Hello,
OK, some things I've found out so far. The ground connection
to the ADIT chassis wasn't really to ground (fixed that, it
made FXS card happy when connected).
Taking a cue from another post I also reduced the number of
options specified in zapata.conf to:
[trunkgroups]
[channels]
Hi all
I have installed Astrix on FC4 and running successfully
and installed Astbill on top of the server
and able to mange accounts
i have made 2 extenstions
17612 17349
and iam able to use soft SJPhone
and able to register
and when i try to call 17349
i get an error
Address incomplete
John Todd wrote:
I'm having a problem with Asterisk sending too many INVITEs to a peer for a
single call. I can't quite figure out why there are these rapid INVITEs sent
to the call proxy. The call completes correctly (to, in this example, an
echo test found via ENUM) but the number of
Leif Neland wrote:
On 08:48, Tue 29 Nov 05, [EMAIL PROTECTED] wrote:
From memory (at a previous installation) you will need a newer
version of
Asterisk than 1.09 for the lights to work.
on 1.0.9 the lights work.
In this way:
person is on the phone: light is on
Person is not on
What does the TE406 leds indicate?
Both the ADIT 600 led and the TE406 led are green, the ADIT
has zeros in the error counters. Syslog has this as a final
message after running ztcfg:
Nov 28 02:31:08 xxx kernel: Registered tone zone 0 (United States /
North America)
Nov 28 02:36:21
I'm trying to implement some of the star services such as *61 for
weather or *71 for wakeup call, etc. I think I have asterisk setup
properly because I can get them to work fine using normal extension
numbers. However, if I try to use the 'star' numbers, my Polycom IP500
never sends the
If you edit sip.conf in 1.2 and put
Vmexten = voicemail
Fromdomain = yourip or domain of the asterisk box
Then in extensions.conf
exten = voicemail,1,VoicemailMain(${CALLERIDNUM})
That works and look nicer on the snom phones.(it dials voicemail)
Under 1.2 you can put in sip.conf
Fromuser =
(Zap/3-1, record-enable|38626540259|OUT) in new stack
-- Executing GotoIf(Zap/3-1, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(Zap/3-1,
recordingcheck|20051129-095434|1133254470.611) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin
Well... I don't have an ADIT box around, so can't help on that.
Do take a close look at the channel assignment stuff, both in zaptel.conf
and zapata.conf. Are you absolutely sure the ordering of the cards
and channels are right (haven't moved any cards around or removed any)?
Your statement it
I still cannot get this to work on 1.0.9.
I am trying to test with two extensions:
Here is the config I am using:
exten = 451,hint,sip/451
exten = 451,1,Dial(SIP/451,20,tr)
exten = 451,2,Voicemail([EMAIL PROTECTED])
exten = 451,102,Voicemail([EMAIL PROTECTED])
exten = 453,hint,sip/453
exten =
On Nov 28, 2005, at 3:55 PM, BJ Weschke wrote:
On 11/28/05, Martin Joseph [EMAIL PROTECTED] wrote:
snipI am only able to get comedian voicemail (ie dialing 1234) to
record or
playback messages if I use the GSM codec? Is this normal and
expected?
If I use ulaw or alaw I get either trash
Hi
Luke,
It's
important to compare apples and pears though.
The
card you mentioned has 24 on board Digital Signal Processors that enable it to
do the following:
Tone
Detection
Voice
Activity Detection
Conferencing with automatic Gain Control and echo
cancellation
Hello,
On Tue, 2005-11-29 at 02:25, Rich Adamson wrote:
Well... I don't have an ADIT box around, so can't help on that.
Do take a close look at the channel assignment stuff, both in zaptel.conf
and zapata.conf. Are you absolutely sure the ordering of the cards
and channels are right (haven't
How can I set a variable in a .call file?
I wanted to add a fax header with SpanDSP / txfax, and the information
on soft-switch.org says:
If the variable LOCALHEADERINFO has been set when txfax is run, the
value of that variable will be used as the user defined part of the
header text.
