that sends and receives
faxes almost 24/7 (milions of fax pages every month).
From your experience in this, can Asterisk handle the heavy load? I
intend to purchase a Saphir V PRI ISDN
Hi
We
are putting some efforts on having asterisk work as a PTT server over GPRS.
Anyone
interested to part of it , Please email me privately
Best
Regards
Mustafa
N. Deeb
___
Asterisk-Users mailing list
Hello,
I have a Incoming/Outgoing SIP Trunk setup to
Broadvoice, is there a way to send a "Flash" over the trunk, for example, to do
flash transfers and call-waiting?
I tried to use Flash() but it seems to not work on
the sip trunk, only my zap trunks.
Please let me know, thanks! :)
On Saturday 13 Aug 2005 07:29, Eric Bishop wrote:
Hi all,
Anyone able to remotely reboot their password protected Sipura
SPA-3000 from command line. I am trying:
Sipura SPA-3000 from command line:
# wget http://admin:[EMAIL PROTECTED]/admin/reboot
The strange thing is it works fine when
On 11:44, Sat 13 Aug 05, Christian Peter wrote:
Hi all,
I have a cologne chip card which is connected directly to the ntba.
Outgoing and incoming calls work fine, but incoming calls from ntba have
the wrong callerid (first 0 is missing). I'm using current jolly misdn
drivers and
Michiel van Baak schrieb:
I have a cologne chip card which is connected directly to the ntba.
Outgoing and incoming calls work fine, but incoming calls from ntba have
the wrong callerid (first 0 is missing). I'm using current jolly misdn
drivers and chan_misdn-14_04_05 with asterisk stable.
This chan_misdn version is old, use a newer one. It seems that
TypeOfNumber interpretation has not been integrated in this verison.
Best regards
Hans
Christian Peter schrieb:
Hi all,
I have a cologne chip card which is connected directly to the ntba.
Outgoing and incoming calls work fine,
Ok, I am going for [EMAIL PROTECTED] with the CentOS iso disk. Installed and
just
checking a few things out.
My other question is this, which I forgot to ask before. We have no
Broadband here and more than likely will never have, so I am just
looking at building Asterisk to handle inbound and
On 14:43, Sat 13 Aug 05, Stefan Gofferje wrote:
Michiel van Baak schrieb:
I have a cologne chip card which is connected directly to the ntba.
Outgoing and incoming calls work fine, but incoming calls from ntba have
the wrong callerid (first 0 is missing). I'm using current jolly misdn
I have successfully configured asterisk to make outgoing calls over FWD,
but cannot receive incoming calls. The console shows no messages,
even though an XTEN client on the same network has no problems receiving
incoming calls.
This is the relevant part of sip.conf
[general]
.
register =
On 15:14, Sat 13 Aug 05, John Fawcett wrote:
I have successfully configured asterisk to make outgoing calls over FWD,
but cannot receive incoming calls. The console shows no messages,
even though an XTEN client on the same network has no problems receiving
incoming calls.
This is the
Tom Rymes wrote:
Chris,
Maybe you could write a generic config file and post it to the wiki?
I tried to post as a comment but the XML was excluded. How do I do that?
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
Any idea, how to send sip messages in a call to any of then if they
are sip or just one is a sip device, i check the code in sendtext but
i dont know how to change the current channel i need to send the
message to the other side, how can i know to whom is connected a
channel and change the code
[ Subject changed so people looking at the list index will actually have
the minimal clue as to what this post is about ].
On Sat, Aug 13, 2005 at 01:50:16PM +0100, Sean Rima wrote:
Ok, I am going for [EMAIL PROTECTED] with the CentOS iso disk. Installed and
just
checking a few things out.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alexandru Thomae
Sent: Saturday, August 13, 2005 7:55 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Fax
Hi,
I plan on setting up an asterisk server to be used as an
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Sean Rima
Sent: Saturday, August 13, 2005 8:50 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] One more newbie question
Ok, I am going for [EMAIL PROTECTED] with the CentOS iso
You may be duplicating work that has already been done.
http://www.zapatatelephony.org/app_rpt_article.pdf
Mark, KC2ENI
Mustafa N. Deeb wrote:
Hi
We are putting some efforts on having asterisk work as a PTT server over
GPRS.
