bmw suzuki wrote:
Hello all ... mY first ever post in here.
I am bit or (full) confused on what this program does.is
http://does.is it useful if i have a alcatel pabx system.And i can
bill my guests for their call charges etc..
can i use it on calling another computer on the network via
Thanks Kristian,
but i just answered to call, how can i use the Read application?
Thanks
Giordano Grandis
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Kristian Kielhofner
Inviato: lunedì 6 marzo 2006 18.15
A: Asterisk Users Mailing List -
How to upgrade h323 from Asterisk add-ons (from version 1.2.1 to 1.2.2)?
In INSTALL they don't say anything about upgrade...
Thank you for your time!
--
Tomislav Parcina
tparcina#lama.hr
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Hi group!
Does anybody knows about any news server that works the same way that Gmane
www.gmane.com/ does it? I was satisfied with Gmane for few months, but now it
seams that it doesn't work any more (no new posts in past few days). Now I'm
looking for alternative.
--
Tomislav Parcina
Hello,
Load wctdm with debug=1, i.e, add this line to /etc/modprobe.conf:
options wctdm debug=1
Then watch /var/log/messages (tail -f /var/log/messages will do it),
and check when you are getting the first polarity reversal, you should
get it before the first RING. If it happens that you get
On Tue, 7 Mar 2006, Zach A wrote:
I have SPA3000 receiving PSTN calls and also have a SIP line on the same
Asterisk server with 5 extensions. Now there is a fax too which comes
through the PSTN line. Fax calls have short rings. Can Asterisk somehow
detect those short rings and send the call to
Hans Witvliet wrote:
Can anyone inform me if voip can be used on a IPv6 network?
Does any hard phones/soft phones/Asterisk support it?
Google told me that there was/is a bounty on it,
but that expired august last year.
Furthermore, there used to be a patch (Bernhard Schmidt), but that one
is
Hei.
I have a question about how to get the periodic-announce to work within
my queues. I got the following test:
extensions.conf --
exten = s,2,Queue(test|rtT|||200)
Queues.conf --
[testqueues]
strategy = ringall
context = testcontext
timeout = 250
periodic-announce-frequency=60
On Mon, 6 Mar 2006 18:53:30 +0100 (CET)
Christian Benke [EMAIL PROTECTED] wrote:
Hello!
i'm trying to set up transfer without using the respective
asterisk-function but with the built-in phone functions. my goal is to
have the first callleg billed to the caller and the second callleg to the
Try this ...extensions.conf --; Queue with Music on holdexten = s,2,Queue(test|mtT|||200)
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On Mon, 2006-03-06 at 10:54 -0500, Sina Bahram wrote:
Here is the compilation process of zaptel
I did edit the makefile and uncommented the #ztdummy, although, after I did
that, I get the make error of ztdummy being defined more than once.
[snip]
You don't need to uncomment ztdummy in the
Hi i have configured extensions to record voice conversions ON DEMAND on my [EMAIL PROTECTED] so how will we start the recording when the call is in progress.thanksGiridhar Bandi
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Hi group,
I am currently looking for a prepaid application that
can do the following:
Use the Caller ID/Card Number for authentication
Can map a rate plan on a specific Caller ID/Card
Number
Supports prepaid functionality in terms of trunk
connection.
These functionalities seems
However, as I pointed out in my email, that doesn't make any difference.
If I leave it commented ... I get the exact same thing: just minus the make
file error
Same behavior, same error messages with the /etc scripts, the modprobe's and
with everything else.
Take care,
Sina
-Original
Hello
I attempt installing H323 at my [EMAIL PROTECTED] for this use
asteriskathome-h323-1.0.zip but have next problem
chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory
chan_oh323.c: In function `oh323_show_channels':
Please help for resolve this problem
Viktor Tatianin
On 03/07/06 01:14 Douglas Garstang said the following:
Hi Doug. I worked it out. I had commented out chan_zap.so in
modules.conf as I didn't think I needed it. It was doing weird stuff,
including not playing the participants joining. Weird.
MeetMe needs a timing device to work correctly. you
On 03/07/06 00:44 Douglas Garstang said the following:
Anyone seen this? If not I guess I'll have to post it as a bug.