So
I'm testing asteriskathome with an ISDN card
00:0a.0 Network controller: Cologne Chip Designs GmbH ISDN network
controller [HFC-PCI] (rev 02)
I found there is the module hisax and I loaded it:
hisax 456177 0
crc_ccitt 2113 2 hisax,zaptel
isdn
Alejandro Vargas schrieb:
I'm testing asteriskathome with an ISDN card
00:0a.0 Network controller: Cologne Chip Designs GmbH ISDN network
controller [HFC-PCI] (rev 02)
I found there is the module hisax and I loaded it:
hisax 456177 0
crc_ccitt 2113 2
Hi, I have a trivial setup on a 2.4GHz Xeon Dell PE 1750 SCSI machine
dealing with 4 ports of E1 in an 'inline PBX' arrangement.
My extensions.conf is simply:
[general]
static=yes
writeprotect=yes
[frompstn]
exten = _31.,1,Dial(Zap/g2/${EXTEN})
exten = _31.,2,Congestion
[fromaxxess]
exten =
Hi,
I'm trying to setup an asterisk-based PBX on a Debian Sarge distro. I'm
using a K8N-E deluxe asus motherboard which gives me some problems (but
I'm not sure is the motherboard causing the problem):
- if I plug a TDM400 REV J, Debian cannot recognize it
- if I plug a TDM400 REV E/F,
sure? have you tried latest drivers?
could be simply a pci-id problem.
matteo.
Il giorno mar, 29/11/2005 alle 11.59 +0100, gincantalupo ha scritto:
Hi,
I'm trying to setup an asterisk-based PBX on a Debian Sarge distro. I'm
using a K8N-E deluxe asus motherboard which gives me some problems
2005/11/29, Tomasz Chmielewski [EMAIL PROTECTED]:
you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card,
not HiSax (well, technically, you could use HiSax too, but avoid that if
possible).
I prefered to use hisax because it is already included in
asteriskathome (why bristuff
James B. MacLean wrote:
Rich Adamson wrote:
From: James B. MacLean [EMAIL PROTECTED]
Asterisk*CLI zap show status
Description Alarms IRQ
bpviol CRC4
Wildcard TDM400P REV E/F Board 1 OK 0
0
Thanks, I will try thats.
El lun, 28-11-2005 a las 17:23 -0500, C F escribió:
Looks like it's losing it's connection to the DNS server, make sure
you don't have any names that need to be resolved to IP address in any
of the config files for asterisk. Just use IP address.
There are other known
I have successfully upgraded to 1.2, but there is no change at all.
Asterisk sees the subscriptions fine:
asterisk_test*CLI sip show subscriptions
Peer UserCall ID ExtensionLast state
Type
195.27.242.113 320 3c26700c2e6 453 Idle
José Luis Gómez ha scritto:
Thanks, I will try thats.
There was an issue in the ast_sip_ouraddrfor function. When the dns is
down it fails to get the right address, you can easy patch it looking to
the new code
Sergio
___
--Bandwidth and
I've used one with a Snom SIP server system
it worked quite well but not tried it with * unfortunately.
Voxtream support team are excellent though I'm sure they'll help
you get it working.
Robert Rozman wrote:
Hi,
we're having quite some problems with new hardware we're testing -
Im trying to get Asterisk to send
out voice alerts in conjunction with Nagios.
Basically what happens is depending on the
type of failure Nagios has seen a file will be created with the correct
contacts phone number in the file.
It will also put the correct context in
the file depending
Hi all,
I would like to run my perl agi script when the call is hungup. I did
one script to calculate calling balance and duration.
I made one timer Where the dialstaus is Answered But i am am in
confiuse how i can stop my timer when the dialstus will be hangup.
Please give me an advice to
Hi all,
How i can call my perl agi script when the call is hungup. Because i
am making some external Cdr calculation.
