Hi,
I just received a DTA310 Terminal Adapter from Packet8. The Advanced
Configuration is password protected. Does anyone know the default
password or algorithm necessary to get into it?
Thank you,
Barton
___
Asterisk-Users mailing list
[EMAIL PROTE
Bruce Hedreen wrote:
Has anyone succesfully integrated * with a cisco voice gateway ?
Works well with AS5350 and ATA186.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
On Mon, 15 Dec 2003 23:05:56 -0500 (EST), [EMAIL PROTECTED]
wrote:
>Hello All
>When I open up iaxcomm, it registers fine with the asterisk server. If I
>call into it, iaxcomm will ring; however if I leave iaxcomm sitting idle
>for awhile (I haven't figured out exactly how long) it seems to miss
>
Hello All
When I open up iaxcomm, it registers fine with the asterisk server. If I
call into it, iaxcomm will ring; however if I leave iaxcomm sitting idle
for awhile (I haven't figured out exactly how long) it seems to miss
calls. I can see the calls coming in on the asterisk server but they
nev
A new minor release of pyst is available at
http://sourceforge.net/projects/pyst/
Completed is the manager interface allowing access to all manager
commands. The manager interface is event driven and designed to be
useful for driving gui applications.
Also included is a small bug fix to agilib f
> 3. Supposed I have 2 fxo cards (right now I have one already) and 3
> fxs, and one of the fxo will have two phone (running pararell), is
> there any way for * to:
> a. It always dial the first fxo, if the fxo is busy or is being used
> (have other people conversation), will * be able to switch it
On Mon, 2003-12-15 at 13:14, Cees de Groot wrote:
> Steven Critchfield <[EMAIL PROTECTED]> said:
> >You are just as capable of dialing the parked call number as the other
> >person.
> >
> I don't completely understand that remark, but it seems to imply "don't
> whine, you can emulate feature X by
On Monday 15 December 2003 20:15, Isianto Istiadi wrote:
> Dear all,
>
> I have some questions, I'm sure it's pretty stupid for most of you,
> but I need you guys to help me. Here are my questions:
> 1. Music On Hold, it doesn't play any sound on the parked call or
> hold call. But if I do ps-ax, i
- Original Message -
From: "Isianto Istiadi" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, December 15, 2003 8:15 PM
Subject: [Asterisk-Users] Beginner couple of questions
> 1. Music On Hold, it doesn't play any sound on the parked call or hold
call.
> But if I do ps-ax, it sh
On Mon, 2003-12-15 at 18:07, Klaus-Peter Junghanns wrote:
> Hi isdn folks,
>
> i just released some (still) experimental drivers for Zaptel
> BRI isdn support. Currently it includes kernel modules for our
> quadBRI PCI ISDN card. A driver for the HFC-S PCI A chipset based
> el-cheapo isdn cards wi
On Mon, 2003-12-15 at 16:13, Florian Overkamp wrote:
> Hi,
>
> First off, let me state that _YES, I am fully aware that what I am doing is
> insane, prone to major havoc and bad for general health_ :-))
>
> Scenario: My GF needs an analog modem to use with her banking software
> (sodding backs
Dear all,
I have some questions, I'm sure it's pretty stupid for most of you, but I need
you guys to help me. Here are my questions:
1. Music On Hold, it doesn't play any sound on the parked call or hold call.
But if I do ps-ax, it shows mpg123 .( I forgot the exact line). I'm using
slackwa
On Tue, Dec 16, 2003 at 11:28:38AM +1100, Gonzalo Servat wrote:
> On Tue, 2003-12-16 at 10:34, Michiel Betel wrote:
> > Is case anyone wants to know... The Fritz! USB ISDN box works fine with
> > Asterisk!
> > I'm running CAPI 0.3.0 and love it, because the mini ITX server I have
> > only takes o
On Tue, 2003-12-16 at 10:34, Michiel Betel wrote:
> Is case anyone wants to know... The Fritz! USB ISDN box works fine with
> Asterisk!
> I'm running CAPI 0.3.0 and love it, because the mini ITX server I have
> only takes one PCI slot which is now filled with a 4 port Digium card.