Extensions.conf has this: exten = 123,1,Meetme(|dMic|)
I dial 123, and enter my conference number. Asterisk asks me to enter my
name. At this point I hang up. If I type at
Hi everyone,
I just spend the last two hours trying to get two asterisk boxes to
transfer calls between eachother using SIP. I dont know why but I *could
not* get the calls to authenticate! I think I got everything setup.
There was Server A and Server B. I was trying to place a call
You can activate the on demand recording on the [EMAIL PROTECTED] by
adding w and W in the asterisk dial command option. It
is located on the general settings under setup. Either
the caller or the called party can initiate the
recording by pressing *1.
Leonimar,
--- Giridhar Bandi [EMAIL
Hi Viktor,
What is the version of the asterisk you are using? You
should use the right version of Openh323 and pwlib to
be able to compile chan_oh323 successfully.
Currently using asterisk 1.2.4 used
openh323-Mimas_patch2-src-tar.gz and
pwlib-Mimas_patch2-src-tar.gz for compiling
chan_oh323.
On 3/6/06, Olle E Johansson [EMAIL PROTECTED] wrote:
7 mar 2006 kl. 02.45 skrev Scott Call: Since e164.org added DNC and ADDRESS records my enum configuration has failed. Using both the old EnumLookup app and the new ENUMLOOKUP function,
the lookups have consistantly failed since e164.org added
ya i found it it *1 to start recording from the caller end thanksGiridhar Bandi On 3/7/06, Giridhar Bandi
[EMAIL PROTECTED] wrote:Hi i have configured extensions to record voice conversions ON DEMAND on my
[EMAIL PROTECTED] so how will we start the recording when the call is in
Hi everyone,
I want to call from one Asterisk to another Asterisk via SIP, but i dn't know how. I have found out something in these links:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels
but I don't
2006/3/7, María Chóliz [EMAIL PROTECTED]:
I want to call from one Asterisk to another Asterisk via SIP, but i dn't
know how.
If you are connecting one asterisk to another, I sugest you to use
IAX2. It is better in some ways. And I sugest you to ensure you
compiled the speex codec and use it if
2006/3/7, Zach A [EMAIL PROTECTED]:
I have SPA3000 receiving PSTN calls and also have a SIP line on the same
Asterisk server with 5 extensions. Now there is a fax too which comes
through the PSTN line. Fax calls have short rings. Can Asterisk somehow
detect those short rings and send the call
Hello Gabriel,
IMHO, using IAX between * servers is a good choice, I dont see any problem in it. Actually I used it for sometime and never encounter any issue, but i had max 5 concurrent connections.
regards,
Umair bari
On 3/7/06, Gabriel Afana [EMAIL PROTECTED] wrote:
Hi everyone, I just spend
2006/3/7, bmw suzuki [EMAIL PROTECTED]:
how can i call a regular PSTN landline phone Via this software through
internet?Do i need dedicated hardware for this or an ethernet card will do.
To call to PSTN lines there are some alternatives:
1) install FXO hardware in your asterisk server. This
Hi,
I have two Asterisk server linked by a IAX2 trunk with two PRI+DID:
Site1: XXX100-499
Site2: YYY100-499
(I masked real number with XXX and YYY)
PSTN PRI1 --- Asterisk1 ...IAX2... Asterisk2 --- PSTN PRI2
Users:
- keep their extension when moved between sites
- can be reached from PSTN
On Tue, 2006-03-07 at 05:04 -0500, Sina Bahram wrote:
However, as I pointed out in my email, that doesn't make any difference.
If I leave it commented ... I get the exact same thing: just minus the make
file error
Same behavior, same error messages with the /etc scripts, the modprobe's and
Well, with music I manage to get the
periodic-announce to work, but only when the queues had no available members it
will play the periodic announcement within the periodic-announce-frequency.
