Hi all!
I am frustrated.
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
___
--Bandwidth and Colocation p
I am using asterisk CVS 10842 and a TDM 400p withanfxs and an fxo
module and when I dial the fxs channel it rings for a second and then
says no answer after 20 seconds. I also have the latest Zaptel drivers.
Here is a log snippet.
Feb 25 00:53:11 VERBOSE[2015] logger.c: -- Executing
Dial("SIP
On Fri, Feb 24, 2006 at 03:12:17PM +0100, [EMAIL PROTECTED] wrote:
> Hi all, this is another post about this problem.
> I installed from scratch a new Suse Linux 10.0, with latest stable
> asterisk.
> Moreover I add the lines to /etc/udev/rules.d/50-udev.rules, in order to
> let the driver create
Hi Matt -
> I have the same problem. I'm running CentOS, which comes
> with vsftpd, do you know of anyway to do this using vsftpd?
I know what you mean. I run TaoLinux on all our * machines, so they all
had vsftpd installed with the OS. I had to replace it with ProFTPd
because I just couldn'
Just to through another hat in the ring... I use madplay for mp3s...
[default]
mode=custom
directory=/var/lib/asterisk/mohmp3
application=/usr/bin/madplay -Q -o raw:- --mono -R 8000 -a -12
On Thu, 2006-02-23 at 15:23 -0600, Aaron Daniel wrote:
> I'd suggest using the format_mp3 program that's in
Queus seem to work different between IAX and ZAP channels. I have a IAX
trunk today which I want to switch over to ZAP lines. I have a menu
option for a caller to be able to "locate" as person. What happens is a
queue is setup with a ringroup in it. The ringroup is an external # with
a 15 secon
Zach A wrote:
>Hi everybody,
>
>This question is confusing me for some time. From selling point of view
>to a customer, calling asterisk a PBX doesn't look right. According to
>the definitions of PBX or PABX, Asterisk is not just PBX but much more
>than that. My question is, how should I introduce
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: ${REFBY})
exten =>
700,n,ParkAndAnnounce(pbx-transfer:PARKED|20|SIP/${REFBY:5:5}|office,${EXTEN},1)
The Referred-By
Yes, our phones are registered to Asterisk and are all GXP-2000's (no matter
what everyone says about how bad they are--we love 'em). :)
I wasn't aware of the "REFBY" header. I will tinker with that. Sounds like
that would solve the proposed 'p' option. Now I just need a method to do
what my pr
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, L2/L3 COS,
On 2/24/06, Philip Edelbrock <[EMAIL PROTECTED]> wrote:
>
> Philip Edelbrock wrote:
> >
> > Whoo hoo! I just received my WIP300 from voipsupply. I have to let it
> > charge before I can play with it.
> >
>
> After it charged and I started using it, I had three crashes. Once
> during a call (exac
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 resolve that or jitter
I have the same problem. I'm running CentOS, which comes with vsftpd,
do you know of anyway to do this using vsftpd?
Thanks,
Matt
Noah Miller wrote:
Hi Again Damon -
I just remembered that the FTP server setup can be tricky, too. The default
username has capitalized letters, and this do
Philip Edelbrock wrote:
Whoo hoo! I just received my WIP300 from voipsupply. I have to let it
charge before I can play with it.
After it charged and I started using it, I had three crashes. Once
during a call (exactly 3 minutes into it, according to the frozen
display), and twice whil
Steven Ringwald wrote:
I am trying to use ImportVar to get some information out of a SIP/ZAP
channel. I cannot seem to find an example of the syntax, or what
variables I can access.
Basically, I would like to output which person is being called. i.e:
SIP/25 calls SIP/21. 25 executes a macro,
Steven:
I'm assuming your using IP phones registered to Asterisk in this
example. I don't do that but I use ParkandAnnounce for IP phones
registering to a SER server. To handle the call back part of your
question I use a snipet like:
exten => _700,5,SIPGetHeader(REFBY=Referred-By)
exten =>
> > It's stupid. Don't ever connect 2 different building with copper.
> > Just wait until you get some kind of lightening hit or electrical
> > fault, but make sure you are no where near it. Use fibre.
>
> That's a great rule of thumb, but the reality isn't quite so black and white.
>
> A direct
Hi,
we've connected Sphinx4 through eagi script (modified eagi example) to
Asterisk. Users can now say their wishes - but for gradual evolution we
would like to provide "older" way of DTMF navigation too - can we recognize
DTMF while reading sound in eagi ?
