Hello,
we make XC-AST and can install it for you, or we can help you installing
it. How big is your call center? Under which environment did you try to
install it?
Thanks
l.
In data Fri, 10 Dec 2004 14:58:09 -0500, John Bittner [EMAIL PROTECTED] ha
scritto:
I have spent the last 3 days
if a phone number starts with one of 50+ prefixes,
i want to send the sip call to gateway X. if it
is in any other prefix, i want to send it to gate
Y.
i am not excited about a long list of extens,
but will do it if i have to.
i suspect there is a database hack, but i lose all
database
If somebody has done it before and has the time, please contact me off list.
The list is worthless if answers are sent by private mail.
ENUMLOOK=123
; Test ENUM lookup watching the CLI
; a file that says no enulm listing found
; was recorded
exten = _${ENUMLOOK}.,1,EnumLookup(${EXTEN:3})
On Sat, 2004-12-11 at 11:53 +1300, Matt Riddell wrote:
Hi,
I've just done up a quick proggy to show me the status of my IAX peers
from my windows box. It plugs into the simple manager proxy.
You can see more information (including a screenshot) at:
On Fri, 10 Dec 2004 21:53:53 -0600, nik martin wrote:
news.gmane.org wrote:
nik martin wrote:
Anyone ever thought about an Ethernet based channel bank? Basically a
rack mount set of 24 IAXys? That would be cool, IMO. No wrangling
with zaptel, etc. IAX as the * - Channel bank
Dnia pitek, 10 grudnia 2004 20:24, Peer Oliver Schmidt napisa:
Marco Parmeggiani schrieb:
I'm trying to use an hfc based pci card with asterisk but every call
fails falling in the congestion extension.
exten = _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr)
exten = _0.,2,Congestion
On Sun, 2004-12-12 at 00:00 +1300, Matt Riddell wrote:
Dave Cotton wrote:
http://www.sineapps.com/down/IAXPeers.zip
Could you please have a look and let me know your thoughts.
First I like it. I can use it straight away.
Cool, that's good to hear! :-)
Only comment at the moment
Is it possible to call a macro, which is defined in extensions.conf from
a realtime extension configured in Mysql.
Beacuse when i try i receive an error - no such context.
-- Executing Macro(SIP/1007-2165, dialnumber_wvm,1004,SIP/1004)
Dec 11 12:51:04 WARNING[22551]: app_macro.c:100
Wilson Pickett schrieb:
ENUMLOOK=123
; Test ENUM lookup watching the CLI
; a file that says no enulm listing found
; was recorded
exten = _${ENUMLOOK}.,1,EnumLookup(${EXTEN:3})
exten = _${ENUMLOOK}.,2,NoOp(ENUM result: ${ENUM})
exten = _${ENUMLOOK}.,3,Hangup
exten =
did asterisk 1.0.3 and chan_capi runs together ?
thx
nico
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Dave Cotton wrote:
But it's opened up another problem I can't have another instance of the
program monitoring another server through the VPN. Message is Run-time
error 380 Invalid property value if I change the ip address.
If you change the ip to what?
On my copy here, I can change the IP address
On 09/12/2004 at 09:22 Eric wrote:
Hi Sean,
Thanks for your reply, but that wasn't exactly what I was getting at.
I don't need to increase the system's imposed limit on the number of
open files. I'm more concerned to see if anyone has run across a
memory or fd leak in asterisk that sucks them
Randy Bush wrote:
if a phone number starts with one of 50+ prefixes,
i want to send the sip call to gateway X. if it
is in any other prefix, i want to send it to gate
Y.
Take a look at http://www.voip-info.org/wiki-Asterisk+app_dbodbc
I run a home server so I have never had the need to do
Title: Message
Does anyone know if
there is a "format-raw.c" routine available for
Asterisk-0.9.0?
Jim
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Around 1 customers.
On Fri, 10 Dec 2004 17:24:56 +0100, Wilson Pickett
[EMAIL PROTECTED] wrote:
How many customers, Sharon?
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I have this good looking IAXy device... I have managed to provision it,
i can see it registering to my asterisk box, however when I pick up the
phone which is plugged in the IAXy I have no dialtone, nothing.
What leds are lit?
What kind of phone is connected to it?
Can you call it? (watch the
What leds are lit?
Looking with the orange bit facing you, the network led on the left
(network) is permanently lit. The led on the right blinks once every 7
seconds or so. There is also the network plug's led which is lit. That's
all.
What kind of phone is connected to it?
