On Sun, 29 Aug 2004, Kris Boutilier wrote:
Is timestamp information calculated purely from the relative timestamps of
each frame of the current incoming stream or is there some degree of RTC
synchronization expected between the two endpoints?
No sync is needed; its all relative.
On Tue, 31 Aug 2004, Storm D. J. Petersen wrote:
I have a problem with jitter over a 2mb up 1mb down satellite connection. I
call my friend over the satellite - I call perfect but they cannot make out
a word I say. However if I leave him voicemail on his asterisk box, it
records my voice
I downloaded the astcc calling card program. Thanks, it is very easy to
setup and works Excellent. Anyway, it says to use DeadAGI to run it
rather than AGI. I don't know what I am doing wrong. I just updated my
asterisk from cvs and rebuilt and reinstalled. I do not have an
application
I tried to send sms messages the other day from a * box connected to a E1
line (BT ISDN30).
Message never arrived, however, I was soon called back on the E1 by an
automated BT system which sent a message stating that you cannot send sms
messages on this line
Is there anything I need to do before
Does anyone know if this phone can or does work with Asterisk ? I have a
potential client who wants to throw away their current mitel ip system
(keeps crashing on a regular basis - not good for a phone-based company),
but are reluctant to purchase new handsets.
Thanks.
Julian.
You need to make sure you have caller ID enabled on the line, as SMS
relies on this to make it work.
Rgds
Tim
Asterisk wrote:
I tried to send sms messages the other day from a * box connected to a E1
line (BT ISDN30).
Message never arrived, however, I was soon called back on the E1 by an
Hi,
-Original Message-
I downloaded the astcc calling card program. Thanks, it is
very easy to
setup and works Excellent. Anyway, it says to use DeadAGI to run it
rather than AGI. I don't know what I am doing wrong. I just
updated my
asterisk from cvs and rebuilt and
Darren Wiebe [EMAIL PROTECTED] writes:
I downloaded the astcc calling card program. Thanks, it is very easy
to setup and works Excellent. Anyway, it says to use DeadAGI to run
it rather than AGI. I don't know what I am doing wrong. I just
updated my asterisk from cvs and rebuilt and
Thanks, but that was something I'd already checked. If I make a call out
from that line to my mobile, then the number comes up as expected.
Julian.
- Original Message -
From: Tim Robinson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Kevin,
Thanks; from what I have read on other Broadvoice threads, that has to do with
comfort noise generation.. more of an asterisk issue than anything else.
As a followup, I did get a response from broadvoice after posting to this
forum indicating that they are checking with ther billing
Hi
I could never get this to work the way I wanted, I then found in the notes about half
way down on this page in the wiki
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sip.conf a reference to
that you really have to listen to hear the difference and that Cisco are working on
Actually, he meant that you need to have caller ID presentation turned on on
the line so that when people call you, you can see their caller ID.
However, I don't believe that that is a requirement for the sending side to
work, just for the receiving side.
I know for a while BT did say at fixed
I met with the BT SMS guys a while back and they told me that they were
going to keep SMS on all PSTN services, i.e. ISDN30, ISDN2/Highway and
analogue.
Do you have inbound CLI enabled on your E1? i.e. can you se inbound
caller ID?
Rgds
Tim
Linus Surguy wrote:
Actually, he meant that you need
Sorry, I meant to say that I also had caller id presentation available as
well.
It's probably the isdn30 that's causing the issue. Do you know if it works
on a standard ISDN line ?
Julian
- Original Message -
From: Linus Surguy [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Hmm. Yes, I do have inbound cli - I can see the inbound callerid.
Julian
- Original Message -
From: Tim Robinson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, September 01, 2004 8:45 AM
Subject: Re: [Asterisk-Users] SMS
has anyone the capiCD() funktion in chan_capi running?
for me it does not do calldeflection.
the capi is messageing that CD is supported.
nico
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To
Hi,
I was wondering if anyone knows where i can find out the standards
used by telco's in different countries... and how to configure asterisk
to support them. Secondly, whenever i try Dialing a zap channel, all i
get is no sound on the phone source, and noise on the destination line.
On Wed, 01 Sep 2004 13:07:19 +0500, Imran Akbar [EMAIL PROTECTED] wrote:
I was wondering if anyone knows where i can find out the standards
used by telco's in different countries...
try this ...
http://artofhacking.com/files/callerid/index1.htm
and how to configure asterisk to support
Hi,
I have a question regarding X100P card.
