David wrote:
Hello All,
I loaded [EMAIL PROTECTED] I have one X100P card. I try to dail out but get
rejected.
Any help...
Thanks, David
Before someone else answers with a violent reply...
Your question, while reasonable, does not help anyone in helping you. Why don't
you try and provide more
Hi All,
I'm developing an outbound call center with 20 agents.
My configuration is like this.
PRI * NetGear Switch 20
iaxSoftPhone
I'm experincing bad voice quality and long delay.
I'm thinking about several possibilities.
1. NIC load - All NIC irqs process by CPU0.
I tried
Hi,
is it possible to transfer an incomming call to another ext. without
answering? I'm not talking about (un)conditional redirection but
functionality, when calee can each time decide whether answer the phone
or transfer it to any other phone.
___
Title: Message
I have configured
call pickup, and this works fine.
Although there are 2
problems, perhaps anyone would know a solution to this;
- When I pickup a
call from another set, the *8 code keeps being displayed in my screen (Snom
220).
I would like
it to show the phonenumber of
On Fri, 7 Jan 2005, Roger Schreiter wrote:
whatever dialplan I'm using for outgoing calls via
PRI (Digium card, chan_zap), the callerid when receiving
calls has no leading zeros, which are normally used to distinguish
local, national and international calls in Europe.
The number has always
Ramon Peek wrote:
- When I pickup a call from another set, the *8 code keeps being
displayed in my screen (Snom 220).
I would like it to show the phonenumber of the person calling me.
This is correct. You are placing a call to *8 which just happens to
connect you to caller. As far as your
On Thu, 6 Jan 2005, Andrew Kohlsmith wrote:
I imagine the Expansion is for more spans -- nothing has been designed for
them at this point. Timing is likely for carrying timing across multiple
cards, Test for testing and ident is for card order when multiple cards are
inserted into one
Hi
When I forward calls from SER (or GNUGK) to Asterisk, the SER ( or GNUGK)
are just used for signaling, but the call streaming passes from the endpoint
directly to Asterisk, isnt it? Or does the streming passes from the
Endpoint to SER and then to the Asterisk?
Thanks
Joao Pereira
I've an issue with my TDM4000P card and I will be calling Digium later
to ask for their help.
Could anyone help me with a basic configuration so they can SSH to me?
Thanks
John
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Hi,
I'm tryinig to debug SIP call from activex control based on MS RTC (A) to
Asterisk (B). I use Etherreal to follow packages and I would like to ask
short questions:
- Session trace shows following order of packets:
A - BInvite
B - A100 Trying
B - A200
Hi,
With Gnugk, make sure the proxy mode is not enabled if you want voice to
pass directly from endpoints.
Regards
Lamine
- Original Message -
From: Joao Pereira [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday,
Ok,
then I guess the way we use SER and GNUGK to redirect calls to Asterisk
makes the diference.
If we are using them as proxy, the stream will pass through them, if we dont
use proxy, they will be used just for signaling.
Joao
- Original Message -
From: Mamadou Lamine KA [EMAIL
What control is it ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman
Sent: vendredi 7 janvier 2005 11:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Sip protocol question ...
Hi,
I'm tryinig to
Yes,
This mode is generally used when some endpoints have private addresses
behind a NAT while others have public addresses.
In this case all the traffic passes through the GK.
Take a look at paragraph related to Proxy at
http://www.gnugk.org/gnugk-manual-4.html#ss4.2
Lamine
- Original
Hi all,
I'm trying to install a TDM400P card, and I need some
help.
Please, see below...
after dmesg command:
[EMAIL PROTECTED] root]# dmesgvia82cxxx: board #1 at
0xD800, IRQ 5Zapata Telephony Interface Registered on major 196PCI:
Found IRQ 3 for device 00:09.0PCI: Sharing IRQ 3 with
Please stop re-posting the exact same thing over, and over, and over
again.
Then, while you are sitting thinking about this, wondering why you
haven't yet got a response, how about you work out how to switch off
HTML emails. Send it in plain text, more people will bother reading it,
and
On Thu, 6 Jan 2005 23:50:40 -0500, David Ishmael wrote:
What about when users switch to 100% VoIP? I've been considering getting
DirecTV with the HD PVR and I've heard it can't use broadband, in a case
like that I would have to route a modem call through VoIP (or is there a
better way I'm just
On Fri, 7 Jan 2005 10:36:50 +, John Middleton wrote:
I've an issue with my TDM4000P card and I will be calling Digium later
to ask for their help.
