On 08/01/2005 15:01 Dean Thompson said the following:
Just not sure whether it is SJPhone playing up and not sending the right
signal for the #, or whether there is a more underlying problem here.
in which case you could try using a dialplan with Read() and SayDigits() to
see of the SJPhone is ac
On Fri, 7 Jan 2005, Eric Bishop wrote:
> The problem definately seems to be G4 server related, most folks with
> G3's seem to be unaffected. Anyone know where there might be a
> changelog of G3->G4? That might reveal some pertinent information.
2nd hand info from HP seemed to be that the main
Hi,
Dinesh Nair wrote:
On 08/01/2005 11:35 Dean Thompson said the following:
Thanks for pointing that out. I have made the appropriate
configurations, but the system stil refuses to accept the pin for the
conference.
The definition now looks like the following:
meetme.conf file:
conf => 1234 OR
On 08/01/2005 11:35 Dean Thompson said the following:
Thanks for pointing that out. I have made the appropriate
configurations, but the system stil refuses to accept the pin for the
conference.
The definition now looks like the following:
meetme.conf file:
conf => 1234 OR
conf => 1234,5678
+-->
On 07/01/2005 03:22 César Davi Ávila do Nascimento said the following:
Hi all,
I'm trying to install a TDM400P card, and I need some help.
Please, see below...
**
*after dmesg command:*
did you run ztcfg to configure the zaptel devices /before/ starting
asterisk ?
--
Regards,
The company I work for has gotten the go ahead to start dipping its foot
into the shallow end of the asterisk pool. The client we will be setting
up for currently has an NEC PBX of some sort with 8 analogue lines in.
They use lines 1-4 as indial on a rotary group, lines 5-6 as indial for
two 1800-
Thanks Matthew, Asterisk and PBX is new to me.
There is my sip.conf file. Below that is the extention.conf file.
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = default ; Default context for incom
Can someone help me answer this question?
Where would you most likely find a file with the line "+::"?
What does it do?
I have been racking my brain a buddy of mine is testing me and I don't want
him to win.
Thanks
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Hello All,
I have Cisco 7960's, Cisco 2950 Switch. Here
is my issue I can dial out no issues but when someone calls in the phone rings I
answer and the phone disconnects the call.
Call from my cell to my house I answer the cisco
phone is disconnects at the same time on the cell I hear 4
Michael Welter wrote:
Steve Underwood wrote:
If your TIFF are consistently that small you probably have a TIFF
library problem. Frame slips generally give much more eratic results. If
Steve, when you say that some Canon fax machines "disconnect during
negotiation", does this negotiation happen
I am having a major dillema here, I have been trying to get my sip phone (hard
phone) to communicate with the asterisk server. Below is my configuration:
sip.conf
[1201]
type=friend
username=1201
secret=
mailbox=1201
host=dynamic
[1202]
type=friend
username=1202
secret=
mailbox=1202
host=dynami
How about Houston TX?
On Fri, 07 Jan 2005 20:07:04 -0800, Steven P. Donegan wrote:
>On the same general subject, Asterisk users get-togethers, who might be
>interested in sharing conversation in the Disneyland/Knotts Berry Farm
>area in Orange County, California?
>_
On the same general subject, Asterisk users get-togethers, who might be
interested in sharing conversation in the Disneyland/Knotts Berry Farm
area in Orange County, California?
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MINNESOTA: TwinCities Asterisk Users Group - Meeting Saturday 01/08/05
Hello Fellow Asterisk Users!
This email is to remind everyone of the TwinCities Minnesota users
groupmeeting. Information is located on the Wiki site.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20User%20Group%20Tw
Hi!,
[...]
I know this question may have been asked before (although the archives
don't seem to suggest it), but has anyone had any problems with Asterisk
accepting a PIN number for a conference room.
At this point in time I have established the conference definition in
the meetme.conf file as well
> -Original Message-
> From: Adam Fineberg [mailto:[EMAIL PROTECTED]
> Sent: Saturday, January 08, 2005 2:38 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] New 'n' priority
>
>
> Christopher L. Wade wrote:
>
> > 'n' and 's' as well as
Christopher L. Wade wrote:
Bill Seddon wrote:
Can anyone point me to documentation covering the new 'n' priority.
I've
downloaded and have working v1-0-2. But when I attempt to use the n
priority - for example:
exten => s,1(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten =>
s,n(dial),Dial(SI
Hi
all,
Does anyone know how
to get the channel ID on the other side of the call?