I have one X100P card in an * box.
I have the telco line connected to the line port of the X100P card, and an
analog phone connected to the phone port of the X100P card.
My question is:
How to make ringing the analog phone connected to the
On Tue, 31 Aug 2004 19:01:06 -0500, Christopher L. Wade
[EMAIL PROTECTED] wrote:
Hi all,
Has anyone gotten custom ring tones to work using ALERT_INFO with the
Cisco 7940 SIP phone? I've read the wiki, but just can't get this to
work. I'm currently using the 7.2 SIP image.
It works just
Gilbert,
The phone port is only a loop thru port for the analogue line.
It is not an FXS port.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: 01 September 2004 09:32
To: Asterisk-Users; Asterisk-Dev-Admin
Subject: [Asterisk-Users]
I think there is a bit of confution
over the term 'ringtone' I started out thinking that I could use
Alert_Info to define a different ringtone that I had setup on the tftp
server - ie it was the same as going to the phone and changing the ringing
settings manually.
This is not the case and the
GIBERT Frédéric wrote:
I have one X100P card in an * box.
I have the telco line connected to the line port of the X100P card, and an
analog phone connected to the phone port of the X100P card.
My question is:
How to make ringing the analog phone connected to the phone port when you
receive a VoIP
Luis Vazquez wrote:
Does anybody knows if it's posible or if there is some develoment in
course to be able to use longer transmit packet sizes (as long as I know
this is fixed in 20ms now) with the compressed voip codecs in asterisk
(g729, g726, gsm, etc).
I need to use asterisk to connect
Hi,
I have an asterisk box connected to a PRI LINE, some extensions are
trunked by IAX to another box that's connected via ISDN BRI to a PBX.
That's what's happening
call comes in via PRI to the first box and is sent to the other box
exten = _N.,1,Dial(IAX2/sip:[EMAIL PROTECTED]/*${EXTEN})
Hi all,
I have any information more. I have configured sip.conf with
bindaddr=0.0.0.0. I have observed traffic with tethereal and I have seen
the next. First REGISTER goes out from my asterisk to my SIP Provider.
My SIP Provider respond to my with a 401 Unauthorized meesage, but
Asterisk
On Wednesday 01 September 2004 02:16, [EMAIL PROTECTED] wrote:
It sounds like you just don't have enough throughput in the one direction.
Voicemail is fine because it doesn't need realtime capacty - the voice
frames arriving from your side go into the captured file as they arrive,
doesn't
I've read the wiki and other resources on how to connect Vonage / Voicepulse
and all these other services to Asterisk... We are attempting a connection
to a Lucent iMerge. Lucent has told us that it won't work - but we feel
confident that it will. Has anyone worked with the Lucent iMerge - or
i'm new here and i need help on how where can i get
software version 4.0.x of the mediatrix and how can i
install it...
mediatrix unit im using has a software version of
2.4.9.57. i would like to use H.323 not SIP...
please need help asap!... hope to hear from anyone of
you soon..
I am hooking up to a DMS500 (100250 together) and wanted to see if
anyone had any experience with this. We have the GR-303 span up, the
IDT is built.
I have not yet heard of anyone doing this, but would be _extremely_
interested in your experiences. Please keep in touch with me
Hello!
I have asterisk updated from CVS on 31/8/2004 with
sample configuration. I have just changed the
sip.conf to register asterisk with sip proxy in out
intranet.
Then I can successfully make call to asterisk and go
to demo IVR, but no response to dtmfs.
I try to make call from several sip
Is this the 3300IP system? Those systems *can* be quite good, but
need to be up to the latest revision and require a bit of tuning. I'd
be interested in your finds regardless and could probably experiment
with a 5005, 5010 or even the higher-end ones.
I'll ask a former co-worker if I can borrow
A customer of mine has 3 TDM400P cards in a box running asterisk. On
each card he has four FXO modules.
I have set up the dialplan to dial via group 1 for an outgoing call.
Channels 1-12 are in group 1.
If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when
he dials
Hi all,
Did anyone manage to make the GotoIf command work with regular expression ?
I would like to make the following thing:
${DTMSeq} contains a menu choice. Only 1, 2 and 3 are allowed menu choices.
If
DTMFSeq contains 1 or 2 or 3 = OK, Goto 4
else
Goto 2
I've tried the
Hi all,
Did anyone manage to make the GotoIf command work with regular expression ?
I would like to make the following thing:
${DTMSeq} contains a menu choice. Only 1, 2 and 3 are allowed menu choices.
