Doug Lytle wrote:
> [EMAIL PROTECTED] wrote:
>
>> On Mon, 3 Oct 2005, Corey S. McFadden wrote:
>>
>>
>>
>>> Am I just using the Set() command wrong? It seems pretty
>>> counter-intuitive not to enclose multi-word strings in quotes but if
>>> that's the problem let me know.
>>>
>>
>>
>> Yeah,
Hi ,
Does anyone encounter this problem ? We have installed Asterisk at Site
A and have 128k Frame Relay over to Site B.
We are using Zyxel 2 port FXS at Site B and Linksys PAP2-NA at Site A.
We are using Ulaw at Site A and G729 at Site B.
When the calls are originated from Site A to Site B,
IMHO, Snom is moving in the right direction as they have migrated from
the Snom 190, to the 200/220, to the 320/360. Their phones have always
worked well, but from an aesthetic standpoint their earlier models were
not the best. They continue to improve the overall design, look and
feel of the
Hi!
I am facing some problems with my asterisk-1.0.3.I am using DIAX phones as clients ,but sometimes they donot register with the asterisk server.Also if I don`t restart my asterisk frequently the registration of DIAX phones expires.
Anyone who can help me please reply
Regards,
Amna
___
Hi All,
I've been having a weird issue with one of my servers and it's Asterisk
installation.
The server is running Slackware, and Kernel 2.6.12. I'm running the
latest CVS-HEAD
edition of *. I also have 4 other asterisk servers with the same
software configuration,
but different hardware an
I use them and like them a lot. Like all IP phones they have...charming
quirks. But they are IMO no more or no less hassle than other IP phones, and
the firmware at least is updated regularly and snom goes through great pains
it seems to ensure that the phone is properly supported. They are also
On 10/04/05 05:54 Matt Roth said the following:
This post documents moving the calls from the RAM disk to a hard disk on
a remote machine via NFS. The setup is not resource intensive on the
Asterisk server and should not impact call quality. As always, I welcome
suggestions for improvement an
This is my debug with the same issue
The agi terminates during the "sub tell_time()"
and exits without calling "sub setinuse()" or completing the reset of the
script.
AGI Tx >> agi_request: astcc.agi
AGI Tx >> agi_channel: Zap/49-1
AGI Tx >> agi_language: en
AGI Tx >> agi_type: Zap
AGI Tx >> ag
On Mon, 3 Oct 2005, Aryanto Rachmad wrote:
> I sent an email to Digium support and got only a reply like this:
>
> "Although the card is being shown as an 'Unknown Device', it should still
> work properly."
>
> To be honest, I am not happy with that answer.
Well, does it work properly? I
Hi, everyone:
I'm in the processing of deciding what IP phones we should use with our
Asterisk server, and I wanted to get feedback from the user community on
the quality, reliability and ease of operation of Snom phones.
What do you have to say about these phones? Are there other phones you'd
su
Can you please post the output with debug agi on ?
Darren Wiebe
[EMAIL PROTECTED]
Scott Wolfe wrote:
I download and installed ASTCC over the weekend and I am having an
issue where the INUSE flag will not get set back to 0 if the user
drops a call while the balance is being played. All other t
I have no idea about the 3XXX series of
phones. The 2XXX used to have SIP firmware but I could never get my hands
on it. I used to see the SIP 2XXX phones selling on Ebay from time to
time. I imagine that even if you can locate the SIP firmware for the old
phones, you would have to uploa
Hello!
Are you having RTP timing problems? Silence suppression having to be
disabled? Well, here's a patch that *MIGHT* solve all of your problems,
available at: http://bugs.digium.com/view.php?id=5374
If you are willing to give it a try, let me know if you experience any
problems, but posting fe
Juraj Bednar wrote:
Hello,
has anyone seen or created a Debian Sarge package for 1.2beta1?
I saw some for Sid, but I prefer not upgrading glibc right now, as
this is a production server (Asterisk on it will be for testing).
Thanks,
Juraj.
You take the source for the sid packages an
Which version of asterisk and zaptel are you using?