Any advice or examples ?
Thanks in ad
Hello All, I have installed asterisk at home and it is behine firewall. I have done all the changes.I have opened all the required firewall ports. I am able to dial but when I dial to some extension on the internet I get the phone ringing but when the phone is picked up there is no sound. I hav
Does anyone know if the SIP firmware for the cisco 7940/7960's support
subscriptions to lines for hinting? Basically we're trying to set up
secretary phones and I'm seeing a lot of info about how the phone has to
subscribe to a line to do it, but not sure how/if you can do it on the
Cisco's.
We've had a regular Park function in the past but recently I found the
ParkAndAnnounce() application and I love the idea behind it. Here's a snip
from the wiki
(http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce)
so that we're all talking the same language:
|| ParkAndAnn
I'm trying to
compile Asterisk 1.2.4 on a Redhat Enterprise system, kernel
2.4.21-27.0.2.ELsmp
I'm getting the
following errors and then the compile stops.
/usr/kerberos/lib/libgssapi_krb5.so.2: undefined
reference to `add_error_table'/usr/kerberos/lib/libgssapi_krb5.so.2:
undefined refe
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 * servers at our data center that
Hello list,
Is it possible to send polarity reversal signalling on FXS lines in Asterisk using the TDM240X cards?
I need to send that signalling to legacy PABX using TDM boards of my Asterisk box.
The unique clue I could find was:
http://lists.digium.com/pipermail/asterisk-users/2004-Decembe
Matt wrote:
> I too have noticed this but received no solution =\ I was running 1.2.0
Did you try it again after updating to the latest 1.2 release? Did you
report the bug on the bug tracker and provide a backtrace so someone
could try to fix it?
If not, how did you expect a solution to be creat
I was thinking smaller... just Communications Suite
Michael Collins wrote:
Hi everybody,
This question is confusing me for some time. From selling point of
view
to a customer, calling asterisk a PBX doesn't look right. According to
the definitions of PBX or PABX, Asterisk is not just PBX bu
Yes, it is a PBX. It is also a softswitch. It is also a VoIP or PSTN proxy.
It is also a messaging application platform. A fax server. An SMS relay
point. etc. It is all of this, and more. Taken as a whole, Asterisk is
greater than the sum of it's parts.
This also makes it impossible to pigeonhol
Am Friday 24 February 2006 21:17 schrieb yrving rivas:
> Thomas: does it work in your case?
> Do anybody have the fax working w/tdm?
>
Yes, as I wrote before, receiving faxes with a tdm400p card works perfectly!
> Thomas Artner <[EMAIL PROTECTED]> escribió:
>
> Am Friday 24 February 2006 16:
I too have noticed this but received no solution =\ I was running 1.2.0
On 2/24/06, Dov Bigio <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I had 3 users spying on a call from the queue.
> On the exact time that the 4th user called the ChanSpy extension, Asterisk
> went down!
>
> Is there something wron
> Nitin Joshi wrote:
> > Hi All,
> >
> > I have installed a Digium TE110P card on an Asterisk 1.2.1 system.
Its
> > connected directly to the PSTN. But I am unable to make outbound
calls
> > on the zap channels. The light on the card is green. Asterisk CLI
> > shows all 24 channels when I give the
> Hi everybody,
>
> This question is confusing me for some time. From selling point of
view
> to a customer, calling asterisk a PBX doesn't look right. According to
> the definitions of PBX or PABX, Asterisk is not just PBX but much more
> than that. My question is, how should I introduce Asterisk
Thomas: does it work in your case? Do anybody have the fax working w/tdm?Thomas Artner <[EMAIL PROTECTED]> escribió: Am Friday 24 February 2006 16:48 schrieb Anton Krall:> Any modification made to zapata as far as echo and gains?>> Should echocancel be on or off?i have echocancel switched on, fax
Zach A a écrit :
Hi everybody,
This question is confusing me for some time. From selling point of view
to a customer, calling asterisk a PBX doesn't look right. According to
the definitions of PBX or PABX, Asterisk is not just PBX but much more
than that. My question is, how should I introduce
Sorry Olle, I bet you wanted this from the SIP Proxy:)
Chris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, February 24, 2006 7:53 AM
To: Asterisk Developers Mailing List
Cc: asterisk-users@lists.digium.com
Subject: Re: [a
Hi,
I had 3 users spying on a call from the
queue.