To manipulate this I figured out that you can
set the timeout within queues.conf to:
Dear friends:I have seen Tzafir, Anton, Cosmin, Colin and other very interesting peopleworking very hard with the fax, almost at the point to write a book (I hope some day they will for all of us). I have been reading and saving all of those mails carefully to find the key to my needs. But
Hi
I use Asterisk 1.2.1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of leonimar cape
Sent: Tuesday, March 07, 2006 12:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] and H323
Hi Viktor,
Wilson Pickett wrote:
I use 3 which is the default on my 501's and 600's
No echo here
Actually, admin docs warn us NOT to change this value, but I am not in
the US. I don't always have echo, but when there is echo it almost
always goes away by lowering the level into the handset (or
Someone have good sound on ChanSpy with SIP channelsa at an Asterisk 1.2.4 ?Mine is cracking all the time.-- Adrià Vidal
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I have done my make webvmail, what else do I need to do? How
do you get to the site? Any help would be appreciated.
Jordan Novak
Communications Technician
Logistics Health Inc.
1319 Saint Andrews Street
La Crosse WI 54603
1-800-666-2833 x299
(608) 783-7560 x299
Hi everybody,
I have the same problem, but I have just upgraded to 1.2.5.
The changelog says this problem is fixed in this version, but I don't think so.
Asterisk is still crashing.
[]s
Alex Robertson
2005/12/18, Mohammad Shokuie shokuie at hotmail.com:
Hi there,
Any one confronted a crash
Good day!
Is is possible to change dialtone (and other tones as well) in BT-102?
pgpH067L1PT2h.pgp
Description: PGP signature
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|While I'm asking about the Polycom ip500, the answers for all phones
|where mic/handset/headset levels are adjustable would be of
interest to
|many I'm sure.
|
|For the ip500, the default value for the handset seems to be
|voice.gain.tx.analog.handset=3
I have a number of IP600s and 601s
I'm almost afraid to ask, but is the HT 386 known for having a lot of troubles?
I just installed one at home about 2 weeks ago, and knock on wood, it's only
locked up once, and this was when I was still in the process of tweaking the
config to work optimally w/ [EMAIL PROTECTED] I can't say
Hi try http://www.grandstream.com/y-downloads.htm
Download the IP Phone Custom Ringtones Generation Tool
Unzip and read the readme
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dmitry
Ivanov
Sent: 07 March 2006 13:40
To:
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] Up
Dial(SIP/[EMAIL
I al surprised that you are hearing anything at all.
the setting you have in your sip.conf ionstructs * to allow the end-points to
send the 'voice' directly betrween them.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Guarnido Sent: Tuesday, March 07,
On Tuesday 07 March 2006 15:49, Lee Archer wrote:
Download the IP Phone Custom Ringtones Generation Tool
Unzip and read the readme
Ringtone != dialtone.
pgpEye3ebfT7t.pgp
Description: PGP signature
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Sorry... Just ignore me.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dmitry
Ivanov
Sent: 07 March 2006 14:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to change Budgetone dialtone?
On Tuesday
Hi All,
I am looking to see if this is possible and any
pointers if it is. It seems straight forward but not too
sure.
I have 4 extensions 2000 to 2003
I have one voip external account with Sipdiscount.
I want any of the 4 extensions to share that single sipdiscount
It will lockup on you every 10 days or so and if you use both ports
and have calls at the same time then good luck. Does your call
waiting work on both ports? Mine does not.
On 3/7/06, Steve Jones [EMAIL PROTECTED] wrote:
I'm almost afraid to ask, but is the HT 386 known for having a lot of
Regards!
During the use of areski a2billing software
I'm getting same problem all the time.
Actually, after 15 minutes ofspeaking
to someone over calling card, connection brakes.
Installation was as smooth as it could be
so I don't think I made same kind of a mess in that domain. This is
How can I send recordings, that I have recorded with One Touch Record, to
e-mail address that is defined in voicemail.conf?
Thank you for your ideas.
--
Tomislav Parcina
tparcina#lama.hr
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Hi all,
I got also your question, how to use nwebmail?
Nwebmail is used for administration mail reports, i think.
Take a look on this:
http://www.vozdigital.org/modules.php?op=modloadname=Newsfile=articlesid=95
I've made login with
user: admin
password: mypassword_for_admin
I'm developing a
On Tue, 2006-03-07 at 12:08 +0200, Viktor Tatianin wrote:
Hello
I attempt installing H323 at my [EMAIL PROTECTED] for this use
asteriskathome-h323-1.0.zip but have next problem
chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory
chan_oh323.c: In function
Hi all,
I'm using [EMAIL PROTECTED] 2.5, and i've done:
1-Create a SIP extension.