Will they work with 1.0.9 ?
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
FWD: 512237
Your telco must provide remote disconnect supervision on that trunk in order for disconnects to be recognized by your card.
On 10/3/05, Leigh Fereday <[EMAIL PROTECTED]> wrote:
I realise I am not alone in this, but I don't seem to be finding manysolutions.I am running a CentOS box and *
1.09 wit
I can send/receive just fine on an eicon bri to a zaptel analog
interface.
I would say, if you wish to use faxing on a regular basis to a remote
proxy though, you're possibly better off with a landline.
Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] O
I realise I am not alone in this, but I don't seem to be finding many
solutions.
I am running a CentOS box and * 1.09 with a TDM11B (1xFXS/1xFXO).
Everything works great accept hang-up detection of incoming PSTN calls.
* answers the call, but if the incoming caller hangs up, * does not release
t
In my case it's not MWI (I didn't have it enabled - 0 splash), also had most
Supplementary services turned off. The rings are actually generated from the
FXO side of the spa, presenting to Asterisk as an incoming call.
Mine is likely to be a power issue, I experience frequent quick power "even
That's what I had thought originally too, but apparently there is/was an issue with the 3.1.5gw and below firmware where it was possible for AC noise coming from the power supply to be falsely identified as a ring. Sipura has apparently just released
3.1.7 to deal with this.
I've never had t
Hi,
My wildcard TDM 400P's FXS port(connexts to a analog phone) has ticking noise recently.
Dialing completlyfails because of this.The only thing still working is generating rings when
being dialed.
I have followed http://www.voip-info.org/wiki-Asterisk+Hardware .
and make sure wctdm is not shar
I am jumping into this thread late, so forgive me if I missed relevant
info, but the single ring you hear is more than likely the "MWI Splash"
which can be disabled by simply setting the duration of the splash to 0
seconds.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-
I thought maybe someone was using 0.63 with code they
developed themselves. Where do you find 0.62.x?
--Richard Cook[EMAIL PROTECTED]T: 705-223-2000
x2010
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy
KuoSent: Monday, October 03, 2005 7:16 PMTo:
[EMAIL PROTECTED];
Hi,
People on the list just told me that we can only use 0.62.x
AK
On 10/3/05, Richard Cook <[EMAIL PROTECTED]> wrote:
Hello,
Is anyone using FreeTDS version 0.63 with *?
--Richard Cook[EMAIL PROTECTED]T: 705-223-2000 x2010
___--Bandwidth and Col
Hello,
has anyone seen or created a Debian Sarge package for 1.2beta1?
I saw some for Sid, but I prefer not upgrading glibc right now, as
this is a production server (Asterisk on it will be for testing).
Thanks,
Juraj.
___
--Bandwidth and C
Hi Paul,
Paul Dugas wrote:
I'm not using the SPA3k as an extension at the moment; just as an FXO
interface. The SPA is initiating a SIP call to the Asterisk server then
DELETE'ing it 2 secs later. Asterisk is ringing other IAX/SIP extensions
in response. The FXS interface of the SPA3k *is* se
I would think I could do this but for some reason I am stymied.
I have a PRI from RCN connected to a cisco 3640 (in my day "cisco" is
all lower case :-)). My config looks something like this on the cisco...
-
voice-card 3
dsp services ds
I am using Kphone which works great for my purposes! You can look at
twinklephone as well at http://www.twinklephone.com/
rajesh
- Original Message -
From: "Wayne Gemmell" <[EMAIL PROTECTED]>
To:
Sent: Monday, October 03, 2005 3:11 PM
Subject: [Asterisk-Users] sip phones on x86_64
> Hi
On Mon, 2005-10-03 at 17:27 -0400, Paul wrote:
> As for X on the same box as *, it only seems to affect calls when I do
> something that uses enough cpu. I can be logged in with a gnome or kde
> desktop without causing problems. It's a P4 2.4 with 1 gb DDR 333.