If
DTMFSeq contains 1 or 2 or 3 = OK, Goto 4
else
Goto 2
I've tried the
I left my phone at home I think Im using sip 1.3.1.. It's 1.3.
something
Asterisk CVS-HEAD-05/12/04-13:23:20, Copyright (C) 1999-2004 Digium.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent
Franks
Sent: Tuesday, August 31, 2004 10:14 PM
To:
Hi all,
Did anyone manage to make the GotoIf command work with regular expression ?
I would like to make the following thing:
${DTMSeq} contains a menu choice. Only 1, 2 and 3 are allowed menu choices.
If
DTMFSeq contains 1 or 2 or 3 = OK, Goto 4
else
Goto 2
I've tried the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Lyle,
Lyle Giese wrote:
| The standard for loop start does not send answer supervision, so * and all
| other telcom devices that do CDR records have to 'assume' that the call was
| answered.
So, it means that is impossible to say if the call was
Yeah, I took care of that. The only thing I can't do as of now is get it
to set a default ringer and get the phone to display name and number
while rining...
ringType
DEFAULT se.rt.modification.enabled=1 se.rt.enabled=1
se.rt.1.name=High Double Trill se.rt.1.type=ring se.rt.1.ringer=7
On Wednesday 01 September 2004 01:15, [EMAIL PROTECTED] wrote:
They would like to be able to unplug lines and use them for other
purposes at times.
Out of curiosity, why are they unplugging the lines? i.e. what are these
other purposes ?
-A.
___
Some telcos execute a line polarity reversal to indicate Answer
Supervision. This might be detectable on the TDM400P cards. ISTR that
Mark said this was available although not implemented.
Other exchanges will sometimes send meter charge pulsing at 50Hz
longitudinal or 12 or 16 Khz. These
I am very surprised that you have had
problems with a mitel 3300ICP. We have had one installed here for the past
year or so and it is a very good system. We did suffer the occaisonal crash
and then our system was wiped out by a lightning strike, we had a replacement
controller in place and fully
Hi all,
Did anyone manage to make the GotoIf command work with regular expression ?
I would like to make the following thing:
${DTMSeq} contains a menu choice. Only 1, 2 and 3 are allowed menu choices.
If
DTMFSeq contains 1 or 2 or 3 = OK, Goto 4
else
Goto 2
I've tried the
Selim [EMAIL PROTECTED] wrote:
Did anyone manage to make the GotoIf command work with regular expression
Does re-posting the same question lots of times usually work for you?
I suppose it generated a response in this case. I usually just delete
duplicates, along with the original.
I've
I'm using SIP 1.3.1 with Boot RPM 2.50 and so far everything's running great.
I don't use # transfer though, so haven't tried that. I use the softkeys
instead to transfer.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew Marlowe
Sent:
I'm using the same SIP version, everything is running great except as I've said before
that setting the default ring type isn't working and incoming calls only displays name
and not name and number..
From: [EMAIL PROTECTED] on behalf of Reid A. Forrest
Sent:
Hi
everyone,
I want to have a
group and dial multiple phones/lines simultaneously. If I use this Dial
command:
exten =
222,2,Dial(${TRUNKBP}/246SIP/258${TRUNKBP}/243,20,tTr)
... all phones
ring just once, after that only the first one continues ringing and only that
one can answer. Can
Ditto
that. On a Mitel box, A Service Contract Is Your Friend and a Good Interconnect
Is Your Best Friend. Our 3300 was loosing conn w/ the phones and loosing sync
with our PRI. Our interconnect blamed everything from the LAN to sunspots except
the 3300. Because we had a service contract, I
Hi Julian-
I was using a BT BRI line, with caller ID option enabled. Also, I had to
send 1470 before the call because my customer had blocked his outgoing
number on this line.
So I'm certain that it works on BRI. BT says in their SIN document
(Supplier's Information Note), number 413, that
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi List,
I'm using the apllication AgentCallBackLogin so agents can login to a
queue. They just need to
enter the password, the CallBack Extensions is the ${CALLERIDNUM}
Is there a way to AgentsLogOff withou using the AgentCallBackLogin
application. I
It may work if you can make the iMerge register with Asterisk, currently
( to the extent of my knowledge ) mgcp can only act as the registrar.
On Wed, 2004-09-01 at 06:59, Huddleston, Robert wrote:
I've read the wiki and other resources on how to connect Vonage / Voicepulse
and all these other
Sorry for the re-posting, I received a delivery failure mail for each
one of my emails ...
