On Apr 25, 2005, at 10:55 PM, Ronald Wiplinger wrote:
Hi Ronald,
What happens in your Asterisk box when you press the Speed Dial
number in
IPS?
Can we make it so that you FIRST answer below questions, please?
| | Let's try it together:
Ronald: wow. Take a breath before you torch a generous
One clarification:
On Mon, Apr 25, 2005 at 05:07:31PM +0200, lenz wrote:
See http://www.oinko.net/astrecipes
All content is licenced as creative commons, so if you got a recipe to
spere, feel free to post it.
Creative-Commons is a group of licenses. You seem to refer to
CreativeCommons
Hi Tim,
which hardware did you use in the asterisk box for the job?
Francesco
Tim Connolly
[EMAIL PROTECTED]
Isn't the zapata.conf only for Digium hardware? I use an Eicon Card.
Eric Wieling aka ManxPower wrote:
Kib Eki wrote:
Hi,
what do i have to configure to get a busy tone when dialing out over
ISDN channel with my Polycom 500 IP?
Try priindication = inband in /etc/asterisk/zapata.con
Hello,
Is it possible to make BT100 phones ring in different ways based on where
the call is coming from?
The general idea is that I need the BT100 ring in 2 different ways depending
on whether the call come from Zap1 or Zap2.
It's because this system is for a receptionist answering two
In data Tue, 26 Apr 2005 09:13:52 +0300, Tzafrir Cohen
[EMAIL PROTECTED] ha scritto:
One clarification:
On Mon, Apr 25, 2005 at 05:07:31PM +0200, lenz wrote:
See http://www.oinko.net/astrecipes
All content is licenced as creative commons, so if you got a recipe to
spere, feel free to post it.
Robert Goodyear wrote:
On Apr 25, 2005, at 10:55 PM, Ronald Wiplinger wrote:
Hi Ronald,
What happens in your Asterisk box when you press the Speed Dial
number in
IPS?
Can we make it so that you FIRST answer below questions, please?
| | Let's try it together:
Ronald: wow. Take a breath before
All,
I found that there is no ringback to the caller (a-party) for
VoIP call but when I make call to registered user, I can hear the
ringback tone.
Beloware the debug log for the two cases:
I wonder if anyone who can tell me why?
Thanks.
Raymond
Case 1: no ringback to the caller
Hi all,
I was just wondering if someone could help me with info on VOIP Gateways.
We are planning to do an * install in an apartment building, this
building is going to require somewhere in the vacinity of 20 E1 lines
(each with 30 voice channels).
Short of buying 20 Servers with Digium cards,
We have a problem whereby once
a call has started there is no noticeable delay however as the call continues
you notice a delay building up, after say 10mins it becomes really
noticeable.
The setup is XTEN with VPN
(Cisco) - Cisco 7206 - VPN (Cisco) XTEN
The RTP and signalling are
going
Kanuri, Seshu (Company IT) wrote:
If you look at your iax.conf lines as under, you will notice that the
two contexts are illegal as they both have same name:
I don't believe that part of your advice is correct.
I have a number of such entries in my iax.conf and they seem to work
without problem.
Hi all,
To my surprise, I change the Dial statement in extensions.conf
from:
exten =
_852.,1,Dial,SIP/[EMAIL PROTECTED],r
to:
exten = _852.,1,Dial(SIP/[EMAIL PROTECTED],20,r)
I can hear ringback tone now. I don't know why but it
just works.
Cheers.
Raymond
- Original Message
Hi,
Look into your "*.conf" files
Res_mysql.conf is a good start
Check for user-id password
Also check the dbsock=.
(the default
value did not correspond to my 'default' installation of sql).
I have now dbsock=
/var/lib/mysql/mysql.sock
Look for THAT file in your system.
Wait until the new card of Digium go out, withit you only will need 1 server
more powerful.
regards.