For smaller volumes of calls (10-2
Rich Adamson wrote:
My office has been running Asterisk 1.0.8 and a TDM04B for a few months
now without too much trouble. After a while we discovered that after a
certain period (about a month), asterisk stopped acknowledging inbound
calls. Since I was out of the office the first time it hap
Bruce Ferrell wrote:
a file, /etc/asterisk/voicemail.conf for the voicemail system defaults
AND the /etc/asterisk/extconfig.conf entry for the mailboxes.
Or am I totally stupid and just making this difficult?
No, you are not. That is exactly how it works; 'dynamic realtime' mode
reads indiv
It took me a whole lot of reading to catch what is likely not all that
subtle to many, but it was subtle to me.
Static: Is the global config stuff from voicemail.conf
mailboxes aren't done in the voicemail.conf file.
The perl script, ast2sql.pl will parse a given config and
Aryanto Rachmad wrote:
Is the status of "Unknown device" a normal status?
Yes, it's normal and expected.
"Although the card is being shown as an 'Unknown Device', it should still work
properly."
To be honest, I am not happy with that answer.
Why? It works, and the PCI vendor/device ID o
All the ‘unknown device’ means
is that your ‘lspci’ doesn’t know what the card is. That’s
all. Nothing more.
--Rob
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad
Sent: Tuesday, 4 October 2005 7:43
AM
To: asterisk-users@lists.digium.com
Subj
Hi all,
I have an asterisk-server (cvs-head from august) connected to a
carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems
with DDI (standard 'official pstn' number plus extra digits for
'internal' use)
Basically, when the entire number (including the extra digits) is
dialled
Hello,
Is anyone using FreeTDS version 0.63 with *?
--Richard Cook[EMAIL PROTECTED]T: 705-223-2000
x2010
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.co
List members,
My previous post "SUCCESS - 512 Simultaneous Calls with Digital Recording" documents using a RAM disk to eliminate the I/O
bottleneck associated with digitally recording calls via the Monitor application. By recording directly to a RAM disk I was able to maintain good call quality
Hello everybody, I have been googling for hours and also
searched on http://www.voip-info.org/wiki-Asterisk,
but I still can not find any information for the problem I have. So I
hope one of you could help me out. I have actually very little
experience in Asterisk and also Linux. But by fol
Roy Sigurd Karlsbakk ha scritto:
hi
is it possible to use asterisk as an sms central to send SMSes directly
to clients on PSTN instead of just communicating with a central? the
telco to which we're currently connected doesn't have a central
Yes, as far as you can spoof the Caller ID ;
Steve,
I'm glad to know what the problem is. We're back to normal now. FWIW,
this was working up until about a week and a half ago and didn't affect
our non-Cisco phones... I'm not sure what component (Asterisk, chan_sip,
79xx firmware, etc.) became less tolerant of the error between th
Christopher Dobbs wrote:
Matt wrote:
Extremely good point... I myself am a Linux person, but manage several
Windows machines (several meaning 25 or so). There is definately a
time and place for Windows.. I'm just not sure a real-time-VoIP server
is the time or place.Being semi-half serio
On 10/3/05, ashley wright <[EMAIL PROTECTED]> wrote:
>
>
> Hi,
>
> Can any one help I'm trying to install asterisk on suse 9.3 pro from cvs
> release v1_0 version 1.0.9 and when I try to make from the asterisk
> directory I get the following error.
>
>
>
> Is there anybody that could give me a poi
[EMAIL PROTECTED] wrote:
On Mon, 3 Oct 2005, Corey S. McFadden wrote:
Am I just using the Set() command wrong? It seems pretty
counter-intuitive not to enclose multi-word strings in quotes but if
that's the problem let me know.
Yeah, that's the problem.
Steve
In my case, I'm
A stale nonce is more of a warning than an error. In SIP your
authorization credentials are encoded in the SIP headers. To prevent
people from capturing that data and using it later to make calls on your
account a nonce is used.