On the exact time that the 4th user called the
ChanSpy extension, Asterisk went down!
Is there something wrong with
ChanSpy???
Thank you
Dov
___
--Bandwidth and Colocation provided by Easynews
Hi everybody,
This question is confusing me for some time. From selling point of view
to a customer, calling asterisk a PBX doesn't look right. According to
the definitions of PBX or PABX, Asterisk is not just PBX but much more
than that. My question is, how should I introduce Asterisk to a
custom
Sorry Olle, I bet you wanted this from the SIP Proxy:)
Chris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, February 24, 2006 7:53 AM
To: Asterisk Developers Mailing List
Cc: asterisk-users@lists.digium.com
Subject: Re: [a
I have included two files, one from asterisk 1.0.10 and one from 1.2.4.
Thanks Olle
Chris Modesitt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, February 24, 2006 7:53 AM
To: Asterisk Developers Mailing List
Cc: asteris
I have included two files, one from asterisk 1.0.10 and one from 1.2.4.
Thanks Olle
Chris Modesitt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, February 24, 2006 7:53 AM
To: Asterisk Developers Mailing List
Cc: asteris
We're using a TE205P. lsmod indicates that it's using the wct4xxp driver.
Hope this helps; I'll give it a try with disabled vpmdtmf.
Matt.
C F Wrote:
what zap device are you using?
IIRC disalbing the vpmdtmf on a 406 or 411 might help you. I think
it's done in wctxx4p.c
On 2/24/06, Matt K
Steve,
You wrote this referring to monitoring a call in Asterisk, how about from an
IP phones LCD display screen:
>1. go to www.google.com
>2. type "asterisk monitor application"
>3. click on the first result
>4. read and implement
>5. google is your friend
>
>I hope I made myself clear too
I am trying to use ImportVar to get some information out of a SIP/ZAP
channel. I cannot seem to find an example of the syntax, or what
variables I can access.
Basically, I would like to output which person is being called. i.e:
SIP/25 calls SIP/21. 25 executes a macro, and the result is SIP/21
what zap device are you using?
IIRC disalbing the vpmdtmf on a 406 or 411 might help you. I think
it's done in wctxx4p.c
On 2/24/06, Matt King <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I'm posting this to the list in case others run into the same issue.
>
> I've recently been connecting * to
Hi,
I support multiple context on one asterisk server. I have a situation
where there is a spa that has seperate voicemail and extensions and a
resturant on the same campus that has different extensions and
voicemail. They both use the same asterisk server but I do need the
ability to transfe
Hi,
On the general section of my sip.conf I had a
disallow=all.
Then I put disallow=all, allow=g729, allow=ulaw on
my users.
It didn't work until I removed the disallow=all
from the header.
I know disallow=all in the header is totally
useless in this case (since I have it for every u
Hi,
We're having problems dialing out to Asterisk from
our Grandstream GXP-200 phones. About 2 of 3 times, when we dial, nothing
happens. Looking at the console in max debug mode, there are no messages
except the following:
Feb 24 10:29:20 WARNING[2475]: chan_sip.c:1208
retrans_pkt: Ma
Hello,
I'm posting this to the list in case others run into the same issue.
I've recently been connecting * to a legacy Avaya InDEX switch over
E1 ISDN PRI here in the UK. Everything was working OK, except that DTMF
digits were not being recognised by * when sent by the Avaya switch t
it seems im not undestanding your question then. Could you provide a practical example?On 2/24/06, Chuck Bunn <
[EMAIL PROTECTED]> wrote:Hi,Okay but then how do you transfer across contexts then?
ThanksMoises Silva wrote:> you need to set a TRANSFER_CONTEXT, either for the transferer or> transferee
> Aha, micro seconds in networking terms is normally written usecs or us
> (actually it's the greek letter mu as in ulaw) rather than ms which are
> milliseconds seconds - what had me puzzled was that it was stated that this
> could harm the voice path!
>
> > The difference can also cause unneces
Hi,
Okay but then how do you transfer across contexts then?
Thanks
Moises Silva wrote:
you need to set a TRANSFER_CONTEXT, either for the transferer or
transferee channel. I dont know why, but res_features give priority to
the transferee TRANSFER_CONTEXT, if not found, then use the transfere
Are you sure you're supposed to be using E&M?