- Original Message -
From: Callum McGillivray [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, April 26, 2005 9:29 AM
Subject: [Asterisk-Users] VOIP Gateways Asterisk
On Tue, 26 Apr 2005, raymond wrote:
To my surprise, I change the Dial statement in extensions.conf from:
exten = _852.,1,Dial,SIP/[EMAIL PROTECTED],r
to:
exten = _852.,1,Dial(SIP/[EMAIL PROTECTED],20,r)
I can hear ringback tone now. I don't know why but it just works.
In the first line
My problem is that this installation is most likely to occur prior to
the release of the new card (and definitely prior to it's vigorous
testing in the field).
If anyone can give me ideas at this point it would be appreciated.
Callum
Dpto. Tcnico (Softec). wrote:
Wait until the new card
hi everybody
I'm a new Asterisker.
I have a very simple configuration : 1 Sip proxy and 2 grandstream 102 in
ethernet with
private adress
sip proxy : 192.168.2.194
ip phone address : 192.168.2.144
192.168.2.195
I want to make a communication between 2 ip phone with the SIP proxy but i
have 2
Does anyone know if it is possible to resolve an IP
from outside a small LAN. I would like to be able to specify a SIP client
that is outside my office LAN. The problem is that the isp will not
provide a static IP that's affordable. I use a DYNDNS.org address with
it. When I want to use
You can only set 1 distinctive ring if by caller id. There is a tool on the
website to record custom ring tone.
- Original Message -
From: Tomas Florian [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, April
Can anyone recommend any free IP SoftPhones that
are maybe open source?
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Ronald,
I am more than happy to give you the 3 suggestions, when you
appologise to the list. Yes getting things to work can be
frustrating, and sometimes the answers are not as helpful as you'd
like, but I do refuse to help people who get irate on a public list
Especially when the outburst is
Hi everybody,
I am having a problem while setting up queues in Asterisk. Callers are kept
in the queues and told to wait while there are available agents. Even if I
use ringall as strategy the call is not always sent to all free agents. Is
there a problem with Automatic Call Distribution in
Best I have used is fireflyJ
Dinesh.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William C. Lohr Jr.
Sent: Tuesday, April 26, 2005 4:51
PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] IP
Softphone Recommendations
Can anyone
Thanks Stefan, you rule...
now, tell me just one more thing please,
I putted in capi.conf :
msn=12345678
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=siemens
devices=2
and in extension.conf :
[siemens]
exten = 930,1,Dial(SIP/joao)
but this means that when 930 is dialed, user joao
Hi !
Starting this morning I start seeing this in my
/var/log/asterisk/messages :
Apr 26 08:52:13 WARNING[18321] file.c: Unexpected control subclass '17'
Apr 26 08:52:57 WARNING[18321] file.c: Unexpected control subclass '17'
Apr 26 11:21:17 WARNING[20842] file.c: Unexpected control subclass
Meetme is a great tool ..
It can give you a great conference capabilities ...
Also using Faxing will be great ...
Mohamed Farid ,,
-Original Message-
From: Craig Simon [mailto:[EMAIL PROTECTED]
Sent: Monday, April 25, 2005 4:00 AM
To: Asterisk Users Mailing List - Non-Commercial
Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes?
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On 4/26/05, Callum McGillivray [EMAIL PROTECTED] wrote:
My problem is that this installation is most likely to occur prior to the
release of the new card (and definitely prior to it's vigorous testing in
the field).
If anyone can give me ideas at this point it would be appreciated.
There
Then you'll need to check the value of DIALSTATUS and run Busy when
needed. See extensions.conf.sample's [macro-stdexten].
Kib Eki wrote:
Isn't the zapata.conf only for Digium hardware? I use an Eicon Card.
Eric Wieling aka ManxPower wrote:
Kib Eki wrote:
Hi,
what do i have to configure to get
On 4/26/05, Stefan Gofferje [EMAIL PROTECTED] wrote:
Stefan Helbing schrieb:
Hello,
the incomingmsn line in chan_capi's capi.conf is limited to 80 characters
(AST_MAX_EXTENSION default value).