Hi Michiel,
T
> [2203]
> type=friend
> username=2203
> secret=2203
> host=dynamic
> defaultip=192.168.0.2
> dtmfmode=inband
> canreinvite=yes
Add here:
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=inband only works with ulaw (g.711), so better use a different
setting here.
> the console on * when r
Hi!
> I'd like to connect phpgroupware to asterisk: when a user click on a phone
> number, his phone rings and he gets connected to the number he just clicked.
>
> I've tried by putting various files in /var/spool/asterisk/outgoing, without
> results (we are using SIP phones + CAPI channels).
Th
On Monday, December 15, 2003 9:51 AM, Steven Critchfield
[SMTP:[EMAIL PROTECTED] wrote:
> On Mon, 2003-12-15 at 07:18, Michiel Betel wrote:
> > What if the internal party is busy and answers with voicemail
> > how then do I get my original call back to me? Pressing flash will
> > conf in the voic
Is case anyone wants to know... The Fritz! USB ISDN box works fine with
Asterisk!
I'm running CAPI 0.3.0 and love it, because the mini ITX server I have
only takes one PCI slot which is now filled with a 4 port Digium card.
___
Asterisk-Users mailing l
good to hear theres going to be support for this phones, but why not put
it on the wiki??? so we can have all the faq in one place.
Miguel
On Mon, 2003-12-15 at 22:23, mattf wrote:
> I just got off of the phone with Scott Willard at Polycom and things seem
> promising. He's going to send me the la
Sorry if this question has been asked before. I am very new to VoIP. I have
a Cisco ATA 186. Do I need a Sip Server w/ Asterisk to make this thing work
or does Asterisk take care of everything? I hope someone can point me in the
right direction.
Thanks,
___
Hi isdn folks,
i just released some (still) experimental drivers for Zaptel
BRI isdn support. Currently it includes kernel modules for our
quadBRI PCI ISDN card. A driver for the HFC-S PCI A chipset based
el-cheapo isdn cards will follow soon.
We now also have the quadBRI cards in stock.
Find the
Hi to all,
today I have released the version 0.3 of ast-ax-snmpd.
It is some code that adds the snmp subagent functionality to asterisk,
using the ucd snmp framework and following the agentX standard.
It has tested under debian woody, with specific version of ucd_snmp,
so your mileage may vary.
I a
Hi to all,
today I have released the version 0.3 of ast-ax-snmpd.
It is some code that adds the snmp subagent functionality to asterisk,
using the ucd snmp framework and following the agentX standard.
It has tested under debian woody, with specific version of ucd_snmp,
so your mileage may vary.
I a
On Monday 15 December 2003 15:16, Dawid Mielnik wrote:
> I can not send voicemails as an attachement. When setting the
> "attach=yes" option in voicemail.conf the mails get rejected from
> the mail server:
This is not a problem with Asterisk. Have a talk with your mail admin
about this destinatio
On Monday 15 December 2003 15:33, Jon Creasey wrote:
> Hi all,
>
> New user to asterisk having just got it compiled and installed.
>
> Running with no digium hardware (yet) and no soundcard in asterisk
> box.
>
> Problem is using the sample configs with a sip phone added as
> follows
>
> [2203]
> t
On Mon, 15 Dec 2003, Florian Overkamp wrote:
> At the time of writing it served my purpose, but it had some problems with
> asterisk becoming unresponsive after a large number of manager logins/logouts.
> I suppose this has been solved by now, but I stopped testing it heavily since
> I had no
- Original Message -
From: "Dawid Mielnik" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, December 15, 2003 4:16 PM
Subject: [Asterisk-Users] voicemail as an attachement
>
> Hi,
>
> I can not send voicemails as an attachement. When setting the "attach=yes"
> option in voicemail
I just got off of the phone with Scott Willard at Polycom and things seem
promising. He's going to send me the latest stable boot firmware for the
Soundpoint 500/600 phones and I will make that available on a webiste
somewhere for people to download.