Thank you Kevin for your Clues.
Does any one else have used regex (or extended regex) with GotoIf ?
Selim
On Wed, 1 Sep 2004 14:34:49 +0100, Kevin Walsh [EMAIL PROTECTED] wrote:
Selim [EMAIL PROTECTED]
Tomica Crnek [EMAIL PROTECTED] wrote:
(Article auto-converted from unnecessary HTML to nice plain text.)
I want to have a group and dial multiple phones/lines simultaneously. If
I use this Dial command:
exten = 222,2,Dial(${TRUNKBP}/246SIP/258${TRUNKBP}/243,20,tTr)
... all phones ring
Fair enough, The MItel is not the most
expensive VOIP solution, but as you say - the reason Asterisk is so attractive
is its effectively free and you are not locked to a single vendor or support
channel.
I have only just started looking at
Asterisk, I like it a lot, particularly its openness -
Does anyone know where to obtain the 2.5 / 1.3.1 bootrom/app? My
account on polycom's site keeps pointing me at documentation only.
Regards,
-Steve
On Sep 1, 2004, at 10:00 AM, Matthew Marlowe wrote:
I'm using the same SIP version, everything is running great except as
I've said before that
Hi,
need a quick help ... it should be easy but ...
exten =_9898,1,Answer
exten =_9898,2,VoiceMailMain([EMAIL PROTECTED])
Accepting overlap call from '342' to '9' on channel 0/2, span 3
-- Executing Answer(Zap/8-1, ) in new stack
-- Executing VoiceMailMain(Zap/8-1, @domain) in new
I will get a packet sniffer on one
in a minute
Don't bother, won't work. I already tried. Spoke to someMitel
mucky-mucks too, and they said nope. You have to get Mitel's SIP-specific
phone which*is* a 5220 that's been reflashed. Unfortunately, I have 80
5020's which
Oh great. BT at it's best.
Spoken to 4 different product managers / isdn help desk / customer service
weenies. Each one says Eh? What? and I've got to point them to your link.
I'll get back to you on that is the next response.
Still waiting
But thanks for the link - without that I would
The quick and dirty way:
In rtp.c, function ast_rtp_write, in the switch statement,
AST_FORMAT_G729A case, change the smoother creation to something
larger. E.g.:
rtp-smoother = ast_smoother_new(40);
Keep in mind that you must set this into something valid
(45
Hello
I's trying to install an asterisk-server with a wildcard e100p and get
some errors while starting asterisk:
Sep 1 18:49:34 WARNING[1024]: chan_zap.c:724 zt_open: Unable to specify
channel 1: No such device or address
Sep 1 18:49:34 ERROR[1024]: chan_zap.c:5883 mkintf: Unable to open
http://www.freedomphones.net/polycom/files/
Best regards,
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven Kokinos
Sent: Wednesday, September 01, 2004 11:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Any got experience w/ PWLIB - sorry I know it's somewhat off topic...
I do not have a bison.simple file located on Fedora RC2...
But when make'ing PWLIB I get
../common/getdate.y:106:1: warning: YYPURE redefined
../common/getdate.tab.c:43:1: warning: this is the location of the previous
I'm not sure what the problem is with my source. I'm running the latest
copy, as of this morning, from cvs. DeadAGI will not build on my server
but it will build on my laptop. I'll have to have a look at my source.
I just copied the res_agi.so file across this morning and everything
works
Andrew Kohlsmith wrote:
On Tuesday 31 August 2004 17:36, Kevin Walsh wrote:
Spam-dialling should be made illegal. I, for one, wouldn't spend two
seconds adding features to support this sort of usage.
I can think of at least one legitimate use for this -- reverse spam dialling,
or at least real
Hi,
I have any information more, I have noticed that asterisk
receives 401 Unauthorized message but If I do a sip denbug I can read
next:
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK6a231e4d
From: sip:[EMAIL PROTECTED];tag=as0f12fef4
To:
ok - just spoke to some (non-tech) at
mitel...apparently they are going to release a firmware image for the 5220
that is dual boot (Minet and SIP). One of their techs is gving me a call
back today or tomorrow (hopefully) with more details fingers crossed
it will be a downloadable item and
Huddleston, Robert wrote:
Any got experience w/ PWLIB - sorry I know it's somewhat off topic...
I do not have a bison.simple file located on Fedora RC2...