Anyone interested to part of it , Please email me
Couple questions since I finally got chan_sccp to work. 1. Has anyone
successfully gotten it to work behind NAT, i called my isp and asked
them for 35 more ip's and they laughed at me + the cost of getting them
would far outdo the benifits. and 2. can someone show me a config with a
multi
I think I have a bad FXS module on my TDM400P.
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
2 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
The zaptel loads ok, but the wctdm reports
Tzafrir Cohen wrote:
[ Subject changed so people looking at the list index will actually have
the minimal clue as to what this post is about ].
On Sat, Aug 13, 2005 at 01:50:16PM +0100, Sean Rima wrote:
Ok, I am going for [EMAIL PROTECTED] with the CentOS iso disk. Installed and
just
Tom Rymes wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Sean Rima
Sent: Saturday, August 13, 2005 8:50 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] One more newbie question
Ok, I am going for [EMAIL PROTECTED] with
Hot off the wire:
http://www.sineapps.com/news.php?rssid=927
Hi, we have put together a small application for Windows to allow you to
check IAX network statistics.
Basically all you need is the .Net framework and the
user/pass/host/extension/context details.
There is one parameter available
lspci -v what output do you get? Also, what OS are you using?
Jeff Borders wrote:
I think I have a bad FXS module on my TDM400P.
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
2 channels configured.
ZT_CHANCONFIG
Jeff:
Which operating system are you running?
From: Jeff Borders [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TDM400P Card (Rev
Has anybody used a Cisco 7905G or similar model with Asterisk using skinny?
How can i set it up with an asterisk box?
Thanks,
Orlando
___
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Asterisk-Users@lists.digium.com
When executing: Dial (SIP/[EMAIL PROTECTED],60)
I get about 15 seconds of ringing, the called party rings, but if not answered
in the ~15 seconds I get back SIP 480 temporarily unavailable.
If the call is answered everything is fine and the call will
continue as expected.
The call is
Jason wrote:
Couple questions since I finally got chan_sccp to work. 1. Has anyone
successfully gotten it to work behind NAT, i called my isp and asked
them for 35 more ip's and they laughed at me + the cost of getting them
would far outdo the benifits. and 2. can someone show me a config
Orlando Guitián wrote:
Has anybody used a Cisco 7905G or similar model with Asterisk using
skinny? How can i set it up with an asterisk box?
Are you using the latest version of chan_sccp?
http://www.voip-info.org/tiki-index.php?page=chan_sccp2
The driver link can be gotten directly from
Hi all,
I just connected 4 * box (by IAX) and now i'm thinking about this: can i
exchange the extensions list between this boxs ? The clinets/phones can known
which other clients are connected ?
Thanks,
Gio
___
Asterisk-Users mailing list
Hi,
I'm having prolems with attended transfer configuration.
Does this feature had been implemmented by any of you???
What is the best * version to make this work??
Some sample example??
I'm using like this in features.conf:
[featuremap]
atxfer = 900 ;
My linux box speaks pppoe to external DSL modem.
Nortel NTEX35 BAAB. It's up 24/7 and provides
web service...etc.
Has 6 nics, one of them is fiber.
Asterisk is on the same box.
Don't have any IP phones yet.
The asterisk default is to listen on all 6
enet interfaces? (this is what I'd want).
Hello All
i need to transfer CDR data from linux to MS SQL Serever (on Windows). writing by Perl. I have download and install UnixODBC, DBI, DBD from CPAN,
when i tested isql -DSN -UID - PWD, that's successful, but when run by perl, message alert
could not loaded driver database,
anybody
Hi,
I searched a while about T.38 decoding, and learned about the
bounty for T.38 support for asterisk and some softdecoders and
some hardware de- and encoding T.38.
Now I wonder, if there is already any (almost) ready to use solution
for decoding of T.38 faxes?
My szenario would be:
-
On Sat, Aug 13, 2005 at 08:10:03AM -0800, Cliff Savage wrote:
The digium board will be in the same box.
Does this mean:
Channel 4 to incoming phone line.