For example: When
SIP/50 calls SIP/21, and the call is answered by SIP/21 I
get:
SIP/21-6735 answered
SIP/50-b456
${CHANNEL} will show
me SIP/50-b456.
Is there a parameter
or a workaround to get the SI
The zaptel 1.0.3 source fails to compile just as you have described. (I have
been trying to debug it for an RPM I build).
Use the CVS version - it works on 2.6.9 with FC3 so I imagine it should work
with 2.6.10?
Jeff
- Original Message -
From: "Dave Cotton" <[EMAIL PROTECTED]>
To: "Aste
save biiling info into database instead of file.
On Saturday 08 January 2005 04:04, you wrote:
> Hello
>
> Finally, i authenticated the user with database. and got this message.
>
> -- SIP Seeding '12345' at [EMAIL PROTECTED]:5060 for 1800
>
> Special thanks to Mr. Matthew Boehm.
>
> Can you plz
Hello
Finally, i authenticated the user with database. and got this message.
-- SIP Seeding '12345' at [EMAIL PROTECTED]:5060 for 1800
Special thanks to Mr. Matthew Boehm.
Can you plz. guide me how i can configure prepaid with this setup.
Thanking you again.
__
Bill Seddon wrote:
Can anyone point me to documentation covering the new 'n' priority. I've
downloaded and have working v1-0-2. But when I attempt to use the n
priority - for example:
exten => s,1(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten => s,n(dial),Dial(SIP/1001)
Asterisk generates e
- Original Message -
From: "Serge Schumacher" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Friday, January 07, 2005 12:45 PM
Subject: RE: [Asterisk-Users] Sip protocol question ...
> What control is it ?
>
http://www.pernau.at/kd/voip/ActXP
Can anyone point me to documentation covering the new 'n' priority. I've
downloaded and have working v1-0-2. But when I attempt to use the n
priority - for example:
exten => s,1(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten => s,n(dial),Dial(SIP/1001)
Asterisk generates errors containing t
Hi,
I would like to know what would be acceptable latency
on a connection to the termination server( but still
having good quality voice )
Thanks,
robert
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Steve Underwood wrote:
If your TIFF are consistently that small you probably have a TIFF
library problem. Frame slips generally give much more eratic results. If
Steve, when you say that some Canon fax machines "disconnect during
negotiation", does this negotiation happen at the end of each page
Hi,
If you have done so I would appreciate copies of the
relewant configuration files.
Thanks,
robert
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The reason, David, that nobody has responded to your help request is because
you haven't given any information for us to help debug your problem. You
need to provide alot more information. snippents from console are good,
snippets from sip.conf, extensions.conf and the like.
"Hi my car won't start
On Fri, 7 Jan 2005, Michael Welter wrote:
> Nils Segerdahl wrote:
>
> > I suspected frame slips. But sound quality was ok. Everything was ok.
> > Except maybe to many interrupts on the line cards.
> >
> > But when I changed my devicedriver to one that was more safe regarding
> > lost interrupts it
On Friday 07 January 2005 03:45 pm, Steve Underwood wrote:
> If your TIFF are consistently that small you probably have a TIFF
> library problem. Frame slips generally give much more eratic results. If
> you believe you have a good version of the TIFF library on your machine,
> check you don't have
Hello All,
I loaded [EMAIL PROTECTED] I'm using SLPhones and can connect to mailboxs
on the system. I have one X100P card. I try to dial out but get
rejected.
Any help...
Thanks, David
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ht
test
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Nils Segerdahl wrote:
I suspected frame slips. But sound quality was ok. Everything was ok.
Except maybe to many interrupts on the line cards.
But when I changed my devicedriver to one that was more safe regarding
lost interrupts it worked like a charm.
Device driver? TDM device driver? Would you
On Sat, 8 Jan 2005, Steve Underwood wrote:
> Ryan wrote:
>
> >On Friday 07 January 2005 01:35 pm, Lee Howard wrote:
> >
> >Just to be clear. What I am getting is about the top inch of the fax and
> >that's all. The total size of the TIFF is ALWAYS about 3KB or less.