On Feb 24, 2006, at 5:39 AM, Nitin Joshi wrote:
Hi All,
I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its
connected directly to the PSTN. But I am unable to make outbound calls
on the zap channels. The light on the card is g
Some more recent phones have the possibility to be connected to seperate
GSM-boxes. E.g. there is a plug-in for the (older) Nokia 9210(i)/9290(i)
Communicators and most of the Symbian phones with Bluetooth support can be
connected to any Bluetooth-enabled GPS-mouse ...
I think, getting the pos
Hi, i have several incoming sip peers (mostly ciscos) , with 1.0 i
always registered them like this:
register => @prepago-in
[prepago-in]
type=friend
host=192.168.10.120
context = from-external
dtmfmode=rfc2833
insecure=very ; required for incoming FWD calls
Now with 1.2.4 it doesnt work any
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.
The WAN comes in from the Cisco IAD and into a LAN switch (DLink
DGS-1005D w/ 802.1p) where the two public IPs are switched to different
devices.
Paul Tinsley wrote:
Does anyone know of a way to specify what extension is dialed when 0
is pressed in the voicemail system. I have a situation where there is
more than one secretary and they want the 0 to redirect to the
appropriate secretary for the two groups of people.
So an example would
Brian Roy wrote:
Not totally true. A PRI is 23b 1d. A DS1 (US) is a 24 channel circuit.
Nitin - When you stop/start asterisk does it load all 24 channels? Any
errors? How about zap show channel 1 in the CLI?
Learn something new every day.
Doug
___
Anything under 1ms is so far below the threshold of perceivable sound
quality, echo, delay etc. that it's a mute point to discuss IMHO. Not even
in any cumulative effect it may have.
I can certainly see the advantages of SNMP for remote troubleshooting but
hard to justify for small offices with l
Am Friday 24 February 2006 16:48 schrieb Anton Krall:
> Any modification made to zapata as far as echo and gains?
>
> Should echocancel be on or off?
i have echocancel switched on, faxdetect is on, rx- and txgain is not used.
(commented out).
my var/log/messages says:
Found a Wildcard TDM: Wild
Anton Krall wrote:
Any modification made to zapata as far as echo and gains?
As a rule, don't let anything manipulate the audio at all... even echo
cancellation. That said, I have seen situations where gain had to be
increased.
Should echocancel be on or off?
Off, most definitely o
Anton Krall wrote:
Well, I have the same effect on my TDM as in the E1 using unicall... Faxes
get here as garbage :(
I really would like to see sometime some audio recordings made by
IAXmodem for people that had problems with TDMs and faxing with rxfax/txfax.
Not that I have some hope of
Hi list,
I got a question:
When I try to ChanSpy a SIP channel I only listen one
channel, for example,
I call from 302 extension and I have two active channels:
SIP/r1-voip-1b7b
(None)
Up Bridged Call(SIP/302-f1f1)
SIP/302-f1f1
[EMAIL PROTECTED
Date: Fri, 24 Feb 2006 14:56:54 +
From: Steve Kennedy <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] GPS-enabled cell phone/PDA
On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote:
> Its my understanding the cell phone coordinates are sent to the cell phone
> provider and
Andrew Kohlsmith wrote:
What is being discussed here is basically what I was planning on doing for an
automatic VOIP quality check. Using miliwatt and analyzing it for
pop/jitter/etc as well as sending other known waveforms and comparing what
was received to what was expected and coming up wi
Mike,
Were you able to get this working?
Even after with a entry in the dialplan.xml does not work for me.
Thanks,
Ken
On 6/20/05, Michael J. Tubby B.Sc (Hons) G8TIC <[EMAIL PROTECTED]> wrote:
Andrew,I presume you mean in the Cisco 7940/7960 SIP Phone Administrator's Guide?When you say "mapped",
On 2/24/06, Doug Lytle <[EMAIL PROTECTED]> wrote:
T1s require a D (Data) channel, unless connecting to a channel bank, Itshould be 23 voice 1 data. Also, I would strongly suggest moving to
1.2.4
Not totally true. A PRI is 23b 1d. A DS1 (US) is a 24 channel circuit.