Could anyone help me with a basic configuration so they can SSH to me?
On your router you'll need to port forward port 22 to your Asterisk
hello,
I've tried do it, but nothing happened.
Regards
Csar
- Original Message -
From: Adam Goryachev [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, January 07, 2005 9:20 AM
Subject: Re: [Asterisk-Users]
On January 7, 2005 07:20 am, Adam Goryachev wrote:
While I agree with you completely with your comments on HTML posting and
repeating the exact same information over and over, your advice on
configuration is dead wrong.
zaptel.conf
fxoks=1-2
fxsks=3-4
zapata.conf
[channels]
Yep check out the new generation of set top boxes - all ip based.
eg www.akimbo.com just launched at CES yesterday, both Ethernet cat 5
and wireless connections.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Graves
Sent: Friday,
2. NetGear Switch - I'm using FS-526T Switch, which
has 24 10/100 ports and 2 Gb sorts.
I want to know if this kind of general purpose switch
is not suitable for voip. If so, could you recommand
one?
I've been doing network performance assessments for corporate clients
in 40+ states since
Scott Stingel wrote:
Sid-
Try connecting one port to another. Note that one of the ports must
be set up as cpe and the other as net in zapata.conf when you loop
them together like this.
A suitable crossover cable for testing can be constructed by cutting
up a CAT 5 cable, and connecting:
Pin
On Thu, Jan 06, 2005 at 11:50:40PM -0500, David Ishmael said:
What about when users switch to 100% VoIP? I've been considering getting
DirecTV with the HD PVR and I've heard it can't use broadband, in a case
like that I would have to route a modem call through VoIP (or is there a
better way
Hello People,
I am a newbie asterisk and happy user, i have configured a x100p card and
everything works nice, i can forward incoming connections to a x-lite
software client and works out of the box,
However when i try to make a connection between two x-lite clients then no
audio is transmited,
Chris wrote:
Hey all,
Is there any software or something out there that anyone knows of that
will allow me to have a conference in asterisk (or possibly not if you
know another solution) where I can see who is talking at the time? Kinda
like teamspeak or ventrillo. I'm not getting my hopes up, but
Anyone help me, I've looked at the Wiki and cant see anything
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To UNSUBSCRIBE or update options visit:
I'm trying to install spandsp
But when I try to patch the Makefile it gives this error
[EMAIL PROTECTED] apps]# patch apps_makefile.patch
patching file Makefile
Reversed (or previously applied) patch detected! Assume -R? [n] y
Hunk #1 succeeded at 41 (offset -6 lines).
Hunk #2 FAILED at 67.
is
To get the current stable release, issue the following command:
# cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons
asterisk-sounds
http://www.asterisk.org/index.php?menu=download
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
On Fri, 2005-01-07 at 16:07 +0200, Altus Snyman wrote:
I'm trying to install spandsp
But when I try to patch the Makefile it gives this error
[EMAIL PROTECTED] apps]# patch apps_makefile.patch
patching file Makefile
Reversed (or previously applied) patch detected! Assume -R? [n] y
Hunk #1
Current setup:
Polycom IP3000 - gnugk - asterisk - Cisco 7940
Asterisk and gnugk are on 10.20.98.6
IP3000 is H.323, using G.711 (10.20.98.2)
7940 is SIP, using g711ulaw (10.20.98.3)
I've been asterisk for a while now, only using SIP devices. I'm happy
with that side of things, but I've not
Hello everybody,
does anybody knows from where I can get an list of international area
codes incl. the mobile numbers?
Regards
Bastian
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Hi,
I have some trouble with the Monitor() application. I start and stop it via
the management interface, giving no special parameters except the channel
name. What happens is:
- if I specify WAV as the format, the resulting files are exactly 44 bytes big
and contain nothing at all
- if I
I'm trying to install spandsp
But when I try to patch the Makefile it gives this error
[EMAIL PROTECTED] apps]# patch apps_makefile.patch
patching file Makefile
Reversed (or previously applied) patch detected! Assume -R? [n] y
Hunk #1 succeeded at 41 (offset -6 lines).