> >
> >
> If your TIFF are consi
Steve Underwood wrote:
If your TIFF are consistently that small you probably have a TIFF
library problem. Frame slips generally give much more eratic results. If
you believe you have a good version of the TIFF library on your machine,
check you don't have any others. A *lot* of people have told
Ryan wrote:
On Friday 07 January 2005 01:35 pm, Lee Howard wrote:
Since spandsp doesn't use ECM, what I'm about to say doesn't apply to
spandsp. If you receive truncated (versus corrupted) fax images from
spandsp, then I'm not sure what the problem would be. What I'm about
to say only applies
While I appreciate your response (as much as anyone else's) I wonder if you
are referring to the link on the FAQ here
(http://www.opencall.org/faq/x26.html)? It tends to blame Fax modems where I
stated that I ALSO used a Fax machine. No modems just a big old plain paper
fax -remember those everyone
Basically the changes in the apps/Makefile have progressed while the patch
makefile have not. Here is a current patch that works as of
CVS-HEAD-01/06/05-14:47:06
Regards,
Jim
On Fri, 7 Jan 2005, Altus Snyman wrote:
> I'm trying to install spandsp
> But when I try to patch the Makefile it gives
Doesn't look like you have CVS HEAD with the realtime code... (unless they
have realtime in the stable branch now... I'm just a newbie, please, noone
attack.)
Clay
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, January 07,
- Original Message -
From: <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Friday, January 07, 2005 7:20 PM
Subject: [Asterisk-Users] Loading module app_realtime.so failed!
Can anyone help me with this: -
[EMAIL PROTECTED]:~ > asterisk
[EMAIL P
On Friday 07 January 2005 11:37 am, Nils Segerdahl wrote:
> There was a discussion on this list a short while ago on howto debug
> frameslips.
> [..]
> When I applied Florian Zumbiehls patch the problem went away.
> (The link to the patch can be found in the wiki: asterisk zaphfc)
>
> Is it a pos
> I have used an analog trunk (FXO) AND a station (FXS) both on the same card.
> I thought that it might be related to the hardware so I hooked up an old
> Brother Intellifax 9000 on the station port. Both of these attempts had the
> same problem.
>
> It is my speculation that the 'cutoff' problem
On Friday 07 January 2005 01:35 pm, Lee Howard wrote:
> Since spandsp doesn't use ECM, what I'm about to say doesn't apply to
> spandsp. If you receive truncated (versus corrupted) fax images from
> spandsp, then I'm not sure what the problem would be. What I'm about
> to say only applies to ECM-
Can anyone help me with this: -
[EMAIL PROTECTED]:~ > asterisk
[EMAIL PROTECTED]:~ > asterisk -r
Asterisk CVS-v1-0-12/28/04-11:37:32, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
=
Connected to As
PHP Mechanic schrieb:
Hello everybody,
does anybody knows from where I can get an list of international area
codes incl. the mobile numbers?
Have you tried google ?
http://www.google.com.au/search?hl=en&q=international+dialing+codes
Yes, I had tried that already. The search results containing no
Jeff wrote:
It is my speculation that the 'cutoff' problem was related to some type of
'line noise' and that others successfully using the spandsp code _might_ be
using T1/E1 rather than analog lines (1FL) but when I started testing using
an old Fax machine plugged into a station port with a six fo
I'm interested in it. It would be very nice if you can send me your
list. :-)
But from where you got the Informations?
Regards
Bastian
Sebastian Nocetti schrieb:
I can send a list, mobile is not complete but it has a lot of numbers...
-Mensaje original-
De: [EMAIL PROTECTED]
[ma
<>
Yes, and if the multiple extensions that ring are members of the same group
then any one of the phones can pickup the call.
So the next question is: how does the receptionist put the system into
"group ring" mode. The answer is to have the receptionist call a nominated
number such as **221 (e
Attempted to get this info from Digium but my efforts have failed...
I am thinking of getting a TE410P from digium.
My local Telco uses B8ZSESF and does support PBX customizing ANIs on a per
call basis.
What I need to know is, can I use the SetCallerID command in extensions.conf
to transmit the
ok.
you have in your res_odbc: dsn=> test
but you don't have a dsn called test in any of your odbc config stuff.
-Matthew
- Original Message -
From: "Muhammad Rizwan Khan" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, January 07, 2005
Default for IP 500 (prolly the other too, but not sure)
username: PlcmSpIp
password: PlcmSpIp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Friday, January 07, 2005 9:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
On January 7, 2005 12:26 pm, Matthew Boehm wrote:
> > Seems to be correct, or at least image corruption from a really crappy
> > fax reception. I know I've been receiving between 30-50 faxes a day with
> > app_rxfax without issue.