But when make'ing PWLIB I get
../common/getdate.y:106:1: warning: YYPURE redefined
../common/getdate.tab.c:43:1: warning: this is the
Sergio Serrano [EMAIL PROTECTED] wrote:
SIP Provider---ADSL router---localnet
192.168.20.0---ASTERISK---localnet 172.24.240.0---softphones
first localnet 192.168.20.0
second localnet 172.28.240.0
in second localnet we have softphone and the first localnet is
connected to ADSL
We intend to use Asterisk with a very large dialplan (with a lot of
functionality for 3000+ users). Each user will be able to change several of
his parameters in the dialplan, so we will be forced to reload the diaplan
constantly. Has anybody else any previous experience with a similar
Hello everyone,
I am new to Linux, some help with the following would really
beappreciated :
1.How can I load asterisk automatically in Linux each time the
machineboots up (like autoexec.bat in windows)2.how I can shut down
and restart asterisk automatically every night?
Thanks
San
Look
at the /etc/rc.d and init.d directories... Unix has run levels and there are
shell scripts (like batch files) that are called upon system
booting..
As far
as nightly restart - you need to use cron or atd (at daemon) these
processes allow you to schedule scripts to run...
Otherwise
On Wednesday 01 September 2004 13:41, San Singhania wrote:
1.How can I load asterisk automatically in Linux each time the machine
boots up (like autoexec.bat in windows) 2.how I can shut down and restart
asterisk automatically every night?
I use this:
su - root -c '/usr/bin/screen -d -m
Hi all, I'm having sudden MWI problems. Everything else on the phone
works fine though.
I have three Cisco 7940s.
Asterisk server is behind a firewall running NAT. (192.168.1.202/24)
Phone #1 - On the same subnet 192.168.1.250. Everything works great.
Phone #2 - On a different subnet,
Hello list,
I've posted my problem on BSD list and i still have the
problem.
The remote side receives the call , but there's no voice
on the call.
I tried everything about possible NAT problems ..
but ther're on same net.
My platform:
FreeBSD 5.2.1-Release
Asterisk 1.0-RC2
soft phones : X-Lite
--
I need to know how to setup the data side of the T1 on my Linux Box. I
have found information about configuring a PRI and HDLC but nothing
about the Frame-Relay type setup for data.
Ok, firstly HDLC is just a Layer 2 protocol like IP, so no matter what
encapsulation they use it's still HDLC.
Ok, firstly HDLC is just a Layer 2 protocol like IP, so no matter what
encapsulation they use it's still HDLC.
I meant Ethernet/ARP IP is at layer 3 DUH...
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Asterisk-Users mailing list
[EMAIL PROTECTED]
Does anyone have disconnect supervision working on their TDM400P or X101P
cards (to/from the telco)?
Googling telco disconnect supervision, I found the following description:
http://mirror.lcs.mit.edu/telecom-archives/telecom-archives/TELECOM_Digest_Online/1559.html
[cut]
Re: How Do I Get
On Wed, 2004-09-01 at 12:26, Glen Johnson wrote:
Does anyone have disconnect supervision working on their TDM400P or X101P
cards (to/from the telco)?
Yes on all 4 my X100P/TDM400PFXO ports across 3 servers in two states.
It Just Works. Of course I'm in the USA.
--
Eric Wieling *
It seems voicemail recordings have broken sound. It
cuts out randomly throughout the recording. Has anyone had any similar experiences?
Ive included some snips of my voicemail.conf
Cheers,
Ben
--SNIP---
[general]
; Default formats for writing Voicemail
Fails on loading several of the chan_*.so modules with undefined symbol
__use_ast_pthread_create_instead__.
Notably, these same modules complain during compilation implicit
declaration of function __use_ast_pthread_create_instead__.
Ideas?
--
Alok K. Dhir [EMAIL PROTECTED]
Symplicity
Make sure you delete your /usr/lib/asterisk directory before installing a
new CVS copy...
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I did...
Chris Shaw wrote:
Make sure you delete your /usr/lib/asterisk directory before installing a
new CVS copy...
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Hmmm... Well what that means is that the code is using pthread_create()
instead of ast_pthread_create(), it's not a major thing, all you would have
to do is go through all the affected modules and replace pthread_create with
ast_pthread_create, but this should probably be fixed in CVS too!
Title: Message
What
is your definition of TRUNKBP ?
