Nedi [EMAIL PROTECTED] writes:
I am new to asterisk. My system is ASTLINUX
if receive a Fax on my sipura spa2000
i get this: Feb 25 07:41:00 NOTICE[1708]: rtp.c:564 ast_rtp_read: Unknown RTP
codec 100 received
Probably point 6 on
Cory Andrews wrote:
Has anyone tried the Linksys SRW224P? 24 Port managed switch, 10/100, 2
Gig Uplink Ports, PoE:
a.. Delivers reliable power over 10/100 Ethernet ports using IEEE
802.3af standard
b.. Secure management via SSH/SSL and secure user control via 802.1x
MAC filtering
c.. IGMP
On 02/24/06 10:13 Time Bandit said the following:
unless you client call isn't coming on a zap channel. In that case,
you should look here :
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanSpy
is there any reason why chanspy cant be used consistently for all channels
instead of
On 02/23/06 23:08 Darrick Hartman said the following:
True, but why not accept the app? It sure makes the dial plan alot
nothing wrong with that, i wasnt suggesting rejecting the application or
anything. just pointing out that scripting it within the dialplan makes it
more flexible for
On Wednesday 22 February 2006 23:23, Ben Klang wrote:
This is a known issue. A few patches have been proposed, but nothing
official has been adopted. If you are interested in testing my proposed
solution to the problem check out bug 6334 at
http://bugs.digium.com/view.php?id=6334
Hi,
Steven Andres wrote:
I just tried this out and it didn't work, Steve. Using:
exten = 700,1,NoOp(Park and Announce)
exten = 700,n,Set(REFBY=${SIP_HEADER(Referred-By)})
exten = 700,n,NoOp(Referred-By: ${REFBY})
exten =
I'd like to be able to use my Snom 360 LEDs to view the status of
parking slots, so I'm trying to install the metermaid patch
(http://bugs.digium.com/view.php?id=5779). Can someone help an svn
newbie figure out how to install this patch? I've done the following:
svn checkout
Does anybody know how to set polycom's default ring volume ? Everytime you
restart a polycom phone, ring defaults to a very low volume setting which is
kind of annoying having to set everytime you reboot.
Any hints?
___
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Why is iaxmodem with hylafax more stable than spandsp?
Can you run iaxmodem and hylafax together with spandsp (for running E1
r2mfc)?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Rob Danz
|Sent: Friday, February 24, 2006 9:34 AM
|To: 'Asterisk
That's what I imagined. I read somewhere that echocancel kills faxes.. Also,
I guess hardward cards with echo cancel modules are a nono :)?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Lee Howard
|Sent: Friday, February 24, 2006 10:45 AM
|To:
I know you can login agents to a queue in several ways, for example,
agentcallbackllogin, or by editing agent.conf and adding agents there with
password.
But I was wondering if there is a way to have an agent always on a queue?
___
--Bandwidth and
I have an asterisk pbx conected to internet. I need to connect to
asterisk a sip phone over the net that will be connected to the
internet over a NAT router. I found on the net that SIP is not very NAT
friendly. If somebody has a how-to for my problem please share it
with me . Also any experience
Inline...
Interesting,
So are there any sort of specifications to look for? What your talking
about does not sound like a managed vs unmanaged issue. More like cheap
crap vs half decent. I would never want any switch to drop packets VoIP or
not. Does not sound like QoS could help
On Mon, Feb 20, 2006 at 12:05:47PM -0300, Melcon Moraes wrote:
Anthony Azzopardi wrote:
Where can I get the tar.gz sources of libnewt?
I think this would help you.
http://packages.debian.org/unstable/perl/libnewt-perl
Hmmm, you probably meant something like
On Wednesday 18 January 2006 23:35, you wrote:
Hello,
I have a problem with an LAN-Server behind an NAT-router.
Asterisk Version 1.2.1 or 1.2.2 doesnt matter
10 minutes after starting Asterisk I loose all registrations at external
SIP-proxys.
The reason seemed to be that Asterisk send
Anton Krall a écrit :
Why is iaxmodem with hylafax more stable than spandsp?
Can you run iaxmodem and hylafax together with spandsp (for running E1
r2mfc)?
You're mixing thinks: iaxmodem+hylafax is equivalent to rx_fax/tx_fax,
both are based on spandsp which is the library.