Channel 1 to DSL modem?
Or DSL modem to the incoming line...and then the pass thru
port on the DSL modem goes to Channel 4?
Will
I plan on setting up an asterisk server to be used as an
email-2-fax/fax-2-email server, for a company that sends and
receives faxes almost 24/7 (milions of fax pages every month).
From your experience in this, can Asterisk handle the heavy load? I
intend to purchase a Saphir V PRI
Has anyone
been able to compile app_rpt?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: Saturday, August 13, 2005 4:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
How do you generate those calls? That's what Im interested about.. I do have
multiple asterisk servers that I can use to send the calls but how to
generate them.. That's the question. :)
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Tim Connolly
Hi Michael.
Are there any script already made for doing this? Sending calls from one
asterisk to the one been tested? Something that would simulate your 1 phone
scenario?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|The VoIP Connection
|Sent:
I just create a bunch of call files and copy (Yes I know you are
supposed to move them) into the outgoing calls directory.
Darren Wiebe
[EMAIL PROTECTED]
Anton Krall wrote:
How do you generate those calls? That's what Im interested about.. I do have
multiple asterisk servers that I can use
On Aug 12, 2005, at 7:06 PM, Tim Connolly wrote: I checked the list of what not to use, but am still having no fun trying to find a working box. Can someone suggest a Compaq or Dell or MPC or … any other brand and model that is known to work well with the TE411P ? Will an old Proliant
Try this:
phone1=192.168.7.251
number1=1+0+1
curl http://$phone1/command.htm?key=$number1+ENTER; /dev/null
2/dev/null
sleep 10
curl http://$phone2/command.htm?key=CANCEL; /dev/null 2/dev/null
Available keys:
#define KEY_CANCEL CANCEL
#define KEY_CLEAR CLEAR
#define KEY_ENTER ENTER
#define
Thanks for the
suggestion. One of my problems is that a TE110P worked flawlessly in my MPC
server. As soon as I upgraded to the TE411P, I started having all sorts of
issues. The biggest being an IRQ conflict, which was resolved but only to find
I still get kernel panics under minor load.
Hi guys,
First, changed the thread to generic x100p always
OnHook since I realized my problem is more general
than to just optimum voice.
My x100p card is always onhook, and that's why it
kills the dialtone whenever I connect it to the phone
line.
Some more in depth debug/status:
zap show
On Sat, Aug 13, 2005 at 01:21:13PM +0700, Zvi Kushnaroff wrote:
Jeff,
You might want to try changig the setup of tzaptel.conf and zapata.conf.
I have a TDM400P with 2 FX0 modules and one FXS module.
I found that the stting that digium recommends are WRONG.
It works fine with the following
Hi Stefan,
thanks for the immediate response.
Luckily I found a fix to my MISDN problem so I don't have to rely on the
channel information.
Thanks,
Christian
Am Samstag, den 13.08.2005, 10:53 + schrieb Stefan Reuter:
On Sat, 2005-08-13 at 11:59 +0200, Christian Peter wrote:
I know I
Hi Hans,
it was there but it doesn't work in this particular version. It works
with current daily snapshot.
Thanks,
Christian
Am Samstag, den 13.08.2005, 14:47 +0200 schrieb Johann Steinwendtner:
This chan_misdn version is old, use a newer one. It seems that
TypeOfNumber interpretation has
Jason schrieb:
Couple questions since I finally got chan_sccp to work. 1. Has anyone
successfully gotten it to work behind NAT, i called my isp and asked
them for 35 more ip's and they laughed at me + the cost of getting them
would far outdo the benifits. and 2. can someone show me a config
Hi All,
I am experiencing a very strange problem. I call the FXO channels (Zap/2 and
3) almost at the same time, and then hang both up. The operator extension is
Zap/6, and after the greeting message Zap/6 starts to ring (there is no
disconnect supervision here, and I disabled the busy detect
[thread moved from -dev due to non-dev content]
At 6:40 PM +0200 on 8/13/05, Andreas Sikkema wrote:
On Sat, 2005-08-13 at 12:44 +0800, Steve Underwood wrote:
He doesn't seem to really understand the strengths and weaknesses of
IAX. IAX has drawbacks, but none of the problems he lists
You do realize that t.38 is the act of taking the t.30 stream and
stuffing into UDPTL packet and sending it over a network with a
little ASN.1 header added and some reliable delivery kinda like how
IAX has reliable delivery of UDP packets used for signaling. This is
a very basic
[moved from -dev list due to non-dev topic content]
At 12:44 PM +0800 on 8/13/05, Steve Underwood wrote:
Mike Taht wrote:
but hey, maybe the folk on this list understand where he's coming
from and can explain why sip is better
He is one of originators of RTP and the main guy behind
what is the best voip provider that provides good
service ,good voice quality and good rates . any one
have an experience with voip providers advice me.