It seems like their big stumbling block to giv
Hi,
First off, let me state that _YES, I am fully aware that what I am doing is
insane, prone to major havoc and bad for general health_ :-))
Scenario: My GF needs an analog modem to use with her banking software
(sodding backs don't supply a decent web-application for company use). I am
exper
Hi,
Citeren John Todd <[EMAIL PROTECTED]>:
> astping.tar (http://www.dynx.net/ASTERISK/misc-progs/ and also in the
> mailing list archives) supposedly sends a query to an Asterisk
> server, but I have been unable to get it to do anything other than
> reply with the IP address of the queried ho
I've setup a simple asterisk test environment with an ISDN card
configure in modem.conf and a gnophone client connected to my asterisk
server via IAX.
I can place call and answer, i've also succesfully configured a
voicemail.
The problem is that i cannot redirect call to my voicemail when gnophone
On Mon, 15 Dec 2003, John Todd wrote:
>I have spent some time digging through the archives for comments
> concerning Asterisk and monitoring systems, and I have found few
> results.
>Anyone? (yes, yes, I should do it myself, but why do something that
> someone has already done?)
andre
On Mon, 2003-12-15 at 14:02, Ludovic Drolez wrote:
> Hi !
>
> I'd like to connect phpgroupware to asterisk: when a user click on a phone
> number, his phone rings and he gets connected to the number he just clicked.
>
> I've tried by putting various files in /var/spool/asterisk/outgoing, without
Tilghman Lesher <[EMAIL PROTECTED]> said:
>Please learn to respond at the bottom and trim footers like everybody
>else. (Reformatted to comply with standards.)
>
And while you're at it, Tilghman, trim some of the body as well...
>On Monday 15 December 2003 13:19, Sri wrote:
[28 lines of irreleva
Hi all,
New user to asterisk having just got it compiled and installed.
Running with no digium hardware (yet) and no soundcard in asterisk box.
Problem is using the sample configs with a sip phone added as follows
[2203]
type=friend
username=2203
secret=2203
host=dynamic
defaultip=192.168.0.2
d
> Message: 1
> From: Areski <[EMAIL PROTECTED]>
> To: Asterisk-Users Mailing-list <[EMAIL PROTECTED]>
> Organization:
> Date: 15 Dec 2003 13:06:18 +0100
> Subject: [Asterisk-Users] Howto to test asterisk applications - VoIP
Testing Solution
> Reply-To: [EMAIL PROTECTED]
>
> Hello All,
>
> Anybody c
I am trying to steer my company away from purchasing an expensive and
proprietary 3com NBX100 system. Currently, we are using a norstar
system in conjunction with centrex lines that allows us to transfer a
call to an outside number and, in transferring, release it from our pbx
(the transfer is don
I have spent some time digging through the archives for comments
concerning Asterisk and monitoring systems, and I have found few
results.
check_asterisk.pl.gz (http://www.dynx.net/ASTERISK/misc-progs/) which
gives an error on download, and has no further Google references
astping.tar (http:
Hi,
I can not send voicemails as an attachement. When setting the "attach=yes"
option in voicemail.conf the mails get rejected from the mail server:
- Transcript of session follows -
451 4.4.1 timeout writing message to higgs.elka.pw.edu.pl.: Connection timed
out
with higgs.elka.pw.e
mattf wrote:
Hello,
just use the manager conduit:
Telnet to asterisk server ip_address to port 5038 (as long as you have a
login set up in manager.conf)
and send the following(what is between the dash lines):
---
Action: Login
Username: user
Secret: pass
Action:
Hello,
I've talked with various people at Polycom about getting on their developer
program and they have been stalling me for weeks. The most information I
received was from Scott Willard, head of IP phone sales for Polycom. He said
that the company is not sure of what direction they want to head
Using a Q is nice because you can have another employee or group of
employees cover while the receptionist is on break.
TL
Olle E. Johansson wrote:
Tilghman Lesher wrote:
On Monday 15 December 2003 10:57, Sri wrote:
Hi All
This is one scenario I would like to have some help. I have
searched
Could you post the console output from when you run the softphone application? Maybe
there is a problem with registration.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Hao Zhong
Sent: Friday, December 12, 2003 5:16 PM
To: [EMAIL PROTECTED]
Subject:
Shoval Tomer wrote :
>Hi, could anyone please provide a working sample of how to
>configure asterisk to connect to fwd?
