Hi all,
I'm using a voicetronix openswitch6 card with asterisk. When I try to
dial the vpb phone from my application, I get t he following error:
-- Executing Dial("Zap/1-1",
"vpb/1-9|10|mtT||/usr/local/sbin/parlix_dial_event 2 196") in new stack
-- 1-9 requested, got: [None]
NOTICE[524311]:
Hi Steven,
It looks to me as if you haven't defined 'channel = 9' in your vpb.conf
file ... yet the log output shows that you are attempting to use that
undefined channel.
Try adding it to your vpb.conf first.
Terence
On 10 Feb 04, at 2:24 AM, Steven Kawuma wrote:
Hi all,
I'm using a voic
Hi,
I am trying to use the voicemail feature
of * for a Cisco call manager express setup with 10 7960 phones. * is unable to
recognise the DTMF when mailbox is accessed by voicemail.
Here are my configs in cisco
dial-peer voice 8500
voip destination-pattern 8500 session protocol
sipv2
Hi,
My vpb.conf now reads:
[interfaces]
echocancel = on
board = 1
context = parlix_agents
; Note that V6PCI channel numbers start at 7!
mode = fxo
channel = 7
channel = 8
mode = dialtone
channel = 9
channel = 10
channel = 11
channel = 12
But I still get the same error.
Just in case, `lsmod
Hi,
> -Original Message-
> Could you share your 3.0.0 config?
http://clio.obsimref.com/dev.htm
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I think you'll have to be more specific in posting *exactly* what problems
occur.
Although I haven't got OS X I've been running asterisk on an old Mac for
longer than I can remember - so there aren't that many issues once the
software is built. Post the logs of the build process and one of the
Hi,
Just as we have a discussion about problems with cisco ATA186 devices I
experience this following:
Since cvs update last Sunday my ATA186's using 2.16 firmware with MGCP there
does not seem to be any RTP happening. Rebooting the ATA's does not work.
Strange thing is: I have a SwissVoice IP10
Hi,
> -Original Message-
> Brian> On a broader note, I would love to try to play with the
> Brian> very-low-bandwidth versions of Speex. I could have sworn I saw
> Brian> things on the bugtracker some weeks back on that topic, but I
> Brian> can't find them anymore.
>
> It is bug number:
Soragan wrote:
Ohh.. You better give that to me then. I'll send you my Pentium 133 w/ 16
megs of ram. It works great with the X100P.
LOL, can your Pentium do web server, mail server with spam and virus
checking and ADSL router all together? If it can do without any performance
loses compare w
Hi everyone, I have
a question regarding recording CDR while using overlap digit
dialing.
When you dial from
one PRI to Asterisk and Dial() to another PRI, and when you are using overlap,
the records in CDR don't contain whole dialed numbers. It seams that CDR record
is not constructed af
Florian Overkamp wrote:
I am using X-Lite on some setups. Speex from X-Lite does not seem to work
with asterisk - I just get no sound at all. Disabling Speex and favouring
GSM or G711 works fine.
Need to apply a .reg file to the PC running X-Lite:
http://bugs.digium.com/bug_view_page.php?bug_id=00
Billy Huddleston wrote:
Anyone had any problems with ATA's running 3.0 software locking up?
Thanks, Billy
Shht, can someone send me 3.0.0 version of software
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When I press a key (8) on the phone, it should play a few bits of audio
and go to voicemail for testing. I dont get any sound back, and it
appears the call is progressing without me.
Here is the console output with sip debug:
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.10.
Hi,
How do i set the DTMF mode for incoming
SIP call per context ? Or is there a global config that i can set for all
context?
I am haing trouble getting the DTMF tones
from a cisco router with rfc2833 mode. when i make a call from 7960G via a 3640
(cisco call manager express) to asteris
Hello all,
A new version of asterisk-oh323 is now available. It contains
numerous minor fixes and updates. Among them, a fix for channels
using the G.729 codec (tested with codec_g729b.so codec).