>
> What versions of everything are you using? Using PRI? libti
Hi, i have setting up asterisk for mysql. i using the template-database:
sipfriends.
i have a vpn in the office. i like to setup asterisk:
when a client make authentification request: username and password stores
automaticlly in the sql database.
any users in the vpn can setup the own name and
Theres your problem right there; All of them say line2_X
Nathan.
# Line 2
line2_name: Scott1
line2_authname: "scott1"
line2_password: "scott1"
# Line 3
line2_name: "Line 2"
line2_authname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"
# Line 4
line2_name: "Line 4"
line2_authname: "UNPROVISIONED
You can Dial() extension SIP/line1&SIP/line2
even more you can and that will call both extensions only after a 5 seconds
timeout
exten => xxx,1,Dial(SIP/line1,5)
exten => xxx,2,Dial(SIP/line1&SIP/line2,10)
etc...
that's if I understood what ou needed...
bye,
M.
- Original Message -
Fr
On 2005.01.07 09:42 Jeff wrote:
It is my speculation that the 'cutoff' problem was related to some
type of
'line noise' and that others successfully using the spandsp code
_might_ be
using T1/E1 rather than analog lines (1FL) but when I started testing
using
an old Fax machine plugged into a statio
streamload.com
dropload.com
- Original Message -
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>
To:
Sent: Friday, January 07, 2005 5:25 PM
Subject: Re: [Asterisk-Users] Moderator on vacation?
> On January 7, 2005 11:22 am, Andrew Thompson wrote:
> > I can't get google to show me any, but
-Original Message-
Hey I noticed this posting, is anyone in New York interested in catching
up?
I'd be happy to host it at my place on 72nd/york if it wasn't too big a
group, or we can always head out and grab some lunch or something
somewhere.
Email me your interest and we'll see what the
Someone on the list spotted the problem, there is a typo in my line
definitions.
Thanks all
Scott Henderson wrote:
I set this up manually on the phone and it works just fine so config
files ... I attached the complete config files so maybe someone can
see what I am missing.
==
Check this out.
http://voip.weblogsinc.com/entry/0142584371536804/
David
On Fri, 2005-01-07 at 09:15, Richard Cook wrote:
> Hello,
>
> Is there any way to unlock the Linksys router?
>
> --
> Richard Cook
> [EMAIL PROTECTED]
> Tel: 705-497-9320 ext 2010
>
> --
> This message has been scanne
Please find the attached files,
Thanks
On Friday 07 January 2005 22:24, you wrote:
> post your /etc/odbc.ini and /etc/odbcinst.ini
>
> -matthew
>
> - Original Message -
> From: "rizwan" <[EMAIL PROTECTED]>
> To:
> Sent: Friday, January 07, 2005 10:19 AM
> Subject: [Asterisk-Users] Aster
I set this up manually on the phone and it works just fine so config
files ... I attached the complete config files so maybe someone can
see what I am missing.
argon:/tftpboot# cat SIPDefault.cnf
# SIP Default Generic Configuration File
# Image Version
image_version: P0S3-07-3
im using libtiff-3-7 and im getting corruption constatnly. I posted to
Steve's bug site but I've not heard from him in over a month.
i guess he's still on vacation.
-Matthew
- Original Message -
From: "Ryan" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion
> Seems to be correct, or at least image corruption from a really crappy fax
> reception. I know I've been receiving between 30-50 faxes a day with
> app_rxfax without issue.
What versions of everything are you using? Using PRI? libtiff? spandsp?
asterisk? diagram? I can't get any faxes via r
Title: Re: [Asterisk-Users] Ringing an extension on multiple phones
There are several options here.
You can set up a queue and have the phones ring un the order you like.
Setup an additional extension on every phone.
Set up an AGI script that allows them to login to the receptionist calls.
I have the same problem and thought I would wait for someone else to post...
(just kidding Ryan)
I have used an analog trunk (FXO) AND a station (FXS) both on the same card.
I thought that it might be related to the hardware so I hooked up an old
Brother Intellifax 9000 on the station port. Both o
Title: Re: [Asterisk-Users] can the dialtone be changed after pressing 9?
Yes you can but it only works for zap devices. IP based would be a function of the hardware.