>I've tried the one at www.loligo.com and it doesn't work.
>Not even when calling to 5.
I presume you're looking at the asterisk console (ie. started asterisk with
option -v
If I call to voicemail with G.711, I'm getting the unavailable message I've recorded
in voicemail.
If I call to voicemail with GSM, I'm not getting my message, only thevoice's standard
message
that always says user zero-zero-zero-zero regardless of extension.
Do the user need to record a message
Hello,
just use the manager conduit:
Telnet to asterisk server ip_address to port 5038 (as long as you have a
login set up in manager.conf)
and send the following(what is between the dash lines):
---
Action: Login
Username: user
Secret: pass
Action: Originate
Cha
Please learn to respond at the bottom and trim footers like everybody
else. (Reformatted to comply with standards.)
On Monday 15 December 2003 13:19, Sri wrote:
> Tilghman Lesher wrote:
> >On Monday 15 December 2003 10:57, Sri wrote:
> >>Hi All
> >>This is one scenario I would like to have some h
Sri wrote:
Thanks Tilghman.
Is there a document that lists the functions that you have used in the
script?
If there is no such document, whats a good starting point to get a list
of them.
http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+autoattendant
/Olle ;-)
_
Has anyone on the list been able to locate and try out the 1.1.0 firmware? It was
released in November, but I have yet to get my hands on it. The Polycom site has way
more docs online, but the link to the firmware only brings up the release notes.
-sb
__
Hello
I have configured IP 7905G and SNOM 200 for Asterisk. Now problem is that I can
call from IP 7905G to SNOM 200 but not the other way round. Instead I get "FORBIDDEN"
Message on SNOM 200 LCD when ever I try to call IP7905 phone and asterisk generate
following messages..
Please note 81
Hi !
I'd like to connect phpgroupware to asterisk: when a user click on a phone
number, his phone rings and he gets connected to the number he just clicked.
I've tried by putting various files in /var/spool/asterisk/outgoing, without
results (we are using SIP phones + CAPI channels).
Is there a
Tilghman Lesher wrote:
On Monday 15 December 2003 10:57, Sri wrote:
Hi All
This is one scenario I would like to have some help. I have
searched the digium lists and could not find any posts on this.
How can an Attendant switch on or off the AutoAttendant from her
phone? Eg.
8am -> Attendent ente
Hmm, does not sound like progress...
Thanks anyway. We'll try to fix this as well.
CS
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] Im Auftrag von listas iPfone
> Gesendet: Montag, 15. Dezember 2003 18:52
> An: [EMAIL PROTECTED]
> Betref
On Mon, Dec 15, 2003 at 09:25:05AM -0500, John Todd wrote:
> Paul -
> Yes, your description is correct.
>
> - moving the phone (no ethernet passthrough) results in no symptoms
You might have a virus on that XP box that totally saturates the poor
7960 switch with bogus IP packets.
--
May the L
Thanks Tilghman.
Is there a document that lists the functions that you have used in the script?
If there is no such document, whats a good starting point to get a list of
them.
Tilghman Lesher wrote:
On Monday 15 December 2003 10:57, Sri wrote:
Hi All
This is one scenario I would
Steven Critchfield <[EMAIL PROTECTED]> said:
>You are just as capable of dialing the parked call number as the other
>person.
>
I don't completely understand that remark, but it seems to imply "don't
whine, you can emulate feature X by telling your users to use feature
Y".
Which is a completely
I've already got Asterisk using the new recv only method too.
Mark
On Mon, 15 Dec 2003, Christian Stredicke wrote:
> Hi folks,
>
> in order to establish backward compatibility we made an image that
> automatically detects if the other side does not support RFC3264. Please try
> it out, we would
On Monday, December 15, 2003 10:10 AM, David Gomillion
[SMTP:[EMAIL PROTECTED] wrote:
> The Asterisk system will be used to replace a Norstar MICS. The
> location has two PRI's coming in, with a few hundred DIDs. I know
> how
> to make * use the DIDs incoming, and I know how Nortel uses the DIDs
I dont think this works in the scenario i explained. Basically
tilghman's solution sounds neat!