It is
probably because that channel is being answered first
Rgds
Tim
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomica
CrnekSent: 01 September 2004 15:19To:
[EMAIL
On Sep 1, 2004, at 11:35 AM, Eric Wieling wrote:
On Wed, 2004-09-01 at 12:26, Glen Johnson wrote:
Does anyone have disconnect supervision working on their TDM400P or
X101P
cards (to/from the telco)?
Yes on all 4 my X100P/TDM400PFXO ports across 3 servers in two states.
It Just Works. Of course
Are these softphones ? If so, make sure there isn't any packet filtering
(firewall) taking place. I dont have too much experience with Asterisk
VOIP so the next thing I would try is not registering with
the server and make call directly to the other phone via its IP.
Hope this helps.
Thank you,
Also make sure there isn't any packet filtering enabled on the BSD box as
well.
Thank you,
Steve Maroney
On Wed, 1 Sep 2004, Jefferson Carvalho wrote:
Hello list,
I've posted my problem on BSD list and i still have the
problem.
The remote side receives the call , but there's no voice
on
I've tried this before, with no luck. I've got to try again this
evening, and I'm looking for some help.
Here's my configuration -- pretty simple, really.
Asterisk box - T100P - TA750(20FXS/4FXO) - phones and outgoing lines
I have an analog modem (Ok, it's a TIVO) that I need to be able to
Hi
I've been playing around with asterisk for a while now at home, just trying
to understand a bit of the technology and seeing what I could get up and
running. Here's where I am at:
I bought myself an X100P card and got an asterisk server up and running on a
gentoo linux distro. I got two
Hello All,
My asterisk installation has now been running for over two months
without a hitch, and I've decided it's time to move things around a bit.
It's currently installed on a 2.7GHz Celeron under RH9 installed on a
10GB leftover drive. Thanks to the strange marketing method called
On 07:05 AM 8/31/2004, Deon Rodden wrote:
How do I limit the length of an extension? In my test IVR/Automated
Attendant (whatever it's called), at the beginning it plays if you know
your parties 3 digit extension, you may enter it now) and then it gives
a list of options. If the caller puts the 3
1) should be more than enuf for 1 channel. I use a P2 400 here for
testing and it worked ok for transcoding besides the schedule notices.
2) Depending how much timing you need to do X100P or ztdummy could
even work just fine.
3. -head
4. i'd rebuild it from src and just copy your configs and
On 06:25 AM 9/1/2004, Juan Jose Comellas wrote:
We intend to use Asterisk with a very large dialplan (with a lot of
functionality for 3000+ users). Each user will be able to change several of
his parameters in the dialplan, so we will be forced to reload the diaplan
constantly. Has anybody else
On Wed, 1 Sep 2004 16:38:30 +0100, Asterisk [EMAIL PROTECTED] wrote:
Oh great. BT at it's best.
Spoken to 4 different product managers / isdn help desk / customer service
weenies. Each one says Eh? What? and I've got to point them to your link.
I'll get back to you on that is the next
Hi, thanks for the reply, only just got round to having a look at it again
(annoying how real life gets in the way of the important stuff ;)
I've had a go at ramping up the tx/rx gain but it doesn't seem to make any
difference. FWIW it's the same with the module in normal fcc mode.
Does anyone
On 1 Sep 2004 at 7:30, Rich Adamson wrote:
A customer of mine has 3 TDM400P cards in a box running asterisk.
On each card he has four FXO modules.
I have set up the dialplan to dial via group 1 for an outgoing call.
Channels 1-12 are in group 1.
If he plugs a telephone cable
Hi !
I have a TDM40B and i try to use it connected to modem for incoming call
data transfert.
I have no problem to use it with a phone and a talk communication work fine.
But when we try to use with modem, with most modem, we got data carrier for
few seconds and channel hungup.
[ TYPE: Null
Thanks for the quick response -- I should have clarified this a little
more.
I was using this board before, but couldn't get ztdummy to work because
the board had the wrong USB controller on it. I switched boards and
then added the X100P. I would rely solely on the X100P for timing if I
go back
Hi all
I'm pretty sure someone must have done this before but I couldnt find any
trace of it on the web so I thought I would drop a note about how I ended up
doing it. I have also posted this info on voip-info.
Warning : This is not very elegant and I'm currently trying to write a patch
in order
Hi,
is there a way to force a user authentication using h323 channel from
asterisk sources? Do I have to use gatekeeper for this?
Is there any way to do it in h323.conf just like in sip.conf?
eg:
[mazek]
secret=xx
auth=md5
tia
mazek
--
http://www.marcinmazurek.com/ ::: nic-hdl:
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