2-Leave there a Voicemail message
3-Remove SIP extension
Then I've create another SIP extension but with the same number of the
above one.
I found imediately a voicemail message in my voicemail box.
Is this a
Hi all!
I have the following setup:
Phone lines - traditional PBX - Welltech 3802
- VPN -
Asterisk - Linksys PAP2/Welltech ATA-151 - phone
There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2)
PBX extensions. Asterisk is a proxy here. Each device successfully
register itself. I
On Mon, 2006-03-06 at 15:42, Jerry Geis wrote:
/ here is some of the output. I am no longer the to spcifically do sip
// debug but this is what I have.
// along with my sip.conf snip.
//
// The call to extension 3726 never rings. so it never gets answered.
//
/
Are you sure your sip trunk
Hi guys,
I am busy planning to implement SE Linux on my asterisk box. Either
that or I will use AppArmor from Suse.
I just want to know what are others experiences/incidents with SE Linux
or AppArmor
thanks,
yusuf
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Or hardcode the Digium URL in your script and on failure grab from your
mirrors, and to make absolutely sure your mirror should resolve to a DNS
name and then if *that* fails, a hardcoded IP. That way, you get 3 layers.
-Original Message-
From: Joseph Tanner [mailto:[EMAIL PROTECTED]
Whey you 'destroy' a Sip extension you are only removing the entrys that
allow you to make and receive the auth needed to do so. Your voicemail
files are not tied to an extension but are independent and are only
'married' when you specify it in your sip.conf or other channel configs.
Removing a
Hello everyone,
I am in need of 20 US toll free nos and 10 non toll
free nos, termination using IAX. Are there any reliable companies that you can
recommend?
Thank you
With regards,
San
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My question is about webvmail, not nwebvmail. I have never
used AMP (seems like cheating). My question is in regards to plain jane
Asterisk install. Just like making samples after you compile asterisk you are
able to make webvmail. Basically it is a interface into the voicemail system
fro
Hi Paul
I am looking to see if this is possible and any pointers if it is. It seems
straight forward but not too sure.
I have 4 extensions 2000 to 2003
I have one voip external account with Sipdiscount. I want any of the 4
extensions to share that single sipdiscount
Apologies if this is an old question; I've searched the list and the
wiki but have not been able to find a definitive answer.
I have an Aastra 480i phone registered with * 1.2.4; I want to generate
UK ringback tones when the handset dials another internal extension. On
my Zap channels, I have
I have
a configuration where RTP traffic is going out interface pub0, and coming back
into through pub1.
I have
bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows:
udp 0 788
0.0.0.0:5060
0.0.0.0:*
which
means that Asterisk is listening on all addresses (on all
interfaces?).
Anyway,
On Tue, 7 Mar 2006 09:12:25 -0700
Douglas Garstang [EMAIL PROTECTED] wrote:
I have a configuration where RTP traffic is going out
interface pub0, and coming back into through pub1.
I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an
shows:
udp0788 0.0.0.0:5060
Regards!
During the use of areski a2billing software
I'm getting same problem all the time.
Actually, after 15 minutes ofspeaking
to someone over calling card, connection brakes.
Installation was as smooth as it could be
so I don't think I made same kind of a mess in that domain.
Hi all,
I'trying to use spandsp (app_rxfax) to receive faxes.
When there are more than one page, the system creates a tiff file with only the
first page and the other are lost, even if the full log says:
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c:
Asterisk does not like multiple interfaces in the way you are configured. You
can either:
A) use the bindaddr in the sip.conf to limit where the packsge come and go.
B) use an outside traffic manager
Look up the archives, kpf explained why this would not work, as asterisk can't
do load
Pardon my candour, but for a product Digium calls 'enterprise grade' it sure
seems to be missing a few features.
-Original Message-
From: Alexander Lopez [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 07, 2006 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi friends,
I am using Real Time Asterisk Architecture where I have put
the
Sip users/peers and extensions defining the dialplan in
tables in
a mysql database.