(Wish i had some knowledge of these global varilables and dbget and
dbput functions earlier...Thanks all)
Todd Lieberman wrote:
Put the calls to a Q and have your reciptionist login/logout of the
Qu
Try:
http://callflow.sourceforge.net/
Jorge
iTS [EMAIL PROTECTED] wrote:
nope h323 only sorry...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Areski
Sent: Lunes, 15 de Diciembre de 2003 04:50 p.m.
To: Asterisk-Users Mailing-list
Subject: RE: [Asterisk-
Steven Critchfield wrote:
On Mon, 2003-12-15 at 10:27, Cees de Groot wrote:
Steven Critchfield <[EMAIL PROTECTED]> said:
Your not missing anything essential, and have described a limitation in
analog signaling.
Are you sure? We're setup with DIAX here (and as far as I know '#' with
I have a 4 port card in a regular system and I get that prob sometimes when I copy
large files to that server IRQ problem? if I stop the copy the sound prob goes
away. not a big help but at least you know that your not alone...
Dave..
>>> [EMAIL PROTECTED] 12/15/2003 10:17:35 AM >>>
On M
On Mon, Dec 15, 2003 at 10:05:56AM +0200, Peter Zeltins wrote:
> My Asterisk box also does NAT for internal network, and
> establishes site-to-site VPN tunnel(s). As a result I have
> several internal interfaces with private addresses on them, and
> only one public interface. By trial-and-error I'v
On Mon, 2003-12-15 at 10:57, Sri wrote:
> Hi All
> This is one scenario I would like to have some help. I have searched
> the digium lists and could not find any posts on this.
>
> How can an Attendant switch on or off the AutoAttendant from her phone?
> Eg.
> 8am -> Attendent enters office ->
Hi!
> How can an Attendant switch on or off the AutoAttendant from her phone?
This is not difficult and can be done within the dialplan as defined in
extensions.conf. Take a look at DBput() and DBget() in combination with
gotoif(). Store the system data using DBput(), and get the system status
Stephen R. Besch a écrit:
John Breeden wrote:
Am I assuming that a GS set to early dial to * dosn't work. Or am I
missing something? Tried inband, info and rfc288, all nojoy. I'm
assuming that it's not/supported or GS bug, only asking because it's
assumptions that alwas get me :-)
GS firmw
Hi!
i just tryied the 2.03b firmware.
Now i have that message when the phone boots:
Challenge User: <6466212364662
64662
pressing ok the display shows>> PW: iputmypassword
When i put my password i get a loop returning for Challenge User:
<6466212364662 again
64662 is my FWD number
Now the p
Hi!
> I am trying to change the email body and the from string sent in the
> voicemail notifocation mail.
>
> I have changed the entries in the voicemail.conf but I still receive the
> standard email template from "Asterisk PBX" (instead of my from) and [PBX]:
> in the subject.
I know *that* pro
Jeremy,
So the only way is to change the voicemailcode and recompile ?
the flexible - uncomment these lines does will never work ?
thnx.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara
Sent: Monday, December 15, 2003 5:33 PM
To: [EMAIL
Just a hint:
You could adapt your dialplan to offer two extension entries to dial the
same person, but only one of these extensions will fire up voicemail on
busy or unavailable. Use the other one for transfers. Not perfect, but a
partial solution.
Cheers, Philipp
___
I'm getting this error, and couldn't find any info on what is going on.
Thanks for any help.
NOTICE[1142135600]: File chan_iax2.c, Line 4910 (iax2_poke_peer): Still have a
callno...
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.c
Steven Critchfield a écrit:
On Mon, 2003-12-15 at 09:58, Daniel ANDRE wrote:
Hello,
I would like to have some comparison between E1 cards from Digium and
those from Eicon for a VOIP - ISDN Gateway.
How does they compare on the echo cancel point of view?