Paul Lacatus wrote:
I have an asterisk pbx conected to internet. I need to connect to
asterisk a sip phone over the net that will be connected to the internet
over a NAT router. I found on the net that SIP is not very NAT
friendly. If somebody has a how-to for my problem please share it with
Ron McCarthy wrote:
Hi List,
Im planning on setting up asterisk for a large scale enviorment, with
multiple sites.
We will be doing quite a bit of inner office calling at each site, and
want to place a smaller scale * box at each site with no PRI's, and have
that connect to our main *
Are you sure the signalling is right, play around with other signalling
types and see. also play around with crc on and off
Anthony Rodgers wrote:
Are you sure you're supposed to be using EM?
On Feb 24, 2006, at 5:39 AM, Nitin Joshi wrote:
Hi All,
I have installed a Digium TE110P card on
If memory serves correctly, I believe the echo canceller is automatically
turned off when fax tones/negotiation occurs on a zap channel.
That's what I imagined. I read somewhere that echocancel kills faxes.. Also,
I guess hardward cards with echo cancel modules are a
So, why/how is iaxmodem/hylafax more sucessful in receiveing faxes thru tdm
than rxfax?
I havent been able to get faxes with rxfax, all faxes come in as garbage or
broken or just the first page.
Im hoping and placing my bet on iaxmodem.
|-Original Message-
|From: [EMAIL PROTECTED]
Also one question. Do you still have to play with gains when using iaxmodem
as you need to when using rxfax?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Administrator TOOTAI
|Sent: Saturday, February 25, 2006 7:07 AM
|To: Asterisk Users Mailing
Hi,
Sorry for being very late on this thread but i am trying to make a
decision on which one to go for. Options are
1. Dock n Talk offered by Voxilla (USD139)
2. GSM Gateway by CyberTelecom (GBP60)
I'm having a TDM400P with 1 FXO FXS.
I'm interested in implementing DISA my [EMAIL PROTECTED]
I've been using vsftpd since fedora core 2. There was a period of time
in FC3 when linux wouldn't let me create usernames with capital
letters. No biggie though I just created the user with all lowercase
and then went back and edited the /etc/passwd file to change the
username to the
Michael Welter wrote:
I'm having difficulty with an Asterisk system. The external party has
very good call quality, but the internal party hears clipping and drop
outs.
RX Gains too high
IRQ sharing of the of the ZAP device
High load of the machine
Are a few that come to mind.
Doug
--
Has anyone tried the Linksys SRW224P? 24 Port managed switch, 10/100, 2 Gig
Uplink Ports, PoE:
a.. Delivers reliable power over 10/100 Ethernet ports using IEEE 802.3af
standard
b.. Secure management via SSH/SSL and secure user control via 802.1x MAC
filtering
c.. IGMP snooping,
If you need to play with gains to make rxfax work something is broken in
your setup. It could be your echo canceller is not turning off as the
fax begins. fax modems normally work just fine over a wide range of
signal levels.
Steve
Anton Krall wrote:
Also one question. Do you still have
Mahilal Silva wrote:
Mike,
Were you able to get this working?
Even after with a entry in the dialplan.xml does not work for me.
Thanks,
This is what I have in my dialplan.xml
DIALTEMPLATE
TEMPLATE MATCH=*Timeout=5/ !-- Anything else --
TEMPLATE MATCH=*78
Hope this helps someone else, I found that
the below problem is a bug. Here's the info on it...
Disable SELINUX
vi /etc/selinux/config
SELINUX=disabled
At next release, can the install script be
changed to look
for SELINUX being active or make the
software work under
SELINUX.
Hello again,
Well all out * boxes will be on a channelized DS3 running back to the
main * server, so I guess we could actaully have it be extra smart and
have it route directly to the other * box (not the main one) if it
needs to make a inter-branch call it can. Ill have to look around into
DUNDI
context logic for big systems can be tricky if you dont use AGI,
FastAGI or something similar. What are you using to administrate your
contexts and extensions? Hand editing?? I still dont see the problem,
its just matter of what permissions you want the
transferer/transferee to have, if you want
Sorry for bumping this up (
http://lists.digium.com/pipermail/asterisk-users/2006-February/148059.html )
Any ideas, please?
TIA,
Alex
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To UNSUBSCRIBE or update
So, why/how is iaxmodem/hylafax more sucessful in receiveing faxes thru tdm
than rxfax?
I havent been able to get faxes with rxfax, all faxes come in as garbage or
broken or just the first page.
Im hoping and placing my bet on iaxmodem.