Currently, asterisk points to my single database server as
configured:
--
Hello everyone,
Please forgive the exclamation points but I have been battling this one
off and on for about four days now. Sorry for the cross post.
It all started with a box of IP 501s. I contacted my reseller and
obtained the latest BootRom and SIP firmware. Unzipped, configured,
I compiled the newest version of * from cvs about a week ago on Fedora. I think all I had to do after issuing the make webvmail was install the perl and perl-suidperl packages. I got that information (and anything else I might have done but forgotten) by searching for
webvmail.cgi at
Hi,
In December, I posted an enquiry asking whether anyone had experience
with the Sangoma A104d cards (see below) - I got a couple of
responses, but basically it was that people have started playing with
them, and would publish feedback at a later date.
Does anyone have any further feedback at
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Sent: Tuesday, March 07, 2006 6:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6
kernel
On Tue, 2006-03-07
Are
you kidding? Asterisk doesn't do SRV. If you read all the VOIP books out there,
SRV lookups are _the_ way to achieve redundancy. Digium hasn't gotten to it I
guess with their 'enterprise class' product though.
-Original Message-From: Ramin Nikaeen
[mailto:[EMAIL
Steve,
I just complete a setup of asterisk server in a production environment with
a single A104D and there is no echo and the quality is okay.
goksie
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: Tuesday, March 07, 2006 6:06 PM
Hi d_pejic,
First of all, please never send several times the same question to the
list, it's really
not respectful for the others. Your issues should not pass in priority
from others.
As Kpfleming pointed out, Add-Ons/A2Billing are off topic for this
list, so please redirect
add-ons question
thats good to hear .but there are so many digium cards that does echo distortion then why do you want to do this Giridhar Bandi On 3/7/06, Robert Rozman
[EMAIL PROTECTED] wrote:
Hi,I'm speech recognition researcher and would like to do some research onrecognition robustness in echo distortion of
I spent a weekend battling similar issues with 501s, using FC4/
proftpd. I finally switched from FTP to HTTP.
On Mar 7, 2006, at 9:53 AM, Kristian Kielhofner wrote:
Hello everyone,
Please forgive the exclamation points but I have been battling
this one off and on for about four days
Hi everybody,
this is not directly related to Asterisk, but I'm sure this is the place
to get an answer:
Even with Asterisk not running, the entire system will crash when a call
comes in through CAPI. This only happens when the caller does not submit
his caller-id.
I'm using the following
On Mar 7, 2006, at 2:38 AM, Giridhar Bandi wrote:
ya i found it it *1 to start recording from the caller end
Also pushing *1 again stops recording.
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To
Hi i am using pap2 phone adaptors as clients to connect to asterisk server i am able to make calls but i cannot access voice mail using phone or start recording while call is in progress and when i place a call to local sip extension there is a long pause ( 15 sec )
before the call gets dialled i
7 mar 2006 kl. 17.00 skrev Can2002:
Apologies if this is an old question; I've searched the list and the
wiki but have not been able to find a definitive answer.
I have an Aastra 480i phone registered with * 1.2.4; I want to
generate
UK ringback tones when the handset dials another internal
Hey thanks for the prompt response ( that's what i liked about this list ) i was not able to start recording i have pap2 box as clients and the dial plan of pap2 is as bellow (*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.)
can you suggest if this is causing the problem
7 mar 2006 kl. 18.12 skrev Douglas Garstang:
Are you kidding? Asterisk doesn't do SRV. If you read all the VOIP
books out there, SRV lookups are _the_ way to achieve redundancy.
Digium hasn't gotten to it I guess with their 'enterprise class'
product though.
Are you kidding? We do SRV.
HI All,
What is a good tutorial or article on using Xlite to get into * while
doing so over the Internet?
I have had problems with doing this by having one way audio. I had
searched around and not found an article that addressed the problem
Thanks,
--
Leonard Burton, N9URK
[EMAIL PROTECTED]
I have bought more than 20.
Maybe 2 of them work well...
:-(
I have to make cold reset on the ATA_386 every days...
Regards
Amr
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Jones
Sent: terça-feira, 7 de Março de 2006 13:40
To: Asterisk Users
My bad. SRV lookups work, but Asterisk only uses the first entry right? This
means there's no redundancy.