Is the echocancellation
Put the calls to a Q and have your reciptionist login/logout of the Queue.
xten => 600,1,AddQueueMember(phillyq|SIP/${CALLERIDNUM:6})
exten => 600,2,Playback(agent-loginok)
exten => 600,3,Hangup
exten => 601,1,RemoveQueueMember(phillyq|SIP/${CALLERIDNUM:6})
exten => 601,2,Playback(agent-loggedoff
Check out:
http://lists.digium.com/pipermail/asterisk-users/2002-December/006728.html
David Gomillion wrote:
I am currently working on an Asterisk test system, and will be
presenting a demo to the Board of Directors tomorrow night. I want to
make sure I have all of my ducks in a row.
The Aster
Hi,
the Eicons work fine with chan_capi, also the hardware echo
cancelation works fine.
regards
kapejod
Am Mo, 2003-12-15 um 17.38 schrieb Steven Critchfield:
> On Mon, 2003-12-15 at 09:58, Daniel ANDRE wrote:
> > Hello,
> >
> > I would like to have some comparison between E1 cards from Digium
On Monday 15 December 2003 10:57, Sri wrote:
> Hi All
> This is one scenario I would like to have some help. I have
> searched the digium lists and could not find any posts on this.
>
> How can an Attendant switch on or off the AutoAttendant from her
> phone? Eg.
> 8am -> Attendent enters office
John Breeden wrote:
Am I assuming that a GS set to early dial to * dosn't work. Or am I
missing something? Tried inband, info and rfc288, all nojoy. I'm
assuming that it's not/supported or GS bug, only asking because it's
assumptions that alwas get me :-)
GS firmware 1.0.4.26
Thanx in advan
On Mon, 2003-12-15 at 10:27, Cees de Groot wrote:
> Steven Critchfield <[EMAIL PROTECTED]> said:
> >Your not missing anything essential, and have described a limitation in
> >analog signaling.
> >
> Are you sure? We're setup with DIAX here (and as far as I know '#' with
> IAX does the same as flas
Hey Srs.
I have a little problem with the next scenario:
Internal Phone(801)<-->Asterisk(public IP) <--INTERNET-->ADSL
Router<-->Budgetone(716)
|-->ADSL Router<-->Budgetone(717)
My sip.conf is the next:
[general]
port = 5060 ; Port to bind to
bindaddr
Hi All
This is one scenario I would like to have some help. I have searched
the digium lists and could not find any posts on this.
How can an Attendant switch on or off the AutoAttendant from her phone?
Eg.
8am -> Attendent enters office -> switches OFF auto attendent. He/She
takes in all the
On Mon, 2003-12-15 at 10:26, [EMAIL PROTECTED] wrote:
> Hi all,
>
> Also got a problem with the RxFax app, I'm using the following packages,
> spandsp-20031021
> tiff-v3.6.0
> Asterisk CVS-12/10/03-20:28:08
>
> Using the Digium TE410P card on a E1/PRI line.
>
> The tiff files under /var/spool/as
Steve Rodgers wrote:
Sip phones generate their own dialtone. The ignore pat option is meaningless
with regard to SIP phones. I would check the Qrandstream's dialplan and see if
you can program it to ignore the dialtone after a '9' is pressed. I had to do
something similar for my Sipura SPA-2000
On Mon, 2003-12-15 at 10:41, Thomas Haeger wrote:
> Hi all,
>
> I have installed one TDM20B on an ITX board in a small "cube" chassis.
> When the harddisk is working (when installing something or make a query on a
> database) i can hear nasty noises (like hdd-head is moving) on the connected
> pho
Steven Critchfield <[EMAIL PROTECTED]> said:
>Your not missing anything essential, and have described a limitation in
>analog signaling.
>
Are you sure? We're setup with DIAX here (and as far as I know '#' with
IAX does the same as flash with an analog port), and once you transfer
someone, you've
Hi all,
I have an * installation with one AVM Fritz card, one
TDM 40b and one sip phone. The problem i have is that
when i place calls from the sip phone everything works
fine. When i receive calls to the sip phone i cannot
hear and the calling party cannot hear too.
Is it a codec problem ?
Here
nope h323 only sorry...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Areski
Sent: Lunes, 15 de Diciembre de 2003 04:50 p.m.
To: Asterisk-Users Mailing-list
Subject: RE: [Asterisk-Users] Howto to test asterisk applications -
VoIPTesting Solution
CALLGEN :
Hi all,
I have installed one TDM20B on an ITX board in a small "cube" chassis.