The weakest link in what you're trying to do is
nat=yes
qualify=yes
That works, but it works better if you use a NAT/firewall box that can
do VOIP transformations automatically. The Sonicwall TZ170 can do
this. It rewrites the packets auto-magically so things just work. The
above parameters can be set to no then.
It seems to work more
Anton Krall wrote:
I know you can login agents to a queue in several ways, for example,
agentcallbackllogin, or by editing agent.conf and adding agents there with
password.
But I was wondering if there is a way to have an agent always on a queue?
How about an entry like this in
I have one question,
How does a large file transfer like your excel spreadsheet example, affect
communication between an Asterisk server and SIP phone? The only possible
configuration I can think of that would cause a problem is if the client PC
is sharing the same eternet cable and therefore
Doug Lytle wrote:
Michael Welter wrote:
I'm having difficulty with an Asterisk system. The external party has
very good call quality, but the internal party hears clipping and drop
outs.
RX Gains too high
IRQ sharing of the of the ZAP device
There is no ZAP device (it is a SIP-only
Hi there,
I was wondering if anyone has successfully used Asterisk as a dedicated
Analog PSTN gateway to take the place of, for example, a Mediatrix 1204 or
an 8 port model?
Basically, I am thinking of using a Linksys SPA9000 as the PBX and just need
an Analog PSTN gateway for 4 to 8 FXO
Michael Welter wrote:
Doug Lytle wrote:
Michael Welter wrote:
The machine is totally idle.
The T1 vendor noticed 2% packet loss during a ping flood originating
from outside. We changed the Cisco IAD, and there is no longer packet
I've noted from employees that the volumes levels on the
On 2/11/06, Zach A [EMAIL PROTECTED] wrote:
Hi everybody,
I have an Asterisk box and I want to install just ARI on it for
monitoring the calls. Installing [EMAIL PROTECTED] utilizes too much
resources and
memory and also takes away freedom of configuration asterisk. I like
using asterisk
How does a large file transfer like your excel spreadsheet
example, affect communication between an Asterisk server and
SIP phone? The only possible configuration I can think of
that would cause a problem is if the client PC is sharing the
same eternet cable and therefore the same
mustardman29 wrote:
I have one question,
How does a large file transfer like your excel spreadsheet example, affect
communication between an Asterisk server and SIP phone? The only possible
configuration I can think of that would cause a problem is if the client PC
is sharing the same eternet
...or if your
asterisk server is also a file server (which should never be
done)
I know I'm attracting flames for disagreeing, but sometimes when you're
dealing with small business customers there simply isn't the budget to have
separate machines for doing x, y and z, and often one finds the
I could not find followme app listed when I tried show applications
on the CLI. Is this app patch incorporated into asterisk 1.24 release
tree? If not, what are the plans for the future?
On 2/24/06, Dinesh Nair [EMAIL PROTECTED] wrote:
On 02/23/06 23:08 Darrick Hartman said the following:
Doug Lytle wrote:
Michael Welter wrote:
Doug Lytle wrote:
Michael Welter wrote:
The machine is totally idle.
The T1 vendor noticed 2% packet loss during a ping flood originating
from outside. We changed the Cisco IAD, and there is no longer packet
I've noted from employees that the
Can this be done? I havent seen to much of this on the
mailing list, im guessing each server would talk to the main
* server via a IAX trunk or a SIP peer. Also one other key
point would then be to keep the voicemail for each office on
its local * server instead of having it go to the
Why I would want to use the SPA9000 as the core PBX instead of asterisk is a
topic for another thread. Basically, I think it's a better solution for
some smaller businesses that need BLA and something simple that JUST WORKS.
No hard drives, no fans, no linux configuration. If the auto
I've noticed some other odd thing with rxfax. In my case I can receive faxes
(using TDM400P) just fine. I can only see those faxes using Windows XP's
Fax and Picture thingy, other applications are having trouble. Also
printing those faxes is a bit odd: the preview is just fine but I always
need to
[EMAIL PROTECTED] is believed to have said:
Hi,
Sorry for being very late on this thread but i am trying to make a
decision on which one to go for. Options are
1. Dock n Talk offered by Voxilla (USD139)
2. GSM Gateway by CyberTelecom (GBP60)
I'm having a TDM400P with 1 FXO FXS.