France
Hi Everyone, Merry Christmas :-)
My Asterisk Box doesn't have a sound card, it is running
Asterisk 1.02
Zaptel 1.02
Libpri 1.02
Mpg123 0.59r
All compiled from source with kernel 2.6.9-1.6 on Fedora Core 2
Any help would be very much appreciated.
The error I am getting is
--
Not a European phone expert, but would that phone work on a US POTS
telephone network? Is the signalling and ringer voltage the same as US?
You're right to put that in question. I've had issues with older
Siemens phones (purchased in France) on both IAXy and Digium cards.
They don't ring at
Jean-Michel Hiver wrote:
What leds are lit?
Looking with the orange bit facing you, the network led on the left
(network) is permanently lit. The led on the right blinks once every 7
seconds or so. There is also the network plug's led which is lit. That's
all.
What kind of phone is
Hello,
I have a situation where I need to first check if a previous clip was
recorded, and if so, append to it.. Otherwise create a new file..
I'm using Monitor. Monitor automatically calls sox after the call ends.. Is
there a way to manually control this process, and instruct sox to append to
On Fri, 2004-12-10 at 16:29, Jim Guy wrote:
Hello,
I am just starting to research Asterisk and I would like to install it
on a PC to try out. I have looked around quite a bit but I haven't
found much information on the Linux part. I know you need to put Linux
on the PC first but what
I am sorry that I was not more clear. I am only looking for departments
that will
fit into the string:
press 1 for the DEPT department or press 1 for DEPT
the 'into' is what I should have been clearer about. I am only looking
for words that will fit into the DEPT portion of the above strings.
Hi.
I need to set up H323 on an Asterisk box. I've succesfuly compiled the
asterisk oh323 (including of course all the dependencies: PWlib and
OpenH323), and then compiled asterisk. However, asterisk doesn't report
a registered H323 channel (when it starts, it reports IAX2, ZAP and SIP
Wilson Pickett wrote:
Why do I get a MAILER DAEMON return for every message I post? Is there
something I need to change in my replies?
You'd probably be referring to Aster Risk.
Mr Risk has been returning messages for quite some time now. Maybe it's
been long enough for someone to remove him?
I
am using a card that has an fxo and fxs module.
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On Sun, 12 Dec 2004 04:36:56 +1300, Matt Riddell
[EMAIL PROTECTED] wrote:
Wilson Pickett wrote:
Mr Risk has been returning messages for quite some time now. Maybe it's
been long enough for someone to remove him?
Maybe its time for mailing list moderators? (just throwing that out there :))
I have sent this twice now but, I think, for some reason, it has been
sent as HTML which is causing it to be drooped (and rightly so). I
apologize in advance if, suddenly, those two make it though along with
this one.
Anyway, I should have been more clear in my original message. I am
looking
Im having a problem where I've just switched from
static configs to "realtime" configs stored in mysql
It's all working fine (in terms of it reading the
configs and loading them as it should), except my problem is that if a user
changes there voicemail password via the "Advanced Options
Hi,
I'd like to setup little private enum server. Any more info on how to do
that ?
Regards,
Rob.
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When I make a call from a SIP phone to a speaking extension on *, such
as one that speaks digits or similar, when I monitor * in very verbose
mode I can see it running through the routine associated with the
extension, but I am getting no RTP data stream back to the phone.
Does the machine
We developed IVR machines for a long time (using Dialogic and our own
code)
In order to be able to get the most of prerecorded prompts, you need to
have a folder general sounds (numbers - 1-20, 30-90, 100-900,
1000-9000 and so on, month names, dept. names, etc.)
Then, complete sentences can be
On Sun, 2004-12-12 at 03:46, Eric Wieling aka ManxPower wrote:
Howard Lowndes wrote:
When I make a call from a SIP phone to a speaking extension on *, such
as one that speaks digits or similar, when I monitor * in very verbose
mode I can see it running through the routine associated with
Hi,
I am looking at getting adtran tsu 600 p/n 1200.076L2 for my small office
It comes with 6 FXS ports and I would use 2 X100Ps for FXO ports.
Would that work ? Is there anything I would have to be aware of in such configuration?
What would be a better solution?
robert
On Fri, 2004-12-10 at 20:02, Dealer Backup Admin wrote:
Received errors as follows.
snip
Are you using version 1.0 or CVS HEAD? The patch will probably only
apply cleanly on the 1.0 branch.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
[EMAIL PROTECTED] wrote:
Its not a moderator issue, it is a bounce issue, Mailman can be setup to
deal with this. However, if this guy bounces messages, just remove him
from the list.
He's not bouncing them to the list. He (well, his MTA) is bouncing them
to the original sender, so the mailing
I have been able to upgrade my Cisco 7905G phones to the SIP Image without
any problems, but I just got a 7960, and I can't seem to get to the settings
so I can upgrade to a SIP Image.