A nonce is a disposable number that is added to the string a hash
a
Matt wrote:
Extremely good point... I myself am a Linux person, but manage several
Windows machines (several meaning 25 or so). There is definately a
time and place for Windows.. I'm just not sure a real-time-VoIP server
is the time or place.Being semi-half serious about the GUI there
also
Once upon a time Monday 03 October 2005 3:11 pm, Wayne Gemmell wrote:
> Hi all
>
> Can anyone recommend a good soft phone that can compile on x86_64 (linux)
> platform?
kphone compiles and is available in Fedora extras and im sure is available
for other distros. If you want to get adventurous yo
On 10/3/05, Olle E. Johansson <[EMAIL PROTECTED]> wrote:
> Does anyone know what "stale nonce" is?I've answered this question many times, so you should be able to find
the answer...A stale nonce is when a device tries to re-authenticate with a noncethat is no longer valid. We are telling them that
Just to clarify. These products are not produced by this company, its
Taiwanese brands. The SIP-CPE Gateway is a rebranded VodTel MOSA 3700
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
Michaelson
Sent: den 3 oktober 2005 18:12
To: asterisk-
On Mon, 3 Oct 2005, Corey S. McFadden wrote:
> Am I just using the Set() command wrong? It seems pretty
> counter-intuitive not to enclose multi-word strings in quotes but if
> that's the problem let me know.
Yeah, that's the problem.
Steve
___
-
On Mon, 2005-10-03 at 13:02 -0700, [EMAIL PROTECTED] wrote:
> the app_cepstral.c file had a problem that it was trying use
>
> #include "../asterisk.h"
>
> I had to force it to where asterisk.h was located... in my case it was in
> /usr/src/asterisk/include
> so i changed the #include to say
>
>
Hi all
Can anyone recommend a good soft phone that can compile on x86_64 (linux)
platform?
--
Regards
Wayne Gemmell
Tel & Fax: (011) 894-4081
Cell : 072 836 4325
Email : [EMAIL PROTECTED]
___
--Bandwidth and Colocation sponsored by Easynews.com -
Olle,
Thanks for looking into it. In doing some ngrep work I figured out where
my problem is.
Acutal error from the 79xx inside the SIP header is:
Warning: 399 Bad Request - 'Malformed/Missing FROM: field'
>From looks like this:
From: ""Sales Queue"" Doug Lytle wrote:
> > Olle E. Johansso
the app_cepstral.c file had a problem that it was trying use
#include "../asterisk.h"
I had to force it to where asterisk.h was located... in my case it was in
/usr/src/asterisk/include
so i changed the #include to say
#include "/usr/src/asterisk/include/asterisk.h" and then it would compile
thr
Hi,
I have a SIP.CONF with a user section like this:-
[1234]
accountcode=HABITAZ
type=friend
callerid="HABITAZ/1234"
context=milkshake
userName=1234
secret=1234
host=dynamic
dtmfmode=rfc2833
qualify=yes
callgroup=1
pickupgroup=1
canreinvite=no
When I login from a X-Lite phone, with "Username"
> My office has been running Asterisk 1.0.8 and a TDM04B for a few months
> now without too much trouble. After a while we discovered that after a
> certain period (about a month), asterisk stopped acknowledging inbound
> calls. Since I was out of the office the first time it happened,
> ano
Jenna Cole wrote:
receive the fax via SIP and send it to my faxmachine.
I also want to send a fax from my faxmachine through
the digium card, so asterisk should send the fax via
SIP to the gateway, which also has a faxmachine
connected.
is this possible?
Short answer, no. Long answer can b
Hi.
I'm trying to configurate Quintum AX to work with Asterisk SIP egister/Proxy
server and my problem is that only the first user account get logged in and
only that user is able to make call correctly. It seems to be a problem with
authorization - I have noticed no "Proxy-Authorization" informat
is it possible to achive the following scenario?
faxmachine--tdm40bFXS--SIPnetwork--Gateway--faxmachine
i have found a lot of documents about asterisk
receiving a fax and saving it to a file. But i want to
receive the fax via SIP and send it to my faxmachine.