Download from:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Regards,
Michael.
___
You need to "insmod" vpb module.
Daniel
Steven Kawuma wrote:
Hi,
My vpb.conf now reads:
[interfaces]
echocancel = on
board = 1
context = parlix_agents
; Note that V6PCI channel numbers start at 7!
mode = fxo
channel = 7
channel = 8
mode = dialtone
channel = 9
channel = 10
channel = 11
channe
Hello,
I`ve noticed the following, I`m dialing a number on the PSTN and I see
that PRI signaling returning that this number is unavailable/busy
whatever. this happens almost immediatly, but asterisk send 486 to the
sip phone only in about 10-12 seconds after that. I see the same
behavior for s
I will be using Asterisk to connect remote offices to PSTN - over IP (SIP).
These offices will use fxs gateways such as madiatrix and audiocodes to send
VoIP traffic to Asterisk. Asterisk will in turn push their traffic to PSTN.
The other way round will also work ie. Asterisk will forwards traffic
On Mon, Feb 09, 2004 at 09:26:43AM -0500, Michael Nigrelli wrote:
> Steve,
>
> Did you ever figure out why this happens. I have had asterisk up and
> running for a few weeks and all of a sudden this started happening.
Exactly the same here, it was running fine for about a month or so. Then one
d
I am running about 300 calls at the same time though pbx_spool.
I am getting these error messages. Where you see the Call failed to go
though reason 5, the call is dropped and never tried again. I looked at
the source code in ./pbx/pbx_spool.c and on line 199 I see the message but
I can't fig
Hi,
When I try to make Asterisk I get the following error:
In file included from aescrypt.c:39:
aesopt.h:156:22: endian.h: No such file or directory
aesopt.h:157:24: byteswap.h: No such file or directory
make: *** [aescrypt.o] Error 1
powerbk-g4:/build/asterisk-0.7.2 root#
Some say that zaptel an
Hi list.
Some of the available phones out thera have dual ethernet ports.
As we all know using SIP behind NAT could be frustrating.
Some vendors sells phones with dual ethernet ports. Are these just
incorporating a hub/switch functionality? The reason for my question is
that the normal case for a
The polycom phones (ip 500) have a simple switch built in. I say simple
because we had an issue were a network device was generating large
amounts of traffic to nonexistent mac addresses and the computer behind
the phone would not receive some packets. Probably not an issue though
as once we got
On Monday 09 February 2004 00:26, Soragan wrote:
> > Ohh.. You better give that to me then. I'll send you my Pentium
> > 133 w/ 16 megs of ram. It works great with the X100P.
>
> LOL, can your Pentium do web server, mail server with spam and
> virus checking and ADSL router all together? If it can
On Mon, Feb 09, 2004 at 04:36:41PM +0100, [EMAIL PROTECTED] wrote:
>
> Some vendors sells phones with dual ethernet ports. Are these just
> incorporating a hub/switch functionality? The reason for my question is
> that the normal case for a DSL customer is the possibilty to use one MAC
> adress fro
Hi
Is it possible to have an incoming call forwarded directly to an
internal user (we have ISDN and chan_capi)?
I have internal numbers like 101,102,103,104 and so on.
I need that an external user, that want to talk directly with one of us,
can digit our company number and when * answer the phon
> no why would it need to do that? If I put one in my 64 bit slots the
> machine won't boot.
Digium X100P's do work in 64-bit and 3.3V slots just fine.
Mark
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Shawn L. Djernes wrote:
Do you here the beeps on the phone or on the Console machine. For about the
last 2 weeks I have been hearing random beeps on either of my two sip
phones. I do not have a console running anywhere so I have no text
printing.
No, they were definately on the console, proble
--On Monday, February 9, 2004 8:35 am -0700 Erick Schmidt
<[EMAIL PROTECTED]> wrote:
When I try to make Asterisk I get the following error:
In file included from aescrypt.c:39:
aesopt.h:156:22: endian.h: No such file or directory
aesopt.h:157:24: byteswap.h: No such file or directory
make: ***
The newest firmware from grandstream supports configuration by mac address.