Regards;
jonny
Start your day with Yahoo! - make it your home page
I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO)
answers an incoming call and forwards that call to a SIP softphone (X-lite.)
Seems all is built/installed okay:
# ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves:
Try changing [analog] to this
[analog]
include = test
include = local
exten = s,1,Answer ; Answer the call so we know its getting into *
exten = s,2,Playback(transfer) ; Tell caller pbx is working
exten = s,3,Dial(SIP/1237) ; transfer call to extension 1237
You have not allowed the ZAp
Dear Zaptel and wcfxo devellopers,
Hi, so far I have had no success moving this issue forward. Carl
Andersson has been kind enough to help build various kernels to try,
but with no success.
So, I have tried to debug the problem directly. So far I have applied
the patch below to wcfxo.c. (on
Dear Zaptel and wcfxo devellopers,
Hi, so far I have had no success moving this issue forward. Carl
Andersson has been kind enough to help build various kernels to try,
but with no success.
So, I have tried to debug the problem directly. So far I have applied
the patch below to wcfxo.c. (on
Dave Williams wrote:
I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO)
answers an incoming call and forwards that call to a SIP softphone
(X-lite.)
Seems all is built/installed okay:
# ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS
And btw, I have CentOS 4.1. Could this be related with 2.6 kernel?
Hi All,
I am experiencing a very strange problem. I call the FXO channels (Zap/2
and 3) almost at the same time, and then hang both up. The operator
extension is Zap/6, and after the greeting message Zap/6 starts to ring
hello
i am using asterisk-1.0.9. i have a NAT problem.
without NAT registration is ok. and if user is bhind
NAT it is registring on asterisk. but SJPhone is
showing not registered. i think asterisk is properly
sending request to UA. any commentsthis
sip.conf setting was working
JP Carballo wrote What does the CLI say? Does it show Playing 'value-of-MES' (language 'en')? I'm using 1.0.8 here and I have no problems using A(x) in my dial strings in either ZAP or SIP channels.
Yes, it does say Playing .. (language en)
but there is no sound sent to the called
On Fri, 12 Aug 2005, Matt Florell wrote:
Short answer: NO
Long answer: you have to send it to Digium for them to do an upgrade,
they don't have an official process for this yet and won't give you a
price, I have called and asked them many times. They also mention
upgrades from your 405/410
On Fri, 12 Aug 2005, Bruce Leetch wrote:
Am I banging my head against at Windows/VMware/Linux/Asterisk
incompatibility? Or can this work and I'm just doing something stupid
(always a possibility with me).
It's not going to work. Vmware presents a complete Virtual PC, so unless
EMC / Vmware
Jonathan:
Our provider continue selleing us SPA-841, if you want the contact,
mail me outside the list.
On 8/13/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
Tom Rymes wrote:
Chris,
Maybe you could write a generic config file and post it to the wiki?
I tried to post as a
On Fri, 12 Aug 2005, Tom Rymes wrote:
VMWare is a virtual machine and has nothing to do with the physical
layout of the box (which is why you can migrate vmware images
across machines for example).
If you want to run Asterisk under Linux setup a box to run it.
Agreed. You would
Hi all,
I'm trying to make my cvs STABLE 08/10 srpms build properly on an
updated FC4 box. When I rebuild the srpm with FC4's gcc4 I get this
error:
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O3
I have a repeater using app_rpt, it seems to work just fine.