I also want to send a fax from my fax
Patrick Friedel wrote:
I couldn't find a changelog for 1.0.9 to see if it's worth the
off-hours maintenance window, and we're too dependant on the phones to
try 1.2. Should I try the next step up in the probably unnecessary
preventative maintenance and unload/reload the wctdm module during th
Wojciech Tryc wrote:
I am not following...
Why would you need to integrate Cepstral directly into Asterisk? Just
to be able to call it as Asterisk app from your dialplan? I am running
Cepstral and calling it through the System call.
You could try the howto located here:
http://www.oldskool
I've been getting the same problem with the verbose issue. I just
commented out the line, and it seemed to compile OK.
--
Tom
On 10/3/05, Doug Lytle <[EMAIL PROTECTED]> wrote:
> Dave Cotton wrote:
>
> >On Mon, 2005-10-03 at 14:10 -0400, Doug Lytle wrote:
> >
> >
> >Look at rxfax.c around line 88
Dave Cotton wrote:
On Mon, 2005-10-03 at 14:10 -0400, Doug Lytle wrote:
Look at rxfax.c around line 88 there's an #if statement remove the
references to callerid.
This error has been around for a while.
That took care of the callerid compile error, but not the verbose error:
error:
My office has been running Asterisk 1.0.8 and a TDM04B for a few months
now without too much trouble. After a while we discovered that after a
certain period (about a month), asterisk stopped acknowledging inbound
calls. Since I was out of the office the first time it happened,
another admin
On Mon, 2005-10-03 at 14:10 -0400, Doug Lytle wrote:
> Has anybody been successful with compiling the pre3 version of SpanDSP
> on the current Asterisk CVS? I'm getting:
>
> app_rxfax.c: In function `phase_e_handler':
> app_rxfax.c:77: warning: implicit declaration of function
> `fax_get_transf
I've got a problem with audio output from the Asterisk console. I'd really appreciate any help.
I'm simply trying to dial out to a phone on PSTN. My extensions.conf entry is as follows:
exten => _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})exten => _1NXXNXX,2,HangupAfter starting a
Where is it getting the extra 8 from? It seems like
you are passing an invalid number to the trunk.
Spawn extension (crystal-sip, 8800759, 1)On 10/3/05, Crystal Stream, Incorporated
<[EMAIL PROTECTED]> wrote:
crystalstream*CLI>-- Executing Macro("SIP/3044-5300","outvoip-2|1800759") in
Has anybody been successful with compiling the pre3 version of SpanDSP
on the current Asterisk CVS? I'm getting:
app_rxfax.c: In function `phase_e_handler':
app_rxfax.c:77: warning: implicit declaration of function
`fax_get_transfer_statistics'
app_rxfax.c:78: warning: implicit declaration of
> Subject: Re: [Asterisk-Users] SPA-841 "Decode Latency"?
Luki <[EMAIL PROTECTED]> wrote:
> > Does anyone have any familiarity with "decode latency," specifically
> > with Sipura devices? Why would it take 200+ms to decode a 20ms RTP
> > packet? G.711u has existed for over 30 years, how hard coul
Hi all,
Does anyone know if it is possible to disable the pound key on the 7960 to
not place calls so that other services can be used in Asterisk, such as call
forwarding. Any info is apreciated, many thanks!
___
--Bandwidth and Colocation sponsored
Hi,
I'm trying to set up a call-back system using auto-dialout files. I
want the call to be terminated when a specific timeout (defined in the
.call file) is detected. Both parties should then be hangup.
The problem is that the timeout is never detected... How to solve this?
Thank you,
Pierr
When qualify is set to yes in sip.conf for a "friend" and the OPTIONS packet
gets returned with an ICMP port unreachable message, what is the behavior of
Asterisk?
It looks to me like Asterisk tries sending the OPTION request again right away
(well within a second or two).
Some of our devices are
Hello Everyone,
Please accept my appologies - I've been reading through the handbook
and the online documentation / mailing list archives and can't quite
get my own answer to these inquiries... The biggest mystery is
how the existing handsets are connected to a new machine running
Asterisk.