Hunk #2 FAILED at
Joe Dennick wrote:
Yeah, set the queue timeout to be about 1 second less than the voicemail
timeout (ya know, where you say Dial(SIP/, 15)). That way the queue
times out the agent before the dialplan goes to voicemail.
The more reasonable solution is to just put the agent's direct
Title: Message
I know that my
phone displays *8 because I dailed that.
But it's
definitly not what I would want, or most other people.
Any other
ordinary PBX would show the CID of the caller, but because this is a SIP-based
system we get this problem.
I was thinking
more in line of an
Yes.
Quoting Roger Hanson [EMAIL PROTECTED]:
Is the meeting still on for Saturday 1/8/05?
11:30am at 2375 University Av W STE120, Saint Paul.
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Hi,
can someone tell be about some good and free softphones?
Are they easy to use by non-tecnical users?
Can someone share their experience about the implementation of VoIP
softphones in a company? because usualy people dont want to make changes
in the way they work I would like to know a
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=enq=international+dialing+codes
___
Asterisk-Users mailing list
Matthew Boehm wrote:
If I add a line like this: member = SIP/3044, can I still get
login/logoff functionality? We need agent login/logff functionality AND for
calls to not goto voicemail.
No, I was suggesting using SIP/3044 in agents.conf, not in queues.conf.
If you put it into queues.conf,
/SNIP/
On Thu, 2005-01-06 at 12:00 -0600, asterisk-users-
[EMAIL PROTECTED] wrote:
Andy Burns wrote:
Shoval Tomer wrote:
Can anyone comment why shouldn't we use FC 3 for an * production
system?
when I tried the X100P drivers on FC3 I had problems with udev, the
workaround didn't work
Robert Webb Posted:
-Original Message-
Sent: Thursday, January 06, 2005 3:53 PM
Subject: [Asterisk-Users] Test2
Sorry for all the tests. Please excuse.
/SNIP/
What are you trying to test? The list's patience?
Seshu
NOTICE:
Hi,
This morning I had some failed calls. On the console (and in the log)
I saw the error Unable to allocate channel structure. Before I restarted
the process, I checked it's memory usage in ps and glanced at my free
memory in top. Asterisk was using a normal ammount of memory, about
40M. I
On Fri, 2005-01-07 at 08:08 -0700, Kevin P. Fleming wrote:
Matthew Boehm wrote:
If I add a line like this: member = SIP/3044, can I still get
login/logoff functionality? We need agent login/logff functionality AND for
calls to not goto voicemail.
No, I was suggesting using SIP/3044
Joao,
Thanks for sending the Installation tips as pasted below.
It works.
Seshu
--
Get oh323 fromhttp://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gzGet
pwlib
I can send a list, mobile is not complete but it has a lot of numbers...
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de PHP Mechanic
Enviado el: Viernes, 07 de Enero de 2005 11:57 a.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto:
That's what I'm about to try, I keep getting pulled off of this project
to go do other things. Thanks for the input.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrei
(MPI)
Sent: Thursday, January 06, 2005 5:35 PM
To: Asterisk Users Mailing
-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 the
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
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: asterisk-users@lists.digium.com
Sent: Friday, January 07, 2005 9:35 AM
Subject: [Asterisk-Users]
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
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
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
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
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 sip:[EMAIL
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 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 normal
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 know
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
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
What version of sox do you use?
Lamine
- Original Message -
From: Robert Spielmann [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
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
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.
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.
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
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
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
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
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.
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 be 1.
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
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
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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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
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 SIPMAC.cnf file. If that works, then your .cnf
Hello,
Is there any way to unlock the Linksys
router?
--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320 ext
2010
Blank Bkgrd.gif___
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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,
post your /etc/odbc.ini and /etc/odbcinst.ini
-matthew
- Original Message -
From: rizwan [EMAIL PROTECTED]
To: Asterisk-users@lists.digium.com
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
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
___
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
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
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
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
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
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
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
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:
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: Asterisk-users@lists.digium.com
Sent: Friday, January 07, 2005 10:19 AM
Subject:
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 scanned for
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.
-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
streamload.com
dropload.com
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
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
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
You can Dial() extension SIP/line1SIP/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/line1SIP/line2,10)
etc...
that's if I understood what ou needed...
bye,
M.
- Original Message -
From:
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
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
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? libtiff?
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
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
asterisk-users@lists.digium.com
Sent:
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
You can Dial() extension SIP/line1SIP/line2
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
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