--
Best Regards,
Abdul Lateef Khan
Computer Programmer
Mobile No. : +974 - 5405022
ICQ : 276-994-704
YM! : [EMAIL PROTECTED]
MSN : [EMAIL PROTECTED]
Google Talk : [EMAIL
Hi
Is anyone using a vegastream product with asterisk?
I have various numbers coming into the vegastream vega400 and was after some
exmaple config for use with the asterisk server so it can perhaps reister with
the vega and recieve these numbers???
Any help or pointers in the right direction
2005/11/29, Tomasz Chmielewski [EMAIL PROTECTED]:
you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card,
not HiSax (well, technically, you could use HiSax too, but avoid that if
Ok, I downloaded both bristuff-0.2 and bristuff 0.3. 0.2 don't
compiled. 0.3 yes, but it broke
What's the output of show hints?
office-pbx*CLI sip show subscriptions
Peer UserCall ID ExtensionLast state Type
192.168.2.46 700 3c26700c5f3 703 Idle
dialog-info+xml
192.168.2.46 700 3c26700c557 702
Code Lover wrote:
Hi all,
How i can call my perl agi script when the call is hungup. Because i
am making some external Cdr calculation.
Hi M. Lover,
There are two solutions for you:
- You can call an AGI on hangup by using the extension 'h' : exten =
h,1,DeadAGI(myagi.agi)
- If you're
Hi Matteo,
thanks for answering, your advise seemed right but no pci or motherboard
driver is avalaible on ASUS site.
I think we'll use another motherboard.
This is another motherboard with great problems as Dell hardware.
Thanks
Giorgio Incantalupo
Matteo Brancaleoni wrote:
sure? have
I'm trying to get Asterisk to send out voice alerts in conjunction
with
Nagios.
Basically what happens is depending on the type of failure Nagios has
seen
a file will be created with the correct contacts phone number in the
file.
It will also put the correct context in the file
Hi!
I still cannot get this to work on 1.0.9.
exten = 451,hint,sip/451
* Try hint,SIP/451 instead of hint,sip/451. The bugtracker has an
open ticket on case-sensitivity of the hint priority.
* Make sure that in the advanced settings your SNOM is set to not filter
packets from registrar
On Tue, November 29, 2005 13:17, Alejandro Vargas said:
2005/11/29, Tomasz Chmielewski [EMAIL PROTECTED]:
you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card,
not HiSax (well, technically, you could use HiSax too, but avoid that if
Ok, I downloaded both bristuff-0.2 and
I see two problems. First the dialplan is not finding your context.
The second is that when your call is made over IAX, your box is seeing
it as answered and immediately playing goodbye before it is actually
answered.
I think the reason it just hangs up is it falls back to the default
Hi!
on 1.0.9 the lights work.
In this way:
person is on the phone: light is on
Person is not on the phone: light is off
since 1.2 the lights will blink when the phone is running
and above states work the same.
Side note: Asterisk v1.2.0 comes with a new sip.conf setting:
Tony Spencer wrote:
I think the reason it just hangs up is it falls back to the default context
which is in extensions.conf:
[default]
include = ext-local
exten = s,1,Playback(vm-goodbye)
exten = s,2,Macro(hangupcall)
I read it as if it was trying to match the context on the remote
Seems like it, thnx
Philipp von Klitzing wrote:
Hi!
on 1.0.9 the lights work.
In this way:
person is on the phone: light is on
Person is not on the phone: light is off
since 1.2 the lights will blink when the phone is running
and above states work the same.
Side note: Asterisk v1.2.0
I changed hint using upper case SIP instead of lower case sip, and this
solved my problem.
Very strange indeed.
Thanks to all for input.
Regards,
Joe
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
David Waugh wrote:
This means for example that the card could be used for a conferencing
application with 24 users with echo cancellation/ gain control being handled by
the card - and not having to be processed by the central CPU.
That is correct, of course, but keep in mind that having
2005/11/29, Francesco Peeters [EMAIL PROTECTED]:
Then add to a startup file like rc.local:
modprobe zaptel
modprobe zaphfc
ztcfg -vv
to start and initialize the cards...