-Original Message-
From: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Fri Jan 07
Nils Segerdahl wrote:
On Fri, 7 Jan 2005, Ryan wrote:
I had the same problems using hfc cards with bristuff. (with patched
zaptel drivers).
Which zaptel patches did you use?
Thanks
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
On Friday 07 January 2005 11:24 am, Andrew Kohlsmith wrote:
> I also note that you posted your initial message at 4:14pm, and now, less
> than 24 hours later you are expecting the entire asterisk community to have
> received your message, parsed it in the sea of other messages to the list,
> had it
I use the CDR CVS file for logging my home phone system. Can I force write
data to a CDR Field from an extensions macro? Say if the callerid was empty
and I dumped the call to put data in the CDR to let me know that is what
happened.
Thanks
--John
___
A
post your /etc/odbc.ini and /etc/odbcinst.ini
-matthew
- Original Message -
From: "rizwan" <[EMAIL PROTECTED]>
To:
Sent: Friday, January 07, 2005 10:19 AM
Subject: [Asterisk-Users] Asterisk with MySQL
>
> Hello
>
> I am getting this error message, when i try to authenticate my users
t
Hey I noticed this posting, is anyone in New York interested in catching
up?
I'd be happy to host it at my place on 72nd/york if it wasn't too big a
group, or we can always head out and grab some lunch or something
somewhere.
Email me your interest and we'll see what the numbers are.
Cheers,
De
Hello,
Is there any way to unlock the Linksys
router?
--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320 ext
2010
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To UNSUBSC
> I had not looked at the phones settings yet, thanks for the
> suggestion. The setting indicate that there is no configuration on the
> second line it is listed as "UNPROVISIONED"
Go into the phone and program Line 2 Settings directly, without using
the SIP.cnf file. If that works, then your .cnf
On Fri, 2005-01-07 at 09:18 -0700, Wiley Siler wrote:
> The FTP server option works very well so you should do it when get time.
>
> The phone has an option where you tell it to load via FTP, believe it is
> the server config.
> To get to it, reboot the phone and enter setup on the phone, not the
Greetings,
Does someone have the link to reset your password on bugs.digium.com?
I can't seem to find one.
Thanks.
--
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
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I am using Cisco 7960 phones and have had a request to have the
receptionist phone ring on multiple phones just in case she is not around.
Call pickup is the theory here but the issue is that not all the people
that need to hear the ring would here the receptionist phone ring so I
think I need
Andrew Kohlsmith wrote:
If you got that message it means you posted to the list from an address that
is not subscribed. It's a little misleading -- I've *never* had a moderator
post or deny a message I've posted from a nonsubscriber address, on vacation
or not.
That may not be the only reason f
Um, that's about normal here. It runs like 16 threads on a fresh startup.
Maybe you don't have threading enabled on your box?
- Eric
On Fri, 7 Jan 2005 10:04:59 -0600
"Matthew Boehm" <[EMAIL PROTECTED]> wrote:
> Holy cow! Why are there so many asterisk instances running? There should
> only
extensions.conf has
ignorepat => 9
exten => _9X.,1,Dial(Zap/G2/${EXTEN:1})
The first user to try it asked if instead of keeping the same dialtone
after pressing 9, if I could play a different dialtone. Can this be
done? I'm running asterisk 1.0.0 in case that matters.
__
I think the issue is the context specification. In this application I
had two contexts in voicemail.conf that were not "default". I have
modified the sip.conf as suggested.
Scott
Nathan Alberti wrote:
Ensure you have "mailbox=" in sip.conf, you must also make sure in
voicemail.conf the m
On Friday 07 January 2005 16:08, Eric wrote:
> OK,
>
> I'm trying to send an email to the list the contiune a thread which
> describes a problem I'm having. This particualy email I wish to send
> contains an ls -l describing my problem (too many open files) and is
> apparently too large to be cons
On 2005.01.07 08:13 Ryan wrote:
H. Did I just ask in the wrong forum, or has _nobody_ experienced
image
corruption using app_rxfax that was NOT due to using the wrong version
of
libtiff?
Oh, you can get image corruption on any non-ECM fax, and that doesn't
have anything to do with anything oth
Andrew Thompson wrote:
Find a site, upload it there, post your message with info and point us
at the link.
And then everyone who is not involved in the thread about the OP's
problem will be very thankful!