When the phone boots up, it says Configuring VLAN, Configuring IP, TFTP
..., then Protocol Application
Randy MacKay wrote:
I have been able to upgrade my Cisco 7905G phones to the SIP Image without
any problems, but I just got a 7960, and I can't seem to get to the settings
so I can upgrade to a SIP Image.
When the phone boots up, it says Configuring VLAN, Configuring IP, TFTP
..., then
if a phone number starts with one of 50+ prefixes,
i want to send the sip call to gateway X. if it
is in any other prefix, i want to send it to gate
Y.
Take a look at http://www.voip-info.org/wiki-Asterisk+app_dbodbc
too big a hammer. i finally did the agi hack. for the archive
Thanks, but there is no zaptel file in /etc/init.d/ I'm using White Box
Linux, which is derived from RHEL 3. Kernel is 2.4.x
Did you run make config for zaptel?
If not do the following;
cd /usr/src/zaptel
make config
- Jose
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Asterisk-Users
I had this a couple of days ago ..
Randy MacKay wrote:
I have been able to upgrade my Cisco 7905G phones to the SIP Image without
any problems, but I just got a 7960, and I can't seem to get to the settings
so I can upgrade to a SIP Image.
When the phone boots up, it says Configuring VLAN,
Hello :
Have a Tormenta 2 PCI card - Quad E1.
When I try to modprobe tor2, the following errors are displayed :
/lib/modules/2.4.20-8smp/misc/tor2.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters, including
invalid IO or IRQ parameters.
You
I am sorry to ask such a simple questions.
I have been using Asterisk successfully for well over a year
now on three servers. I was using CVS HEAD, and the last
time I updated was sometime back in July.
I decided to switch to the recent stable 1.0.3. I built
zaptel, libpri and asterisk, and
On Sat, 11 Dec 2004 10:56:53 -0800, Jose Hernandez
[EMAIL PROTECTED] wrote:
Thanks, but there is no zaptel file in /etc/init.d/ I'm using White Box
Linux, which is derived from RHEL 3. Kernel is 2.4.x
Did you run make config for zaptel?
If not do the following;
cd /usr/src/zaptel
make
Sorry if this comes in twice. Wasn't subscribed
first time :-(
Anyone help me here..
It worked once :-(
I have a static IP address which is on my private
network.. Phone is 192.192.192.220 and asterisk server is
192.192.192.22
I have the 7690 with a SIP iamge (Whatever latest
is )
My problem with this phone is I cannot get to the settings to change
anything. This is a used phone, but new to me. I have not had it in
service yet.
None of the buttons on the phone seem to do anything. I assume I have to
configure the phone TFTP settings so I can upgrade to the SIP Image and
Robert Rozman wrote:
I'd like to setup little private enum server. Any more info on how to do
that ?
You just need bind or any other name server that supports NAPTR records
and to setup a zone with NAPTR records...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
Your previous messages came through, but had [Asterisk-Users] Re:
Asterisk-Users Digest, Vol 5, Issue 158 as the subject.
I for one usually skip messages where the person did not think to change
the digest subject to something more meaningfully.
To help others help you could those of you who
Hi,
We're installing an * box next week (pbxtra from fonality) and I'm
trying to come up with a solution for remote users that want a phone in
their home. I need VPN and QoS capability, wireless support would be a
nice to have. Ethernet handoff is fine, i don't need integrated dsl or
cable
One more thing about prompts, it's better to say for sales press 5 than
press 5 for sales, because by the time you hear sales you've already
forgotten what number it was.
So record for sales press and the digits (you could use the digits that
come with *, but a sentence in two voices sounds very
To everybody on this wonderful group.
I am considering to use an asterisk PBX with 70 telephones and I would like
to play background music trough the speakers of the telephones. When the
hook is lifted, the background music stops till the phone is on hook again.
Is this possible? Or can it easily
Warren Burstein wrote:
One more thing about prompts, it's better to say for sales press 5
than press 5 for sales, because by the time you hear sales you've
already forgotten what number it was.
If you add the sounds all you need is For Sales recorded the new sounds have
press # already. So you
Personally I find the ATA adapters to be the most versatile, your mileage
may vary though. When you need more extensions you just buy more ATA's, no
need to tear up the * box or take it down etc.