I'll try... when somebody goes to reset the machine. I'm configuring
it through ssh and it hanged when I was trying
hum, may be a mismatching between the asterisk source and the mysql
module source. Where are you getting the sources and please explain how
are you starting the compilation.
Best RegardsOn 11/27/05, Abdul Lateef Khan [EMAIL PROTECTED] wrote:
Hi all,Did anyone installed asterisk-addons
First remember that for each change in zapata.conf you must restart
asterisk, not only reload configuration. Now, could you provide a
link to show us your zaptel.con and zapata.conf?
when you type ztcfg -vv ? what does the output says exactly?
best regardsOn 11/26/05, Scott Geist [EMAIL
I am trying to install the qozap driver, but when I doing: make all the shell command show error in qozap.o.What can I doing for compiling qozap.o?Thanks
Yahoo! Messenger: chiamate gratuite in tutto il mondo ___
--Bandwidth and Colocation
zahfc mode loaded ?
try lsmod to verify
try ztcfg -vvv
sleep 3
ztcfg -vvv
2005/11/29, Alejandro Vargas [EMAIL PROTECTED]:
2005/11/29, Francesco Peeters [EMAIL PROTECTED]:
Then add to a startup file like rc.local:
modprobe zaptel
modprobe zaphfc
ztcfg -vv
to start and
Couple of other items to look at... the 'zap show channels' should look
something like:
pseudoinbound-bus-lin en default
1inbound-bus-dia en default
I don't see the 'Language' colume on your display below. Does your
zaptel.conf include:
So does this problem only surface with delete=yes?
I am using 1.0.9 and do not have the second comma.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
-
I do not know if asterisk uses standard regexp, but in regexp you would use:
[(201)(202)(203)(205)(206)]
This would match any of the groups () of numbers.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - -
On Tue, 29 Nov 2005, Kevin P. Fleming wrote:
David Waugh wrote:
This means for example that the card could be used for a conferencing
application with 24 users with echo cancellation/ gain control being
handled by the card - and not having to be processed by the central CPU.
That is
I'm trying to setup an asterisk-based PBX on a Debian Sarge distro. I'm
using a K8N-E deluxe asus motherboard which gives me some problems (but
I'm not sure is the motherboard causing the problem):
- if I plug a TDM400 REV J, Debian cannot recognize it
- if I plug a TDM400 REV E/F,
Asterisk*CLI zap show status
Description Alarms IRQ
bpviol CRC4
Wildcard TDM400P REV E/F Board 1 OK 0
0 0
Wildcard TDM400P REV I Board 2 OK 0
0 0
---End of
Ben Buxton wrote:
Can't say I've actually tried IM, but Ill give it a go sometime. I think
the wiki needs updating on all this...the eyebeam page is very
incomplete on subscribe, im, etc.
I've got online offline status and the eyeBeam will display messages you send
to it with SendText while in
I'm a bit of newbie to Asterisk so I'm not to sure.
I was just given the task to try and make this work.
You could be correct but I'd have to do some further investigation and speak
to the person that used to admin this server.
All I want to do is call a phone number and play a audio file and
Hi,
I've been using module assistant first time than using make linux26 seems OK
now, meaning I still have PRI and hisax
but in 2nd position and wcfxo OK.
risk2:/usr/src/asterisk-1.2.0# lsmod | grep zaptel
zaptel228644 1 wcfxo
crc_ccitt 2144 2 zaptel,hisax
Halas, I
Hi,
I've been using module assistant first time than using make linux26 seems OK
now, meaning I still have PRI and hisax
but in 2nd position and wcfxo OK.
risk2:/usr/src/asterisk-1.2.0# lsmod | grep zaptel
zaptel228644 1 wcfxo
crc_ccitt 2144 2 zaptel,hisax
Halas, I
Hi all,
i'm wondering if anyone has ever managed to get moh
working on Siemens OptiPoint400?
if yes, can you please explain how you did it...
thx.