Quoting Mustafa N. Deeb [EMAIL PROTECTED]:
Has anyone been able to compile app_rpt?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent:
Wich kind of E1 card do you use at the NORTEL ??
it was a PRI one??? witch model ???
On 8/12/05, Mark Phillips [EMAIL PROTECTED] wrote:
Easily doable. I've done it twice now. Problem is that your users will
never know they have messages waiting.
Install a T1/E1 card into the * box and then
Lokesh kumar wrote:
Hi,
If you will put firewall, then i think you will get
high latency and consequently you will hear voice
jitter in your conversation. so avoid putting
firewall.
Is this a troll or what?
Anyway, there is a valid point here so I will address it as if it were
not. The
Rich Adamson wrote:
That's a crack of crap sold by the marketing (not sales) people selling
firewalls. If you know what you're doing, one can very easily secure any
linux system to function on the Internet (etc) without a firewall. It all
depends on your level of knowledge/skills on how to
If you need a FXS, Vonage starts at $15. If you want to simply go soft-only,
Broadvoice would be a better choice. After the marketing and all the
features that nobody uses are thrown out, it comes down to consistency.
Broadvoice has had some problems in the past 6 months, Vonage hasn't (that I
I'm trying to get the vmail.cgi script to work. Followed the
instructions in the wiki, but I'm getting stuck with this error:
Bleh, no /etc/asterisk/voicemail.conf at /var/www/cgi-bin/vmail.cgi line 96.
I chmodded the files and directories used by vmail.cgi per the wiki
instructions, but it
On that note... IPSec tunnels seem to reek havoc on the echo
canceling/training process. Anytime our Cisco PIX loads up, the echo
complaints start coming in. Stay away from the IPSec tunnels.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Travers
Wiley Siler wrote:
The question was not can I secure a Linux box without a hardware
firewall. The question (or statement really) was will a firewall add
jitter and lower performance.
A good firewall architecture w/QoS will actually prevent jitter and
increase performance, I might add.
You might try to su - apache and make sure apache can read the file.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy Vega
Sent: Saturday, August 13, 2005 5:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] vmail.cgi
I'm trying to
What is the optimum audio format and quality, codec, etc for using to play
voice prompts in Asterisk?
BTW - I am a Windows user, and about to record some prompts.
Thanks
Sam
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TC ADSL should not bother PSTN as long as you use a proper filter. In our
TC case a proper filter was supplied by the phone company when we installed
TC the ADSL line. We Happily use Asterisk with an FXO card and an ADSL
TC connection from the same phone line.
Got a leviton DSL filter mounted
Hi; I've been using Asterisk for a few months now, and I have run into
an interesting issue that I thought someone else in the community may
have run into:
I have an Asterisk install set up to receive helpdesk calls, route
them to several IAX extensions and an extension which is simply a
Hello Everyone:)!,
I have a Incoming/Outgoing SIP Trunk setup to Broadvoice, is there a way to
send a Flash over the trunk, for example, to do flash transfers and
call-waiting?
I tried to use Flash() but it seems to not work on the sip trunk, my
configuration is as follows:
exten =
This works. I've done it on occasion for testing. However, because
virtual
PCs rarely operate on a real-time clock, mostly emulating these
features,
you will find that anything that read/writes to disk will suck badly.
For
example, it is nearly impossible to use the Voicemail features of
At firewall/NAT you have to do port forwarding.
If your phone is at port 5060, NAT device will receive a connection and has
to know that it is destined for your SIP phone. So, forward port 5060 to the
phone.
Rudolf
- Original Message -
From: Kamran Ahmad [EMAIL PROTECTED]
To:
Rudolf Ladyzhenskii wrote:
Hi, all
I am running asterisk and my friends are running FireFly IAX phone. All
is fine except one of them. When anyone tries to talk to him, tehre is
a real bad echo. It is nothing to do with sound setup.
Is he using a headset or speakers and microphone?
Does
Now that we have a well functioning Asterisk system that queues our
calls and distributes them to our CSRs, I would like to implement a
better system for our agents to keep a log of all of their calls, which
we currently do using MS Word. (As you would expect, this is a less than
ideal solution!)
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