When the harddisk is working (when installing something or make a query on a
database) i can hear nasty noises (like hdd-head is moving) on the connected
phone.
Have someone experiences with this manner ?
Thanks for h
On Mon, 2003-12-15 at 09:58, Daniel ANDRE wrote:
> Hello,
>
> I would like to have some comparison between E1 cards from Digium and
> those from Eicon for a VOIP - ISDN Gateway.
>
> How does they compare on the echo cancel point of view?
> Is the echocancellation code for E400 good enough for pr
Dawid Mielnik wrote:
Hello,
I am trying to change the email body and the from string sent in the
voicemail notifocation mail.
I have changed the entries in the voicemail.conf but I still receive the
standard email template from "Asterisk PBX" (instead of my from) and [PBX]:
in the subject. Can an
Hi all,
Also got a problem with the RxFax app, I'm using the following packages,
spandsp-20031021
tiff-v3.6.0
Asterisk CVS-12/10/03-20:28:08
Using the Digium TE410P card on a E1/PRI line.
The tiff files under /var/spool/asterisk/incoming/, are a 8-byte file,
and a 314-byte file.
Because it could
Walker Haddock wrote:
On Mon, Dec 15, 2003 at 05:06:57PM +0200, Dan wrote:
Hi,
- Original Message -
From: "Walker Haddock" <[EMAIL PROTECTED]>
Dan, you say fax works better on the TDM400 than the ATA186. I'm having
problems with faxing on the TDM400.
I had to drop t
Hi folks,
in order to establish backward compatibility we made an image that
automatically detects if the other side does not support RFC3264. Please try
it out, we would be very interested if this image is a progress!
http://snom.com/download/share/snom200-2.03b-SIP.bin
Thanks, CS
___
Hello,
I am trying to change the email body and the from string sent in the
voicemail notifocation mail.
I have changed the entries in the voicemail.conf but I still receive the
standard email template from "Asterisk PBX" (instead of my from) and [PBX]:
in the subject. Can anyone help me out in c
I am currently working on an Asterisk test system, and will be
presenting a demo to the Board of Directors tomorrow night. I want to
make sure I have all of my ducks in a row.
The Asterisk system will be used to replace a Norstar MICS. The
location has two PRI's coming in, with a few hundred DID
Hello,
I would like to have some comparison between E1 cards from Digium and
those from Eicon for a VOIP - ISDN Gateway.
How does they compare on the echo cancel point of view?
Is the echocancellation code for E400 good enough for production
environment?
Best regards,
Daniel
--
Daniel ANDRE
Read the source luke
Just found out the answer to my own question... press FLASH once more...
Now the big question is, which part of the source to comment out
to stop the 3 way conference... so you get a normal consult and flash
will get back the caller. It's very confusing for my users.
Mi
On Mon, Dec 15, 2003 at 05:06:57PM +0200, Dan wrote:
> Hi,
>
> >- Original Message -
> >From: "Walker Haddock" <[EMAIL PROTECTED]>
> >
> > Dan, you say fax works better on the TDM400 than the ATA186. I'm having
> problems with faxing on the TDM400.
> > I had to drop the rx and tx rates o
On Mon, 2003-12-15 at 07:18, Michiel Betel wrote:
> Hi,
>
> We are using threewaycalling & flash transfers over a CAC channelbank.
>
> The following happens:
>
> Call comes in to my extension
> I talk to a party and press flash
> party goes on hold, I get get dail tone
> I dial internal number
>
CALLGEN : An ATM Signaling Performance Evaluation Tool !
Sorry I never hear about this tools before, is it working with SIP call?
Is it possible to analyze the callflow with callgen and to send DTMF ?
Cheers,
Areski
On Mon, 2003-12-15 at 09:45, iTS [EMAIL PROTECTED] wrote:
> Try callgen also an
Hi,
I'm currently using SER, and the voicemail system there is not stable,
and is lacking IVR.
I'm wondering if I could use asterisk as a voicemail system only,
where calls will get redirected by ser to asterisk and users will be
able to leave a message.
Is this setup workable, and have anybody
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