-Original Message-
From: Olle E Johansson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 07, 2006 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Ugh.. That's not good news... I guess I have wither a digium card or a Sipura
FXS in my future unless I'm one of the lucky 10% then!! :-) Thanks for the
feedback!
From: Andre Rodrigues (Cheyenne) [mailto:[EMAIL PROTECTED]
Sent: Tue 3/7/2006 1:00 PM
To:
Is the softphone behind NAT ? If it is insert nat=yes
in your dial plan. Is the server behind NAT ? If it is
you need to open ports 5060,5061 and 1-2.
Dovid
--- Leonard Burton [EMAIL PROTECTED] wrote:
HI All,
What is a good tutorial or article on using Xlite to
get into * while
take a look at the following things- enable nat on asterisk - if you are using a perimeter firewall then forward port 5060 , 1 -2 ( these are default ) - use correct sip proxy address on you xlite phone
--Giridhar Bandi On 3/7/06, Leonard Burton [EMAIL PROTECTED] wrote:
HI All,What is a
Helo List,
First I would like to apologize for my bad spelling as
well as that I did not search the wiki first. I only
have email access at the moment.
I am having trouble setting both variables and global
variables thru an extension.
I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4
with an
Marco Maiolini wrote:
Hi all,
I'trying to use spandsp (app_rxfax) to receive faxes.
When there are more than one page, the system creates a tiff file with only the
first page and the other are lost, even if the full log says:
You need a fax viewer that can handle multi-page tif files
7 mar 2006 kl. 19.03 skrev Douglas Garstang:
My bad. SRV lookups work, but Asterisk only uses the first entry
right? This means there's no redundancy.
That is correct. That is what we try to fix.
/O
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Yeap... very bad feedback...
But I think that the HT 286 model had the same problem, and now they are
working well.
I will have 3 of them next week to replace the HT 386 models that are using
fax lines and working very bad, but consider that the HT 386 hangs a loto f
times and I don´t have
All - I've been muddling around with this for a few days now.. and I'm
trying to figure out why I am not receiving more than one phone call on
each polycom 501 phone. I can make more than one phone call out, but not
receive another one in, while on a call. Has anybody seen this behaivior
I've an Asterisk server running in my office, which forwards all
long distance calls to a third party SIP service using an extension rule:
exten = _1XX0.,1,Dial(SIP/{EXTEN:[EMAIL PROTECTED])
(1XX0 is the international calls rule for Chile)
Also, in my sip.conf, I've defined canreinvite=yes to
Hello,
Works great for me as well. over 3 months in production with no
problems/no echos.
MATT---
On 3/7/06, ADEGOKE ARUNA [EMAIL PROTECTED] wrote:
Steve,
I just complete a setup of asterisk server in a production environment with
a single A104D and there is no echo and the quality is okay.
HTTP's nice, but FTP does the job. Check the docs for supported FTP
servers -- many of the stock Linux FTP servers will give the exact problem
you discussed, below. I should know -- took me almost a week before
trying proftpd, and WHAMMO, worked like a champ.
-Ken
On Tue, March 7, 2006 12:37
Hello,
Is Call Path Optimization (IAX Draft, Section 6.4.4) supported by
Asterisk? If not, is there a roadmap for it?
If there is a URL I can study to get the answer, I will appreciate
a pointer. I scanned recent mailing list archives and couldn't find
any discussion of this topic.
Thank you.
As far as I know, you will need to do this yourself with some creative scripting. There was some talk on the list awhile ago to move the recording tovoicemail, but I dont' know if anyone has made a patch to do it yet.
On 3/7/06, Tomislav Parčina [EMAIL PROTECTED] wrote:
How can I send
On Mar 7, 2006, at 9:51 AM, Giridhar Bandi wrote:
Hey thanks for the prompt response ( that's what i liked about this
list )
i was not able to start recording
i have pap2 box as clients and the dial plan of pap2 is as bellow
(*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.)
Hey all,
I have a client whose previous programmer ditched. I'm his webmaster,
and now he wants me to have an asterisk system set up for serial
number authentication...and I have a digium card from the previous
guy. Are there hosts that will set this up for me? ie, rack space
somwhere? Are
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