Simply upload a file cfg.txt
Does anyone know the format of a cfg.txt? â
[EMAIL
PROTECTED])fjåŠËbú?jË^®+$ºÇ±:5%H$HJ+º—Zµê)¶*'²ø¬ŠØm¶Ÿÿ–+-±ØŠéoæj)fjåŠËbú?jË^®+$ºÇ
Is this a crazy idea? I thought this would be ideal for a failover plan.
Anyone with experience on this?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren Martz
Sent: Saturday, February 07, 2004 1:32 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
The phones I am familiar with have two ports combined as a hub.
Paul
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, February 09, 2004 7:37 AM
To
I did 'sip show channels' during call. I saw 2 channels, one for each
device, that where on G729A. I thought that when a Native bridge was done *
was releasing the call. I did not think a call would require 2 G729 Channels
because * is just initializing the session between the 2 devices. I have
don
Matteo,
try: -
[incoming]
include => default ;default location for internal phones
exten => s,1,Answer
exten => s,2,Wait 10
exten => s,3,Dial(SIP/100)
exten => s,4,Hangup
Make sure that the context of incoming is defined in zapata.conf for pstn
calls.
Dave
-Original Message-
From:
Can anyone suggest a way to password protect a meetme
conference?
In extensions.conf
; Conferencing
exten => 18,1,Answer
exten => 18,2,Wait(1)
exten => 18,3,Meetme
exten => 18,4,Hangup
and this in meetme.conf
[rooms]
conf => 18
Paul Mahler
mail:[EMAIL PROTE
What does your extensions.conf look like? Did you answer() the call first ?
wes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Lee
Sent: Monday, February 09, 2004 6:01 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Help with Sip call problems - Wh
I have a Gigabyte K7 motherboard with an Athlon 2400+ processor. Before
the T1 install I had two T100P cards, one for the channel bank and the
other unused. This ran perfect for a month.
Last week we installed a new integrated T1 into the unused T100P (to
replace POTS lines and DSL.) In BIOS
i saw something about that on the voip-info wiki
On Mon, 2004-02-09 at 11:23, Matthew B Marlowe wrote:
> The newest firmware from grandstream supports configuration by mac address.
>
> Simply upload a file cfg.txt
>
> Does anyone know the format of a cfg.txt? â
>
___
We have an asterisk installation that's on a residential-grade DSL and
its port number (as visible from the outside) keeps changing, every time
it registers. fuser indicates that asterisk is only using port 4569 for
IAX2 (as it should), but when it goes out over the Internet, the port
number i
Matthew B Marlowe wrote:
The newest firmware from grandstream supports configuration by mac address.
Simply upload a file cfg.txt
Does anyone know the format of a cfg.txt? ÃÂÂ
ÃÂ???RÃÂ?f??)?+-?^?+$?K l?Ã ???r???b???v("?oÃÂo?j)fjÃÂ??b???j?^?+$?? ???PÃ
Hi Paul,
I believe it is...
> and this in meetme.conf
>
> [rooms]
>
> conf => 18
conf => 18,1234 ; 1234 is the PIN.
HTH,
Jason
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On Mon, 2004-02-09 at 17:40, Paul Mahler wrote:
> Can anyone suggest a way to password protect a meetme conference?
> In extensions.conf
> ; Conferencing
>
> exten => 18,1,Answer
> exten => 18,2,Wait(1)
> exten => 18,3,Meetme
> exten => 18,4,Hangup
>
> and this in meetme.conf
> [rooms]
> conf =>
Paul Mahler wrote:
Can anyone suggest a way to password protect a meetme conference?