Nitin - When you stop/st
I have talked with of a couple people(don't remember their names) who
had this developed on a contract basis for the 1.0 Asterisk code tree,
they did not want to release it to GPL because of how much it cost
them and the fact that their code supposedly won't run on 1.2, but it
is technically possib
Thank you Lee, Dave, Rich, Joel and of course also Kevin. Between your
various messages I finally understand what's happening and how it works,
and have actually converted everything to alaw, ulaw, slin and gsm and
am not actually using the mp3 side of things at all anymore. The
difference is v
In the UK this is common; several websites enable you to track a cell phone
online:
http://www.traceamobile.co.uk/
and another:
http://www.followus.co.uk/
Works the same way that Steve stated... The police here in Australia have
been using this since the late 90s.
Interesting article:
http:/
you need to set a TRANSFER_CONTEXT, either for the transferer or
transferee channel. I dont know why, but res_features give priority to
the transferee TRANSFER_CONTEXT, if not found, then use the transferer
TRANSFER_CONTEXT. That context is used to match the extension to dial.
So you can set this v
Any modification made to zapata as far as echo and gains?
Should echocancel be on or off?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thomas Artner
|Sent: Friday, February 24, 2006 8:25 AM
|To: Asterisk Users Mailing List - Non-Commercial Discu
Polycom does support Asterisk, Asterisk Business Edition.
-Original Message-
From: Michael Graves [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 23, 2006 6:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] What business IP phone to use
On
Well, I have the same effect on my TDM as in the E1 using unicall... Faxes
get here as garbage :(
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Rich Adamson
|Sent: Friday, February 24, 2006 7:28 AM
|To: Asterisk Users Mailing List - Non-Commercial
do you have a defaultcontext=something parameter in sip.conf [general] section?? If not, the default is... em "default"
RegardsOn 2/23/06, Chuck Bunn <[EMAIL PROTECTED]> wrote:
Hi,I am getting repeated messages in my logs with the following:*Feb 23 07:56:11 NOTICE[2470] pbx.c:
Thank you Armin. This is extremely useful.
Faris.
Armin Schindler wrote:
There are three possibilities to set the CallingPartyNumber (own number for
outgoing):
1) Set(CALLERID(number)=12345)
before Dial()
2) Dial(CAPI/contr1/12345:${EXTEN}/)
3) Dial(CAPI/contr1/${EXTEN}/d,...) and '
maybe you didn't want suggestions, but too bad :).
My favorite up until recently was the polycom 501 and I found it was
good quality and clear calls and priced well. but the production of te
phone is slowing down so I bought a few linksysspa941. and iVll
tell you I have a new favorite pho
Done..
They don't show much but they do show some problems with lost lines or
something
Thx Bartosz
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Bartosz Piec
|Sent: Friday, February 24, 2006 2:54 AM
|To: Asterisk Users Mailing List - Non-Comm
no chance, also with your scipt
ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
1 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
In the US, Sprint's CDMA network will do the fancy GPS+AFLT business,
but like someone else mentioned, it only sends the location data back
to Sprint's network. There is an API that you can use to access this
data for your handsets, but you have to pay some amount of money for
each location fix.
S
Hi Olle,
Will u be there for the speech of Jan Janak?
If yes, you will find a guy, 1m83, with a bear and a red suit, it's me.
You also can call me on my mobile to fix the voip beer (0032495283361).
We will try to have Jan and other guys
Olivier
___
--
I wrestled with this for a long time, as have many others and it just
doesn't work with spandsp and asterisk alone.
Use iaxmodem and hylafax in conjunction with asterisk... it works like a
champ. I have a single POTS line coming in so I get voice & fax with a
single number using fax detect.
ht
Chris Modesitt wrote:
I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is
APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server. When I
use Asterisk version 10.0.10 everything works perfectly, however when I
use 1.2.4 I lose the ability to receive calls from the PSTN
On Wed, 22 Feb 2006 18:02:27 -0800, mustardman29 wrote:
>Just the person I have been looking for. If you don't mind, would it be
>possible to get your opinion on feature for feature comparisons between the
>501 and 480i CT(not including cordless phone).
>
>Things like programmable buttons, display
On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote:
> Its my understanding the cell phone coordinates are sent to the cell phone
> provider and their equipment reads (and holds) that data. Its not part
> of any data available to you in any form unless you talk to the cell
> provider and
Chris Modesitt wrote:
I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is
APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server. When I
use Asterisk version 10.0.10 everything works perfectly, however when I
use 1.2.4 I lose the ability to receive calls from the PSTN
On 2/23/06, Geoff Manning <[EMAIL PROTECTED]> wrote:
How would you categorize a Yellow Alarm sensed by the Asterisk side in a Legacy PBX integration?We have a Mitel SX200 connected to an Asterisk(1.2.4) with a TE110P.Twice today (first time in over a month) we received a Yellow Alarm on the TE110P.