Buying IP phones is OK but you are limited to IP Phones only. With the ATA's
you can buy ANY phone
For reasons unknown
to me, SER and subsequently a Microsoft Live Communcations Server 2005 seems to
have problems, matching a SIP ACK request from asterisk to the ongoing SIP
transaction, I have attached the complete log, but the essential lines
are:
13(2894) DEBUG:
RFC3261 transaction
First of all, I generally skip messages that have the entire digest
subject as well. I am always thinking that it was somebody who has left
the entire digest in their reply. Sorry I missed the subject in my
messages.
Second, currently the plan is to have Allison (the same person who
recorded
What is the secret to getting * to record messages or voicemail.
It goes thru the process but the file created is zero length.
I don't think it can be a perm problem as I am running * as root - maybe
not a good idea but I am only testing at this stage.
Do I need a sound card in the * box? - I
Hi Clive,
if you have openssl already installed, make sure that you also have
libssl-dev installed (development package for ssl, I don't know it is is
called in distributions other than Debian).
You probably have already found your solution, but I am answering this so
it appears in the list
Original Message
From: Ariel Batista
I have Spandsp working fine. Asterisk sees a fax on the zap port and
redirects the call to the fax-in area.
This works if I have a simple dialing rules that goes answers first
and waits 10 secs then goes to the next item. If it hears a fax it
goes
My problem with this phone is I cannot get to the settings to change
anything. This is a used phone, but new to me. I have not had it in
service yet.
None of the buttons on the phone seem to do anything. I assume I have to
configure the phone TFTP settings so I can upgrade to the SIP
Hi,
I'd like to setup 20 BT-100 with Asterisk.
If I got all discussion on grandstreams right, I should put my own tftp
server and point phones to it. On phones is 1.0.5.16 firmware.
Is this one good or should I up(down) grade to certain version ? What
functionality is possible with BT-100 ?
I do have the 7960 SIP Image, but I can't get into the phone to change the
TFTP. Is the phone locked? Did the previous owner mess up an upgrade and
now the phone is a paper weight?
First yes you can unlock the settings on the phone, the default password is
cisco or if you have older
Hello everybody,
After going through multiple posts on Internet and trying different things I can't seems get zaptel wcfxo loaded on the server correctly.
I run dual P3 500 Mhz box. After reading README.udev
I put a file 60-zaptel.rules /etc/udev/rules.d
# Section for zaptel device
Hi List,
I have this good looking IAXy device... I have managed to provision it,
i can see it registering to my asterisk box, however when I pick up the
phone which is plugged in the IAXy I have no dialtone, nothing.
Any ideas what might be going on?
Cheers,
Jean-Michel.
Dave Cotton wrote:
http://www.sineapps.com/down/IAXPeers.zip
Could you please have a look and let me know your thoughts.
First I like it. I can use it straight away.
Cool, that's good to hear! :-)
Only comment at the moment is would it be possible to save the
configuration of the Host and the
Michael Vogel wrote:
I dumped some sample enum configs to http://www.asterisk.net.au/tutorial/7/
and more on e164.org
Are there any test numbers where I can see if ENUM lookup is working?
18005558355 (1800 555 tell, news and weather service etc)
And is it possible as well to test if a number of a
I found the problem !
In the for appdata I have - dialnumber_wvm,1004,SIP/1004
it must be - dialnumber_wvm|1004|SIP/1004
Damian Minkov wrote:
Is it possible to call a macro, which is defined in extensions.conf
from a realtime extension configured in Mysql.
Beacuse when i try i receive an error -
Just turned up a new PRI with DID's in the US. I'm receiving 5 digits
of the DID numbers as I requested.
Assuming I have 100 DID numbers but only define 50 of those in
extensions.conf, is there an easy way to send the incoming calls
for the 20 undefined numbers to a common resource
Why do I get a MAILER DAEMON return for every message I post? Is there
something I need to change in my replies?
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This is the second time that i configure a * box using
a E100P card. The only difference at this time is
that i´m using another Telco and the box have more
one card ( Wildcard - 2 FXS + 2 FXO ).
Well , everything looks fine i don´t have any kind of
alarms on my zttool , the board gives me a
On December 10, 2004 06:26 pm, Rich Adamson wrote:
Assuming I have 100 DID numbers but only define 50 of those in
extensions.conf, is there an easy way to send the incoming calls
for the 20 undefined numbers to a common resource (ivr, operator,
or canned message) without having to define each
I think he means you need multiple lines on the polycom phone in order
to use it's conference.
On Fri, 10 Dec 2004 15:38:35 -1000, Richard [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Wieling
Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisa:
Hi.
I need to set up H323 on an Asterisk box. I've succesfuly compiled the
asterisk oh323 (including of course all the dependencies: PWlib and
OpenH323), and then compiled asterisk. However, asterisk doesn't report
a registered H323
hello are you sure that you have loaded the module in the modules.conf
files?
load = chan_oh323.so
K.