I'm
On 2/25/06, Nilesh Londhe [EMAIL PROTECTED] wrote:
I could not find followme app listed when I tried show applications
on the CLI. Is this app patch incorporated into asterisk 1.24 release
tree? If not, what are the plans for the future?
On 2/24/06, Dinesh Nair [EMAIL PROTECTED] wrote:
Yep, I tried it both ways (I should have put that in the email). Hmm. I did
a SIP debug and indeed I did see the Referred-By header in the debug trace
so it's probably something wrong with me (misspelled?). I'll work on it a
bit more. One thing to note--in your example you have the
Cosmin Prund wrote:
I've noticed some other odd thing with rxfax. In my case I can receive faxes
(using TDM400P) just fine. I can only see those faxes using Windows XP's
Fax and Picture thingy, other applications are having trouble. Also
printing those faxes is a bit odd: the preview is just
Chris Bagnall wrote:
...or if your
asterisk server is also a file server (which should never be
done)
I know I'm attracting flames for disagreeing, but sometimes when you're
dealing with small business customers there simply isn't the budget to have
separate machines for doing x, y and z, and
I know this is a OT but great article
http://www.theregister.co.uk/2006/02/23/rwanda_terracom/
Will be interesting to see how this project goes.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
On 2/25/06, Chris Bagnall [EMAIL PROTECTED] wrote:
It's a fascinating thread, this.
So, for all the criticism, I'll continue using cheap switches, recycled
Chris, I mostly agree.. In Europe a 'small' business often only counts
2 - 5 persons. When the budget doesn't allow it, the only way one
Am Saturday 25 February 2006 19:38 schrieb Steve Underwood:
Cosmin Prund wrote:
I've noticed some other odd thing with rxfax. In my case I can receive
faxes (using TDM400P) just fine. I can only see those faxes using Windows
XP's Fax and Picture thingy, other applications are having trouble.
Thomas Artner wrote:
Am Saturday 25 February 2006 19:38 schrieb Steve Underwood:
Cosmin Prund wrote:
I've noticed some other odd thing with rxfax. In my case I can receive
faxes (using TDM400P) just fine. I can only see those faxes using Windows
XP's Fax and Picture thingy, other
Michael Welter wrote:
I'm not on site, but I remember 1.6.4.
I had in place 1.6.2, and had way to many problems with it. I reverted
back to 1.5.2 and things cleared up.
Is the phone (or Asterisk) performing echo suppression that drops the
last part of the tone?
I believe the phone
Doug Lytle wrote:
I think the only time you need a timing source is if you are mixing
audio streams, i.e. meetme, MOH. In which case you'd probably need to
run ztdummy.
Yes , ztdummy is running.
I'm going to (temporarily) put a TDM card in the system just to
eliminate that
I have one question,
How does a large file transfer like your excel spreadsheet example, affect
communication between an Asterisk server and SIP phone? The only possible
configuration I can think of that would cause a problem is if the client PC
is sharing the same eternet cable and
How does a large file transfer like your excel spreadsheet
example, affect communication between an Asterisk server and
SIP phone? The only possible configuration I can think of
that would cause a problem is if the client PC is sharing the
same eternet cable and therefore the same
Steve
I have tried using r2mfc with E1 and also using TDM over analog. I dont use
hardward echo cancel and my settings on zapata are:
[channels]
language=sp
signalling=fxs_ks
usecallerid=yes
cidsignalling=bell
cidstart=ring
hidecallerid=no
callwaiting=yes
usecallingpres=yes
;sendcalleridafter=1
Well, if anybody has fax over tdm working... Im intere$ted :)
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Rich Adamson
|Sent: Saturday, February 25, 2006 9:41 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE:
Looks good, Ill give it a try, Ive never used member with technology...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Andres
|Sent: Saturday, February 25, 2006 10:32 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re:
Steve, I was trying to convert tiff images to pdf, but seems something is
broken somewhere in my setup as you mentioned becuase pages come ou t white
and blank...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Steve Underwood
|Sent: Saturday,
I cant get faxes right now with tdm, something is wrong but, what do I need
to have in order to convert from tiff to pdf?
I have the mailfax script that invokes tif2ps and ps2pdf but pages come out
blank..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On
Am Saturday 25 February 2006 22:59 schrieb Anton Krall:
I cant get faxes right now with tdm, something is wrong but, what do I need
to have in order to convert from tiff to pdf?
I have the mailfax script that invokes tif2ps and ps2pdf but pages come out
blank..