In extensions.conf
; Conferencing
exten => 18,1,Answer
exten => 18,2,Wait(1)
exten => 18,3,Meetme
exten => 18,4,Hangup
and this in meetme.conf
[rooms]
conf => 18
Try using Authenticate.. eg e
Newbie question coming up ...
Is it possible to use the asterisk to initiate a call to a phone?
What I'm trying to determine is ways for software to connect to a
phone and send it a sound file with a message like:
Hello Mr. Jones. How are you doing today? Press 1 if you're OK.
Press
Hi list,
I have worked a little on debugging the Grandstream cfg.txt file and here is
what i found out so far:
The binary header is like this:
00 00 01 00 (always the same start) 52 d2 (checksum of 2 bytes) 00 0b 82 00
XX XX (phones mac address in hex) 0d 0a 0d 0a (two carriage return and new
line
On Mon, 2004-02-09 at 10:52, Matt Lawson wrote:
> We have an asterisk installation that's on a residential-grade DSL and
> its port number (as visible from the outside) keeps changing, every time
> it registers. fuser indicates that asterisk is only using port 4569 for
> IAX2 (as it should), bu
Yasir Rahman [EMAIL PROTECTED] wrote:
> I am trying to register SJPhone with my asterisk server but my SJPhone
> messages saying NON-INVITE transaction..
>
> [snip]
>
I found your message in my spam bin. Others may not have seen your
message at all. Perhaps you might get a better response if y
Matt Lawson wrote:
We have an asterisk installation that's on a residential-grade DSL and
its port number (as visible from the outside) keeps changing, every
time it registers. fuser indicates that asterisk is only using port
4569 for IAX2 (as it should), but when it goes out over the Internet
Hi,
i'm here in germany still fighting against my problems ...
We have a e100p which is sending out his callerid as 78107-0. But what i
need is to send out the extension of the inside callers to, for example:
78107-14
So what i tried is:
exten => _00XX.,1,SetCallerID(78107${CALLERIDNUM})
exten =
> We have an asterisk installation that's on a residential-grade DSL and
> its port number (as visible from the outside) keeps changing, every time
> it registers. fuser indicates that asterisk is only using port 4569 for
> IAX2 (as it should), but when it goes out over the Internet, the port
> n
Have a look at http://www.plugndial.com/aps_sample.html
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: 09 February 2004 17:03
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Firmware for Grandstream Phones -
Supports CFG by MAC ad
Title: Message
If you
are running 0.72 version... then in meetme.conf you need to
have:
conf
=> ROOMNO,PASSWRD ie. 100,123
Ta
SJ
/me points to sample.call into * source dir...
Matteo.
Il lun, 2004-02-09 alle 18:21, John Chambers ha scritto:
> Newbie question coming up ...
>
> Is it possible to use the asterisk to initiate a call to a phone?
>
> What I'm trying to determine is ways for software to connect to a
> phone
Yes, you can certainly do something like this. We do outgoing applications
for our customers, similar to this.
Basically, you dump a triggering text file into
/var/spool/asterisk/outgoing, which asterisk checks for every second. This
causes an outgoing call to be made based on the dialplan entry
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> John Chambers
> Sent: Monday, February 09, 2004 12:21 PM
> To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Can asterisk make a call to a phone?
>
>
> Newbie question coming up
The way I found out was I installed the new firmware and watched my TFTP server :)
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Suffill
Sent: Monday, February 09, 2004 11:51 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Firmware for
Currently in progress of trying to debug similar
problem on my own system. Sometimes it happened
during call transfers, but this last time,
it happened all by itself at 4:00 AM, no calls even
close. Complete system Freeze, Nothing at all
workings, except the reset button.
You setup is vastly di
use call files there is should a sample in the asterisk src
On Mon, 2004-02-09 at 12:21, John Chambers wrote:
> Newbie question coming up ...
>
> Is it possible to use the asterisk to initiate a call to a phone?
>
> What I'm trying to determine is ways for software to connect to a
> phone and
Probably a dumb question, but what's the best Linux variant to use to
build/run an Asterisk server.