Nitin Joshi wrote:
Hi All,
I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its
connected directly to the PSTN. But I am unable to make outbound calls
on the zap channels. The light on the card is green. Asterisk CLI
shows all 24 channels when I give the command 'zap show c
I currently use asterisk version 1.0.10 with AMP 1.0.010,
our setup is APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk
server. When I use Asterisk version 10.0.10 everything works
perfectly, however when I use 1.2.4 I lose the ability to receive calls from the
PSTN. All I get is th
udev drove me absolutely bat-shit in this regard; udev is a horror in many
respects. Here's how I solved the problem, reliably:
I run this script at boot-time:
#!/bin/bash
mkdir -p /dev/zap
rm -f /dev/zap/ctl
rm -f /dev/zap/channel
rm -f /dev/zap/pseudo
Hi!
I am using tdm400 cards for receiving faxes. It worked quite out of the box. I
installed spandsp for the rxfax application only.
I use it as phone/fax switch:
All incoming calls are answered automatically to listen whether its a fax or
not. If it is a fax, the call is forwarded to the buil-
On Thursday 23 February 2006 13:57, Bob Goddard wrote:
> It's stupid. Don't ever connect 2 different building with copper.
> Just wait until you get some kind of lightening hit or electrical
> fault, but make sure you are no where near it. Use fibre.
That's a great rule of thumb, but the reality i
On Sat, 2006-02-25 at 00:21 +1100, David Ankers wrote:
> Aha, micro seconds in networking terms is normally written usecs or us
> (actually it's the greek letter mu as in ulaw) rather than ms which are
> milliseconds seconds - what had me puzzled was that it was stated that this
> could harm the vo
Hi all, this is another post about this problem.
I installed from scratch a new Suse Linux 10.0, with latest stable
asterisk.
Moreover I add the lines to /etc/udev/rules.d/50-udev.rules, in order to
let the driver create the /dev/zap...
When I plug into usb port my TigerJet adapter, I see on
Andrew Kohlsmith wrote:
>On Friday 24 February 2006 07:56, Paul wrote:
>
>
>>Maybe the first approach should be to setup a test extension for
>>recording the tone. The idea is to get best resolution possible in real
>>time. Then process it as much as needed to get the info you want. Such
>>an ap
I have seen some very expensive switches fail. Nice thing about lower
cost devices is that you can afford to have spares. If you stick to a
standard way of labeling and connecting wires you can use good open
source monitoring software to detect switch failure. If you allow people
to randomly connec
Hi There,
this is very much dependent from your provider, your PDA/cell phone and the
network. For GSM networks in Europe e.g. the providers have different types
of information available through the CB channels of their base stations.
This data can always be read and stored in your SMARTPHONE/P
On Thursday 23 Feb 2006 20:34, Colin Anderson wrote:
> >It's stupid. Don't ever connect 2 different building with copper.
> >Just wait until you get some kind of lightening hit or electrical
> >fault, but make sure you are no where near it. Use fibre.
>
> Thanks for the reply. Unfortunately, the co
Hi All,
I have installed a Digium TE110P card on an
Asterisk 1.2.1 system. Its connected directly to the PSTN. But I am unable to
make outbound calls on the zap channels. The light on the card is green.
Asterisk CLI shows all 24 channels when I give the command 'zap show channels'.
I also
> Ive been testing how to receive faxes using TDM400P cards and so far, after
> playing with gains, echocancell and echotraining on zapata.conf.. Ive ha dno
> luck, faxes come in as garbage or broken or with blank lines.
>
> Anybody has successfully done this? Any tips.. Also I have some ideas:
>
On Friday 24 February 2006 07:56, Paul wrote:
> Maybe the first approach should be to setup a test extension for
> recording the tone. The idea is to get best resolution possible in real
> time. Then process it as much as needed to get the info you want. Such
> an approach would give you more flexi
On 2/24/06, Marco Maiolini <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I configured Buddy Watch function on my Polycom IP 601. It works well, until
> I make a reload of Asterisk. After reload, if I give the "show hints" command
> in Asterisk's CLI, it says that there are no watcher for the extensions th
1 - 100 of 141 matches
Mail list logo