To the OP: There is an obvious reason why the list does not allow
posting larger than a pr
On Friday 07 January 2005 16:04, Matthew Boehm wrote:
> Holy cow! Why are there so many asterisk instances running? There should
> only be 1.
>
> kill them all and start just 1 asterisk
Do not top post, learn to trim.
There is 1 process and many threads.
__
I did set these to the correct poxy serveras well in the SIPDefault.cnf
file.
This is very frustrating problem, I have ready dozens of posts that
refer to how to set this up and I see mto have followed all the suggestions.
I had not looked at the phones settings yet, thanks for the suggestion.
What version of sox do you use?
Lamine
- Original Message -
From: "Robert Spielmann" <[EMAIL PROTECTED]>
To:
Sent: Friday, January 07, 2005 2:40 PM
Subject: [Asterisk-Users] Monitoring
Hi,
I have some trouble with the Monitor() application. I start and stop it via
the management interf
On Fri, 7 Jan 2005, Ryan wrote:
> H. Did I just ask in the wrong forum, or has _nobody_ experienced image
> corruption using app_rxfax that was NOT due to using the wrong version of
> libtiff?
>
Hello Ryan,
I have.
There was a discussion on this list a short while ago on howto debug
framesl
On January 7, 2005 11:22 am, Andrew Thompson wrote:
> I can't get google to show me any, but there are sites that allow you to
> drop off large files and give you a url for retreiving them. Perhaps
> someone can come up with the name of one.
http://pastebin.ca is what is used on the IRC channel al
On January 7, 2005 11:13 am, Ryan wrote:
> H. Did I just ask in the wrong forum, or has _nobody_ experienced image
> corruption using app_rxfax that was NOT due to using the wrong version of
> libtiff?
Seems to be correct, or at least image corruption from a really crappy fax
reception. I kn
On January 7, 2005 11:08 am, Eric wrote:
> I'm trying to send an email to the list the contiune a thread which
> describes a problem I'm having. This particualy email I wish to send
> contains an ls -l describing my problem (too many open files) and is
> apparently too large to be considered a "no
Eric wrote:
Seriously, what gives. Can we make some changes here? I'd like to
post my findings and get some help.
I can't get google to show me any, but there are sites that allow you to
drop off large files and give you a url for retreiving them. Perhaps
someone can come up with the name of on
Hello
I am getting this error message, when i try to authenticate my users through
database.
Jan 7 20:28:08 WARNING[26487]: res_config_odbc.c:69 realtime_odbc: SQL Alloc
Handle failed! Jan 7 20:28:08 NOTICE[26487]: chan_sip.c:7974
handle_request: Registration from 'rizwan '
failed for '192.168
The FTP server option works very well so you should do it when get time.
The phone has an option where you tell it to load via FTP, believe it is
the server config.
To get to it, reboot the phone and enter setup on the phone, not the
web.
Remove the settings if you want no configs from network and
H. Did I just ask in the wrong forum, or has _nobody_ experienced image
corruption using app_rxfax that was NOT due to using the wrong version of
libtiff?
If that's the case, then my secondary approach is going to have to be:
PSTN <-> Asterisk + chan_h323 <-> t38modem + Hylafax
Is there a
Hi
I tried Xten, its very good, because it can stay in the taskbar (next to the
clock) and start when windows starts, and is allways ready to receive calls.
Maybe it s the best way to introduce VoIP to my company workers
But theres a feature that s missing (or I couldnt find), there s no way to
OK,
I'm trying to send an email to the list the contiune a thread which
describes a problem I'm having. This particualy email I wish to send
contains an ls -l describing my problem (too many open files) and is
apparently too large to be considered a "normal" post, so I get a
message that it's be
Holy cow! Why are there so many asterisk instances running? There should
only be 1.
kill them all and start just 1 asterisk
-Matthew
- Original Message -
From: "Eric" <[EMAIL PROTECTED]>
To:
Sent: Friday, January 07, 2005 9:35 AM
Subject: [Asterisk-Users] Asterisk 1.0.2 - Unable to all
I am in the process of setting up an * system using Polycom IP 500's.
I don't want to spend time setting a ftp server for application and
configuration files at the moment, just want to use the web interface
to the Polycoms. DCHP works OK and IP is obtained correctly.
Polycom fails to load .cfg fi
> -Original Message-
> From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
> Sent: Friday, January 07, 2005 3:26 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Queue app following dialplan
>
>
> The more reasonable solution is to just put
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