- Original Message -
From: Rodolfo Grave [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Saturday, December 11, 2004 3:32 PM
Howard Lowndes wrote:
When I make a call from a SIP phone to a speaking extension on *, such
as one that speaks digits or similar, when I monitor * in very verbose
mode I can see it running through the routine associated with the
extension, but I am getting no RTP data stream back to the phone.
Hi , I have this schenario:
SIPUAs---SER---asterisk(sip-h323)GNUGK(RadiusBilling)-h323Clients
|
|
Anybody know of good reliable Asterisk compatible DIDs in the 636
(Missouri, USA) area code?
Voicepulse doesn't go there, and Broadvoice seems unreliable in my Asterisk
installation -- so I'm reluctant to recommend it.
Thanks,
/edg
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format_sln.c is what you want. It should compile on 0.9.0 but WHY are you
using such an old version?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jim O'Brien
Sent: Saturday, December 11, 2004 6:39 AM
To: [EMAIL PROTECTED]
Its not a moderator issue, it is a bounce issue, Mailman can be setup to
deal with this. However, if this guy bounces messages, just remove him
from the list.
On Sat, 11 Dec 2004, Leif Madsen wrote:
On Sun, 12 Dec 2004 04:36:56 +1300, Matt Riddell
[EMAIL PROTECTED] wrote:
Wilson Pickett
I have three remote offices with VPN's into my main office.
Two Offices use the IAXy's with ATT 958 Phones (Functional and inexpensive
$30, does not use the vpn, but I have a port opened to it). The IAXy's are
easy to set up (no NAT to worry about), low bandwidth with the IAX.
The third office
I had a Grandstream 286 at my home hitting my Asterisk box at the office,
all worked well and I received phone calls fine until the device just up and
died.
I replaced this unit with an SPA-2000 because I have been impressed with the
Sipura devices and decided to use them for most of my needs
Hi All,
Asterisk newbie here. Have recently got my system working with Xten
softphones and now want to expand to real ones. Was considering Grandstream
but am concerned about possible RFI problems with them (after reading these
archives for the past few weeks), because I plan to deploy using a
Hi,
I currently have a * server with a IAXy adapter and a Voip phone. My
doubt is: which is the best option? I personally find IAXy to be very
effective, except from the fact that they don't support G729. The other
option would be to use the TDM400P, which I have heard that it has some
Charles S. Antrim wrote:
I am using a card that has an fxo
and fxs module.
I am no where near
an expert but I have my sip phone working through my pstn line and this is my
config.
/etc/asterisk/sip.conf
[general]port =
5060bindaddr = 192.168.69.1context = sipdisallow = gsmallow =
--On Thursday, December 09, 2004 19:19 -0700 Joseph [EMAIL PROTECTED]
wrote:
On Thu, 2004-12-09 at 18:11 -0800, Lee Howard wrote:
On 2004.12.09 17:56 Joseph wrote:
At time to time I receive some junk faxes from some advertising
companies that play smart and don't provide any TSI number so I
I'm testing a TDM400P with FXO module to receive incoming calls from an
analogue line and send it to a SIP device.
To recieve callerid, I need to use cidsignalling=dtmf and cidstart=polarity.
The problem is that when a call is finished, the TDM400P seems to require
about 20 seconds to prepare
Answering my own question - after a few more hours
googling, the way to prevent users changing there voicemail password via the
voicemal "advanced options (0) menu", is described at:
http://bugs.digium.com/bug_view_page.php?bug_id=0002386
All thats required is to preceed voicemail pins
How can I capture a -1 result on a Dial command? Basically, I have the
following setup, and I want to be able to process the audio file after the
outbound call has been done regardless how how it ends.
No matter how the call ends, I can't get macro-record-stop to run.
Any help would be great.
Hi,
I'd like to make small 20 users setup with BTs. I'd like each of them to
have its own context (for recording prompts, conference, ...). For them to
have same extensions I should put them in separate contexts and let BT call
them offhook. But these contexts are pretty similar (for instance
On Fri, 2004-12-10 at 18:31, James Andrewartha wrote:
Hi,
Are there any resellers of phones that can take power over ethernet in
Australia? All I can find for sale online is the BT-10[12], which is cheap
but not featureful enough, and the Snom 190, which is about right, but
neither of
Hi, all.
On a lark, I have gotten Asterisk 1.0.2 (from Debian testing) up and
running, but I have found one problem when I try to use it with Free
World Dialup: Dialing doesn't work properly.
I have everything else working; it receives calls, ringing my phone and
everything, but when I try to
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