Back
crystalstream*CLI>
-- Executing Macro("SIP/3044-5300",
"outvoip-2|1800759") in new stack
-- Executing SetCIDName("SIP/3044-5300", ""CRYSTAL
STREAM NET"|a") in new st ack
-- Executing SetCIDNum("SIP/3044-5300",
"866xxx|a") in new stack
-- Executing Authenticate("SIP/3044-5300
Hi!
On Mon, Oct 03, 2005 at 05:41:38PM +0200, Mark Elkins wrote:
> I'm also using SNOM320/360 phones. Ideally - set up one button to toggle
> the Agent Status (in/out == On/Off) ???
> Kinda make sense if app_devstate (or similar) made it into mainstrean
> Asterisk - so line indication lamps could
On Monday 03 October 2005 12:17, Rich Adamson wrote:
> Think you might have jumped to a conclusion that might not be valid.
> "If" the telco can handle a PRI and will accept callerid from you,
> and each unit has a valid telephone number, then the telco can populate
> the callerid database with nam
I have configured TDMoE sucessfully and I am able to make a Zap connection
from one box to the other.
The question I have is..
I'm getting repeated errors every second on both systems..
Oct 3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected
alarm on channel 1: No Alarm
Oct 3
Hello,
Would like to use IAX /IAX2 to transport 30 channels inter Asterisk.
I don't have any experience with that, so can someone help ??
How much bw do I need and what latency for SIP G711 to IAX and vice-versa , ...
etc ?
Thanks in advance for any info,
Geo
Hi,
Can any one help I’m trying to install asterisk on
suse 9.3 pro from cvs release v1_0 version 1.0.9 and when I try to make
from the asterisk directory I get the following error.
Is there anybody that could give me a pointer as to what the
issue may be?
DDIR=\"/usr/lib/as
On Mon, October 3, 2005 12:44 pm, Rich Adamson wrote:
> Not likely anyone is going to comment on this without looking at your
> traces, s/w versions, config detail, etc. There are just too many ways
> to configure an spa and guessing at what you've done is impossible.
Good point. The trace of wh
Yes. It's gone.
On 10/3/05, Dinesh Nair <[EMAIL PROTECTED]> wrote:
On 09/30/05 03:12 Verlin Henderson said the following:> Xeon server (most likely a Dell PowerEdge 2800, 2850, or similar) with a
> large amount of RAM and RAID-1 SCSI setup. We would add three TE411P or> TE410P cards and implement
On Monday 03 Oct 2005 08:51, Olle E. Johansson wrote:
> Paul Conn wrote:
> > I’m receiving the following error over and over, adnauseam:
> >
> >
> >
> > Oct 1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce
> > received from ‘CNAME-CID ’
> >
> >
> >
> > Does anyone know what “stale
Has anyone used the GSM-SIP gateway product produced by a company at
sipcpe.com? Any comments?
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/li
> I got back from two weeks away and appear to have lost audio on my
> tdm411 fxo. Everything was working properly when I left. I checked the
> logs, config files and can't see any problems, the zap channels and
> modules are all loaded properly, asterisk starts without probs and
> everything
On Fri, 2005-09-30 at 09:38 +0100, Derek Conniffe wrote:
>
> Is anyone out there running two AVM Fritz ISDN cards?
Yes
> Are you using a 2.6.XX kernel?
No
> How are you doing it?
Easily :)
Really just carefully follow the instructions in the "hack" you've
already mentioned.
It works, b
That's interesting for sure. I'd bet if you had some way to monitor what was going on with the FXO (voltage) side of things you'd probably find that something is happening that is causing the spa3k to believe that it's receiving ring voltage on the line. You can tune these settings in "Internation
On 09/30/05 03:12 Verlin Henderson said the following:
Xeon server (most likely a Dell PowerEdge 2800, 2850, or similar) with a
large amount of RAM and RAID-1 SCSI setup. We would add three TE411P or
TE410P cards and implement something similar to Matt Roth's setup, but on a
smaller scale.
ha
how many digits is your callerid passing to the trunk? I am seeing 11
8663xx3 is that correct? I had an issue last week with
passing to many digits to my provider and the call would hang up
immediately.