Hardware is Compaq DL360 with a Widcard 410.
Debian/Fedora Core ?
Steve
--
NetTek Ltd Phone/Fax +44-(0)20 7483 2455
SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19
_
I'm having bad echo between TDM and
SIP. There's no echo between TDM-TDM though. I've seen this post from
JTodd:
; Config notes:; - in /usr/src/zaptel/Makefile, set
KFLAGS+=-DECHO_CAN_MARK2; - in /usr/src/zaptel/Makefile, set
KFLAGS+=-DAGGRESSIVE_SUPPRESSOR ;; I compile with these tw
> The way I found out was I installed the new firmware and watched my TFTP
server :)
>
Maybe they are on the way to support APS configuration as sipphone.com have
made on www.plugndial.com (hope so)
Jens
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RH9. If it happens again I think I'll drop back to 0.7.1.
Jonathan Biggs wrote:
Currently in progress of trying to debug similar
problem on my own system. Sometimes it happened
during call transfers, but this last time,
it happened all by itself at 4:00 AM, no calls even
close. Complete syste
Did you have any active meetme sessions at the time of the freeze?
What Asterisk version are you using?
MATT---
-Original Message-
From: Jonathan Biggs [mailto:[EMAIL PROTECTED]
Sent: Monday, February 09, 2004 1:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] System freeze
Cu
How would you work the dial command to dial out using Voiceglo?
-Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Bittner
Sent: Friday, February 06, 2004 8:15 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RE:voiceglo sip config
Hi,
Aft
On Mon, Feb 09, 2004 at 02:28:02PM +0100, Dawid Mielnik wrote:
>
> Would the t.38 transmission be properly handled by the t.38 supporting end
> points whith mediastrem passing through Asterisk ? (dont have much
> experience with t.38) Has anyone ever tried anything similar / different /
> wierder
Johannes von Drachenfels wrote:
Hi,
i'm here in germany still fighting against my problems ...
We have a e100p which is sending out his callerid as 78107-0. But what i
need is to send out the extension of the inside callers to, for example:
78107-14
[..]
But i still can see only the 78107-0 when
what you feel most confortable with.
there's a best linux distro for *.
Matteo.
Il lun, 2004-02-09 alle 19:20, Steve Kennedy ha scritto:
> Probably a dumb question, but what's the best Linux variant to use to
> build/run an Asterisk server.
>
> Hardware is Compaq DL360 with a Widcard 410.
>
>
On Mon, Feb 09, 2004 at 06:20:54PM +, Steve Kennedy said:
> Probably a dumb question, but what's the best Linux variant to use to
> build/run an Asterisk server.
>
> Hardware is Compaq DL360 with a Widcard 410.
>
> Debian/Fedora Core ?
Not dumb, but you won't get a scientific answer. You wil
Steve Kennedy wrote:
Probably a dumb question, but what's the best Linux variant to use to
build/run an Asterisk server.
Hardware is Compaq DL360 with a Widcard 410.
Debian/Fedora Core ?
Steve
Which ever one you are most happy with is probably the best answer..
_
Did you possibly have astman running on the localhost? I found that I was getting
kernel panics while using astman on an SMP machine with dual T400P cards. Did you see
the message on the console before you reset the box? Did you possibly have a serial
console connected logging console message
> The phones I am familiar with have two ports combined as a hub.
>
> Paul
OK, but how would an scenario look like where You only would be "allowed"
to use one authorised MAC? Is it possible or forget it and try to get that
NAT stuff working!!
/t
>
>
> Paul Mahler
> mail:[EMAIL PROTECTED]
> pho
Hi!
> I have worked a little on debugging the Grandstream cfg.txt file and
> here is what i found out so far:
Have you check this?
http://www.voip-info.org/
tiki-index.php?page=Asterisk+phone+grandstream+budgetone
You are strongly encouraged to add any additional findings yourself on
the Wiki
We've had 2 "unexplainable" system freezes.