__
Yahoo! Mail - PC Magazine Editors' Choice 2005
http://mail.yahoo.com
Client wants to use a *67 feature to block caller id on next call. In the
Wiki I have seen references to this being available but I haven't see any
code to actually make it work. Does anyone have a quick solution for
implementing this type of function?
-Kerry
Assuming AMP style contexts:
PRI:
[from-internal]
exten = *67,1,SetCallerID( )
exten = *67,2,SetCallerIDName( )
exten = *67,3,SetCallerIDNum( )
exten = *67,4,Playback(YourCustomPromptStar67IsEnabled)
exten = *67,5,DISA(no-password|from-internal)
POTS:
[from-internal]
exten =
On Tue, Nov 29, 2005 at 04:35:27PM -0800, Geo wrote:
Hi,
I've been using module assistant first time than using make linux26 seems OK
now, meaning I still have PRI and hisax
The module hisax is harmless. Ignore it. Black-list it (see my previous
mail) if it bothers you ion the logs and
I will install it and test it. Thanks.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Tuesday, November 29, 2005 8:15 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Caller ID
On Tue, Nov 29, 2005 at 03:56:24PM +0100, asterisk183 wrote:
I am trying to install the qozap driver, but when I doing:
make all
the shell command show error in qozap.o.
What can I doing for compiling qozap.o?
Thanks
Start by giving the telepathy-chalanged among us some
Could you just use a different start number?
9 to dial out. 8 to dial out with blocked callerID.
Then just preface the callerID block code for the Telco.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - -
Rich Adamson wrote:
Thanks for the heads up. More dissappointing is that the E/F card is
the newer card purchased. Where can I go to see when certain revisions
were released? Surprising that the newer card just purchased (to me)
is the older rev :(.
You can probably
Is there a way to monitor zaptel errors with something like Nagios?
I have a TE405P and seldomly I see messages like this:
Zaptel: Master changed to TE4/0/1
wct4xxp: Setting yellow alarm on span 4
wct4xxp: Clearing yellow alarm on span 4
which means that somehow the T1 went down and came back
Hi,
I am trying to execute the following asterisk command from one of my AGI script.
By providing 'C' flag, I exected CDR would reset.
Problem is, CDR was reset but CDR didn't grab destination number (extension)
from the Dial command.
Well my AGI script was executed after answering a call on a
I risolved my problem: I have kernel source in /usr/src/linux-2.4.686 instead of /usr/src/linux, therefore the qozap.c doesn't compiling.ThanksTzafrir Cohen [EMAIL PROTECTED] ha scritto: On Tue, Nov 29, 2005 at 03:56:24PM +0100, asterisk183 wrote: I am trying to install the qozap driver,
OK, then this is easy. Instal Asterisk in the central location, along with a
Sipura SPA-3000. Configure that unit to answer the incoming POTS line and act
as a VOIP gateway for Asterisk. Then configure two additional SPA-3000 units,
one at each employee's location. Then, configure Asterisk (I
What's the 'format' line of the [general] section of your voicemail.conf?
Martin Joseph wrote:
On Nov 28, 2005, at 3:55 PM, BJ Weschke wrote:
On 11/28/05, Martin Joseph [EMAIL PROTECTED] wrote:
snipI am only able to get comedian voicemail (ie dialing 1234) to
record or
playback messages if
Actually, Matteo meant zaptel drivers, not motherboard or chipset
drivers from ASUS :)
Mojo
gincantalupo wrote:
Hi Matteo,
thanks for answering, your advise seemed right but no pci or motherboard
driver is avalaible on ASUS site.
I think we'll use another motherboard.
This is another
Actually, why not:
exten = *67XXX,1, {etc}
-Original Message-
From: Steven [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 29, 2005 9:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Caller ID Block (*67)
Could you just use a different start number?