You could also turn debugging on for this so we can get a better log.
iax2 debug peer nu
> This is a wierd one. Can't figure it out. I have an SPA-3000 at the
> house handling my incoming line. It's setup to direct the incoming call
> to asterisk. Works great 99% of the time.
>
> A few times a day, I'm getting calls that ring once internally and are
> then hungup. I managed to g
On Mon, 3 Oct 2005, Giordano Grandis wrote:
> Which models of Diva could work with CAPI and asterisk?
- 'Diva Server' PCI cards with 'divas' driver from melware.net or Eicon source
RPM
- passive Diva cards supported by mISDN
Armin
> Thanks
>
>
>
> Giordano
>
> __
I'm also using SNOM320/360 phones. Ideally - set up one button to toggle
the Agent Status (in/out == On/Off) ???
Kinda make sense if app_devstate (or similar) made it into mainstrean
Asterisk - so line indication lamps could be used at will.
The SNOM320 is so ideal for Call Centres (the Headset co
On Sat, 2005-10-01 at 07:39 -0600, Rich Adamson wrote:
> > >
>
> I believe you meant to say "make update". "upgrade" is not a defined
> parameter.
No, I meant to say exactly what I said.
Read the F Makefile :), line 677
upgrade: all bininstall
--
Dave Cotton <[EMAIL PROTECTED]>
Hi,
I got back from two weeks away and appear to have lost audio on my
tdm411 fxo. Everything was working properly when I left. I checked the
logs, config files and can't see any problems, the zap channels and
modules are all loaded properly, asterisk starts without probs and
everything looks
I am not following...
Why would you need to integrate Cepstral directly into Asterisk? Just to be
able to call it as Asterisk app from your dialplan? I am running Cepstral
and calling it through the System call.
Thanks,
Wojtek
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asteris
I would say the problem here could fall in this category.
Jason Walker a écrit :
> I have run into a similar situation. One of our older faxes at the office
> seems to not work with spandsp module. The newer faxes work just fine.
>
> When I watch the logs, there appears to be communication from
Which models of Diva
could work with CAPI and asterisk?
Thanks
Giordano
Da:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di [EMAIL PROTECTED]
Inviato: sabato 1 ottobre 2005
23.46
A: asterisk-users@lists.digium.com
Oggetto: RE: [Asterisk-Users] Diva
Nope.
Then did you do a make clean / make / make install?
Then do "show applications" at the CLI prompt after you have restarted
asterisk.
"service asterisk stop"
"service asterisk start"
...
> I downloaded Cepstral to my Asterisk Box. I did the install and let it
> install to /opt/swift.
>
> I brou
Think you might have jumped to a conclusion that might not be valid.
"If" the telco can handle a PRI and will accept callerid from you,
and each unit has a valid telephone number, then the telco can populate
the callerid database with names. Those are the only two items the
telco can provide in rea
After "-- IAX2/NuFone/3 is making progress passing it
to SIP/3044-bcd0" I'm getting a "Busy" tone and it's
not even connecting the call.
-- Executing Macro("SIP/3044-bcd0",
"outvoip-2|1800759") in new stack
-- Executing SetCIDName("SIP/3044-bcd0", ""X X X"|a")
in new stack
-- Executing Se
Hi,
I have this setup
DSL ROUTER>LINUX->ASTERISK
LINUX acts as a router with this config:
ppp0 - internet interface (public)
eth1 - private interface: 192.168.1.254
asterisk interface 192.168.1.251
settings on LINUX:
iptables -t nat -A POSTROUTING -o ppp0 -j MASQUERADE
echo 1 > /proc/s
This is my zaptel and zapata. In my logger.conf this is what is enabled :
full => notice,warning,error,debug,verbose.
How can you turn on the log in chan_zap.c and where you can access
it. You can see i'm a newbee :-)
Thanks for your help
Pierre
;
; Zapata telephony interface
;
; Configu
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