I do the following:
exten =
On Sat, 2006-02-25 at 17:19 +, Chris Bagnall wrote:
...or if your
asterisk server is also a file server (which should never be
done)
I know I'm attracting flames for disagreeing, but sometimes when you're
dealing with small business customers there simply isn't the budget to have
Whats mpack tom?
I use sendEmail..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thomas Artner
|Sent: Saturday, February 25, 2006 4:25 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] fax receive
Hi,
Anyone here using sipgate.de ?
It worked for months, but for a couple of days now I'm
unable to register with them.
My account is ok, because I can login to the website.
Asterisk keeps showing me:
Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout:--
Registration for '[EMAIL
On Saturday 25 February 2006 23:56, Michiel van Baak wrote:
Hi,
Anyone here using sipgate.de ?
It worked for months, but for a couple of days now I'm
unable to register with them.
My account is ok, because I can login to the website.
Asterisk keeps showing me:
Feb 25 23:50:18
The above concern have been a major issue with telephone equipment (eg,
central
offices) and the telco's spend a significant amount of money burying very
long
rods in the ground and interconnectng them with the CO hardware using cables
that are larger then 1/4 in diameter (don't
On 00:41, Sun 26 Feb 06, Thomas wrote:
Did you update your Asterisk?
Actually I downgraded from SVN-TRUNK to SVN-1.2
I dont have problems with sipgate. Iam using an different structure what
makes
it more clear. Also there is no need to use type = friend for an external SIP
Provider.
I have the following in my extensions.conf
exten = s,4,MixMonitor(${DATETIME}_${CALLERID}_${EXTEN}.wav)
The ${EXTEN} works correctly, however the ${DATETIME} (I have tried
${TIMEDATE} too) and ${CALLERID} (tried with ${CALLERID(num)}) doesn't.
There just isn't any information assigned to those
Hey everyone,
I know this is a problem with mpg123, but it just started happening and
I have no idea why. I haven't changed any of the audio format settings
yet. Before tonight, I was able to call, listen to the queues, hear the
music on hold, no problems. I added a new context to a dial
I know this is a problem with mpg123, but it just started happening and
I have no idea why. I haven't changed any of the audio format settings
yet. Before tonight, I was able to call, listen to the queues, hear the
music on hold, no problems. I added a new context to a dial plan,
Hey Rich and everyone.
I tried what you suggested, and it didn't work. I even recomplied
everything, moved all of my configuration files out and remade the
samples, so as far as I can tell everything is back to day 1. However,
it is still pulling in the database information. This is really
Hi,
Sorry for being very late on this thread but i am trying to make a
decision on which one to go for. Options are
1. Dock n Talk offered by Voxilla (USD139)
2. GSM Gateway by CyberTelecom (GBP60)
I'm having a TDM400P with 1 FXO FXS.
I'm interested in implementing DISA my [EMAIL PROTECTED]
Hi Aldo,
I was trying to compare the 2 equipments to use with my asterisk.
1. Dock and Talk has bluetooth which will help me use my mobile itself
to connect.
2. Using CyberTelecom Gateway i dont need a phone equipment to support
which is also a positive thing as my intenstion is just to connect
Hello Dan
I can assure you that our GSM Gateway quality is absolutely excellent and
this fact can be supported by hundred if not thousand of our users.
It is also very simple to use and even a newbie can set it up..
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Nice!
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thomas Artner
|Sent: Saturday, February 25, 2006 6:03 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] fax receive using TDM400P
|
|Am Saturday 25
I know some (most?) of you will say this is wrong but...
When using rxfax my faxes get generated in a /fax folder and that folder is
shared using samba :-) It works sooo nice! If I could only get myself to
trust rxfax so I can free the FXS port for some other duty!
-Original Message-
Windoes'es DEFAULT image viewer also has problems showing rxfax-generated
TIFF's. They do show up properly on screen but when printed the orientation
needs to be changed. I don't know if MS's viewer is somehow broken OR the
tiff is somehow broken (also I don't care) but viewing rxfax-ed tiffs is
Hi again,
Kind of sheepish about asking for help, as I have only spent a day
banging my head off this...
I got my new Welltech 3701a, 1FXS,1FXO gateway.
I flashed it with what is seemingly the appropriate firmware (SIP
V1.04). This seems to have gone ok, and it is now registering both
On Feb 25, 2006, at 10:04 PM, Anton Krall wrote:
Nice!
Did you ever think about trimming your messages?
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