9
Are your interrupts getting hogged by anything else? I'd recommend
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting if
you haven't already read it. Have you tried booting with noapic kernel
option? You may then have to shuffle cards around to make your sangoma
not share
Hello All,
It seems that voicepulse is not taking any new orders on the standard
service plans (though vp connect seems unaffected) due to the fcc
rulings.
We'll see what happens, anyone having similar problems with other
services as of today?
Greg
___
On 09:46, Tue 29 Nov 05, Erik wrote:
Leif Neland wrote:
On 08:48, Tue 29 Nov 05, [EMAIL PROTECTED] wrote:
From memory (at a previous installation) you will need a newer
version of
Asterisk than 1.09 for the lights to work.
on 1.0.9 the lights work.
In this way:
On Tue, 29 Nov 2005 06:14:54 +, scott wrote:
Is anyone using a vegastream product with asterisk? I have various
numbers coming into the vegastream vega400 and was after some
exmaple config for use with the asterisk server so it can perhaps
reister with the vega and recieve these
On 18:26, Mon 28 Nov 05, Sascha Deri wrote:
I made an error in what I previously wrote. What actually works in v1.2 is:
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
Which is what Michael originally wrote. My bad!
:) To err is human :)
I know for sure it had to work since I copied it
On 00:24, Tue 29 Nov 05, bram kortleven wrote:
Are there any example configs? Or does anybody have a default config
for this setup:
1 analog digium clone card for an analogue line (my home line)
Several sip phones (a few of them on the outside of my lan (NAT fw
between) and 2 insde my lan)
On Tue, November 29, 2005 16:04, Giovanni Miano said:
zahfc mode loaded ?
try lsmod to verify
try ztcfg -vvv
sleep 3
ztcfg -vvv
Also helpful is
cat /proc/zaptel/*
This'll tell you whether zaptel is loaded, whether the channels have been
defined, and what their status is...
On Tue, November 29, 2005 18:36, [EMAIL PROTECTED] said:
Hello All,
It seems that voicepulse is not taking any new orders on the standard
service plans (though vp connect seems unaffected) due to the fcc
rulings.
We'll see what happens, anyone having similar problems with other
services as
But, star at least works. I've got *xxT in my digitmap and it caught
*69. In fact, my 1.5 admin guide refers to Section 2.1.5 of RFC 3435,
the MGCP rfc, which does allow the * to be used
Moj
Rich Adamson wrote:
I'm trying to implement some of the star services such as *61 for
weather or *71
Since upgrading to 1.2 I'm seeing the following iin my
/var/log/asterisk/messages:
Nov 29 11:50:20 NOTICE[13094] app_dial.c: Unable to
create channel of
type 'Zap' (cause 17 - User busy)
Nov 29 12:02:06 WARNING[12977] chan_iax2.c: chan_iax2:
ast_sched_runq
ran 249 scheduled tasks all at once
On Mon, 2005-11-28 at 12:24 -0800, Innocent Evil wrote:
What is the purpose of cdr_manager.conf?
cdr_manager.conf allows you to configure asterisk to send call detail
records (cdr) via the Manager API.
How I can configure it?
to enable CDR via Manager API a cdr_manager.conf looks like this:
;
Jason Marshall wrote:
OK, then this is easy. Instal Asterisk in the central location, along
with a Sipura SPA-3000. Configure that unit to answer the incoming
POTS line and act as a VOIP gateway for Asterisk. Then configure two
additional SPA-3000 units, one at each employee's location.
Are there any example configs? Or does anybody have a default config
for this setup:
1 analog digium clone card for an analogue line (my home line)
Several sip phones (a few of them on the outside of my lan (NAT fw
between) and 2 insde my lan)
Or a simple way of configging through a
Channel: Local/[EMAIL PROTECTED]Callerid: 01612370660MaxRetries: 5RetryTime: 300WaitTime: 45Context: serverdownExtension: sPriority: 1On 29 Nov 2005, at 15:39, Tony Spencer wrote:I'm a bit of newbie to Asterisk so I'm not to sure.I was just given the task to try and make this work.You could be
Hi everybody:
Is the right behavior of the IAXmodem to display
Registration completed successfully and remote hangup many times?