We have SMP and Redhat 9 2.4.20-20.9. There has been no evidence
anywhere of why our system crashed.
Tan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Welter
Sent: 09 February 2004 18:39
To: [EMA
I've looked, poked, and hoped, but I can't seem to make * understand
the difference between a SIP channel being busy or not being there.
Both come up as 'busy'. I would expect the unregistered SIP to be seen
as unavailable. Am I just missing something obvious, again?
Tim
--
>
You must register with cisco in order to get ata image.
- Original Message -
From: "Anton Tinchev" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, February 09, 2004 8:48 AM
Subject: Re: [Asterisk-Users] Problems with ATA's locking up..
> Billy Huddleston wrote:
>
> >Anyone ha
-do you use hyperthreading
-do you use the LD_ASSUME_KERNEL=2.4.1 b4 loading asterisk
-have you compiled zaptel with the SMP flag on
Can anybody site some real hardcore technical facts
about SMP & hyperthreading support in the RH9 kernel rpm images
I hear what i would call 'old wives tales' about
have you tried this gs-102 with pppoe? verizon dsl uses pppoe. pppoe is
logically like dhcp, but using ppp for added feature like aaa :) can this
unit connect directly to a cable modem?
- Original Message -
From: "Nicolas Bougues" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monda
That's just the way Asterisk's dial command works.
On Mon, 2004-02-09 at 13:16, Tim Sailer wrote:
> I've looked, poked, and hoped, but I can't seem to make * understand
> the difference between a SIP channel being busy or not being there.
> Both come up as 'busy'. I would expect the unregistered S
The Cisco 7960 and 7940's have a two-port SWITCH built into the phone, at
least according to Cisco they do.
John
Paul Mahler wrote:
The phones I am familiar with have two ports combined as a hub.
Paul
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
__
On Mon, Feb 09, 2004 at 01:37:55PM -0600, Eric Wieling wrote:
> That's just the way Asterisk's dial command works.
Hmm. I see. If it can't create the channel for either reason
(busy or not registered), it's handled the same. I think I'll
kludge up a perl script to watch the SIP channels register a
> Currently in progress of trying to debug similar
> problem on my own system. Sometimes it happened
> during call transfers, but this last time,
> it happened all by itself at 4:00 AM, no calls even
> close. Complete system Freeze, Nothing at all
> workings, except the reset button.
>
> You set
I've looked around and found previous discussion about this, but so far I
have not seen any answers that really solve this problem.I'd like
to integrate an intercom system into Asterisk so that users could dial an
extension, the phone on the other end would emit a beep, and then the
speakerphone wo
On Monday 09 February 2004 01:20 pm, Steve Kennedy wrote:
> Probably a dumb question, but what's the best Linux variant to use to
> build/run an Asterisk server.
>
> Hardware is Compaq DL360 with a Widcard 410.
>
> Debian/Fedora Core ?
>
>
> Steve
Nah, go with good ol' DOS 3.3!
You will not have
On Mon, 9 Feb 2004 [EMAIL PROTECTED] wrote:
> How would you work the dial command to dial out using Voiceglo?
I use:
exten => _9NXX,1,Dial(SIP/1801${EXTEN:[EMAIL PROTECTED])
where 'voiceglo' is the name of the service definition in my sip.conf:
[voiceglo]
username=
secret=
...
Greg
There appears to be a problem with Asterisk negotiating a codec with my
x-lite clients (both mac and windows).
All codecs were enable in the clients, and sip.conf contained:
allow=all ; Allow all codecs
Using 'show sip debug' on the * console would print the following when
a c
- Original Message -
From: "Eric Wieling"
To: <[EMAIL PROTECTED]>
Sent: Friday, January 23, 2004 11:07 AM
Subject: Re: [Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway
> The Packet8 8x8 DTA-310 that I have ran SIP when I was using it.
Eric, did you get the DTA fully working w
I had two system freezes this weekend, first time. I just setup Musiconhold. The
kernel panic referenced mpg123. I turned off musiconhold until I could look into
it more.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Mark Spencer
I am trying to muddle my way tthrough getting something - actually
anything to work - with Asterisk. I've acquired a Grandstream phone and
I've got * on a Red Hat 9 box. I've gotten to a point where I can see
(via ethereal) that the phone REGISTER's successfully with asterisk, and
then I try
On Mon, 2004-02-09 at 13:54, Mark Spencer wrote:
> > Currently in progress of trying to debug similar
> > problem on my own system. Sometimes it happened
> > during call transfers, but this last time,
> > it happened all by itself at 4:00 AM, no calls even
> > close. Complete system Freeze, Noth
Please include your sip.conf and extension.conf files. Hard to say what is
wrong without seeing the configuration
- Original Message -
From: "Bill Michaelson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, February 09, 2004 3:15 PM
Subject: [Asterisk-Users] asterisk-grandstream
I would like to request an alternative to the mpg123-only musiconhold. I
could live with just about anything that isn't mp3. Just as an alternative
for those of us who have had nothing but problems with mpg123 and would like
to use the functionality of musiconhold.
MATT---
-Original Message--
> I had two system freezes this weekend, first time. I just setup
> Musiconhold. The kernel panic referenced mpg123. I turned off
> musiconhold until I could look into it more.
Again, please post your comments *on the tracking bug* number 963, at
bugs.digium.com. Please include whether you have S
Hi!
> What I would do i this situation is work out a fax <--> email gateway.
> Best case this could be done entirely with software on the asterisk
> box, worst case a faxmodem hairpinned into an fxs card using hylafax.
Why exactly would hylafax be a "worst case" solution only, why would you
tin
Don't allow=all. Don't ever allow=all. In fact don't even think about
allow=all. Personally I would like the allow=all option REMOVED.
disallow=all
allow=ulaw
If that works then allow whatever codec you want instead of ulaw.
On Mon, 2004-02-09 at 14:01, Ryan Courtnage wrote:
> There appears t
Plug the Packet8 DTA310 into an X100P and go.
On Mon, 2004-02-09 at 14:06, Jim Flagg wrote:
> - Original Message -
> From: "Eric Wieling"
> To: <[EMAIL PROTECTED]>
> Sent: Friday, January 23, 2004 11:07 AM
> Subject: Re: [Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway
>
>
>
Has anyone experienced problems with dialup through asterisk. I'm having some varied
success with dial-in and dial-out.
All my analog extensions are connected to * via Adtran 750 FXS channelbanks using
FXO_KS signalling. I have a longdistance T-1 (e&m_w) from sprint and a local T-1 PRI
from V
Hi!
> I've looked, poked, and hoped, but I can't seem to make * understand
> the difference between a SIP channel being busy or not being there.
> Both come up as 'busy'. I would expect the unregistered SIP to be seen
> as unavailable. Am I just missing something obvious, again?
You are right, th
On Mon, Feb 09, 2004 at 09:31:30PM +0100, Philipp von Klitzing wrote:
>
> Why exactly would hylafax be a "worst case" solution only, why would you
> tink that that the Asterisk solution is better at all?
The "worst case" would be the modem hairpinned into an FXS
port, not hylafax per se.
> > I
Would the following be a doable solution:
1. An Asterisk-box on site with FXS
2. Plug Fax into FXS
3. User uses facsimile machine to call a number - Asterisk answers
4. Stores called number into variable ${FAXDESTINATION}
5. Use RcfFax of * to store fax within asterisk
6. mail stored fax together
Steve Kennedy wrote:
Probably a dumb question, but what's the best Linux variant to use to
build/run an Asterisk server.
Hardware is Compaq DL360 with a Widcard 410.
Debian/Fedora Core ?
Steve
As others have said: Whatever you're most comfortable with. Having said
that though, I'm partial t
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