Regards
Miguel
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or
Entries in the queue_log file do not match what the documents say. The
COMPLETECALLER and COMPLETEAGENT events do not have the 3rd agrument of
origposition. I'm using Asterisk 1.0.9 currently(will be upgrading
shortly). I've checked and this should be done by the old stable
version we are
Alejandro Vargas schrieb:
2005/11/29, Tomasz Chmielewski [EMAIL PROTECTED]:
you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card,
not HiSax (well, technically, you could use HiSax too, but avoid that if
possible).
I prefered to use hisax because it is already included in
Adam Goryachev wrote:
Don't assume that we read this list every 5 secs I haven't read the
mailing list since last week
You're right, thanks for your reply.
In any case, you have two options:
1) Do it with meetme like you do now...
Lee Howard, the author of IAXmodem agrees with
Hi Niklas
Thanks for this information I will be sure to follow it.
Many Thanks
Scott Pinhorne
Niklas Larsson wrote:
On Tue, 29 Nov 2005 06:14:54 +, scott wrote:
Is anyone using a vegastream product with asterisk? I have various
numbers coming into the vegastream vega400 and was after
Anyone know if this can be made to work?
I've only been able to get SIP-SIP call pickup to work.
Steve
---
as far as I know, no.
Il lun, 2004-07-05 alle 18:56, Adolfo R. Brandes ha scritto:
I've looked in the obvious places but haven't found a definitive
answer to the following:
Miguel Soto wrote:
Is the right behavior of the IAXmodem to display
Registration completed successfully and remote hangup many times?
You'd have to show me an example for me to say for certain, but my guess
is that if it looks wrong to you then it probably is wrong. This output
should
I am using this dialplan with DID's to great effect, I have 130 guys doing
exactly what was discussed here. After 12 seconds ringing their SIP or IAX
client, the dialplan calls the cell automatically, during working hours. If
they don't pick up after 18 seconds, voicemail. After hours, both
Steve Underwood also informed me about chan_fax
(http://www.sofaswitch.org/chan_fax/), I'll have a look.
This looks awesome please report back to the list on this if you get it
working correctly.
___
--Bandwidth and Colocation provided by
Hi Johann,
we engineered QueueMetrics out of the queues of * version 0.7, but never
found that origposition argument. And it's not present in our current 1.2.
Where did you find it?
Yours
l.
In data Tue, 29 Nov 2005 19:57:47 +0100, Johann
[EMAIL PROTECTED] ha scritto:
Entries in the
Thanks Colin, this is a fantastic list! All I need to do now is get my
butt in gear and set up the box(es)!
I am using this dialplan with DID's to great effect, I have 130 guys doing
exactly what was discussed here. After 12 seconds ringing their SIP or IAX
client, the dialplan calls the cell
Yes with version 1.2. I have tried already with call-limit and the same.
On 11/28/05, Kevin Hanson [EMAIL PROTECTED] wrote:
Alvaro Parres wrote: Hi list...I have been testing the hint extension. And i detect
that when i have in the sip.fg of the extension the incominiglimit=X (any number) the
Hello All,
I am using * 1.2, BRIstuff 0.3 PRE1, Dual HFC-PCI, 1x TE, 1x NT
I am using DECT phones on a Siemens ISDN phone/DECT-base.
My dial options are rTtWw, automon=*1, blindxfer=##
Whether I am calling (to my cell) or being called (from my cell), only the
caller can initiate recording or
Hi,
I`m a beginning Asterisk and Sendmail user. I am trying to setup my
voicemail to send emails to a certain email address. It doesn't work, and I
think I've figured out what it is. There is probably a spam-feature at my
provider (that I am using as smart host in sendmail) to not accept emails
1 - 100 of 162 matches
Mail list logo