My problem: I have to include several MSNs but NOT all. The interface is a
30 channel PRI
Good day all
I get this error in my cli
chan_zap.c:7407 zt_pri_error: PRI: !! Got a UA, but
i'm in state 0
I have a 4 port Junghannes card connect with 2 bri
isdn lines
It keeps on dropping calls and giving
errors
Please help and advice
Thanks
ALtus
On 4/26/05, William C. Lohr Jr. [EMAIL PROTECTED] wrote:
Does anyone know if it is possible to resolve an IP from outside a small
LAN. I would like to be able to specify a SIP client that is outside my
office LAN. The problem is that the isp will not provide a static IP that's
affordable. I
Hi,
I'm integrating cisco call manager with asterisk
this is what I have in sip.conf
[callman]
type=friend
nat=no
insecure=very
context=dialplan
host=172.16.4.82
port=5060
disallow=all
allow=ulaw
allow=alaw
canreinvite=yes
qualify=yes
and this is my dial statement
Exten =
You entered 'other' as the context in the voicemail_users Database, but
you failed to specify that context when you made the call for voicemail
from the dial plan. The dial plan should be:
VoiceMail(SIP/601-a9a3, [EMAIL PROTECTED])
As stated in other posts on this subject, the Voicemail
Hello
Is thereAsterisk free billing system?if you
know a good system please reply me with it.
thanks in advance
Regards
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To
hi
if receiving a call, I lookup wheather or not that call should be
diverted at user's request. if it should be diverted, I want to use
ForkCDR to keep this entry
orig srcorig dst
and add one more
orig dstdivert dst
but with forkcdr, I only get
orig dst
there is jitter setting for h323 in oh323.conf
where to set min/max jitter buffer for SIP ?
i am getting bad voice via *, maybe this jitter buffer setting will help
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Title: Message
Hi all, looking for
some advice on a good FXO solution for Asterisk, living in the UK brings some
issues (as Asterisk and associated hardwareappears to begeared to
the US/Canadian market) such as calling line ID, impedance settings and echo
cancellation (as a result of the
there is jitter setting for h323 in oh323.conf
where to set min/max jitter buffer for SIP ?
i am getting bad voice via *, maybe this jitter buffer setting will help
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We do some least cost routing and pass calls to a Cisco AS5300 and
are looking at also using a Lucent TNT. It's just a matter of setting
up the dial plan in extensions.conf The cisco is using h323 and if
the client does SIP the protocol conversion is done on the * box,
otherwise it is basicaly
Date: Tue, 26 Apr 2005 12:36:26 +0100
From: Razza [EMAIL PROTECTED]
Subject: [Asterisk-Users] Good FXO for UK use.
To: asterisk-users@lists.digium.com
Message-ID:
!~!UENERkVCMDkAAQACABgAT1LIZaT+QEqJAR
K3kpBGu8KQMG2M/[EMAIL PROTECTED]
Title: RE: [Asterisk-Users] Good FXO for UK use.
So I guess the obvious choice would be the TDM400 with FXO daughter
board, I assume this works with my current zaptel drivers and UK CLID
patch? Can anyone confirm this works fine in the UK or are there other
suggestions?
Hi!
I'm trying to understand how asterisk handles the TON (using the
pridialplan=... directive).
Setting the TON for outgoing calls using pridialplan and
prilocaldialplan works fine. But how can I query and process the TON for
incoming calls?
e.g. in the follwing scenario:
PBX--- asterisk
Title: RE: [Asterisk-Users] Good FXO for UK use.
No the TDM400 does not work, it does not detect calling party termination
correctly, so IVR and voicemail do not see the caller hang up on BT lines.
Digium are aware of the problem, but fixing it doesn't seem to be a high
Hello,
does someone offer DIDs from the areas of shanghai and/or bangalore.
Many thanks,
Marc
--
CTOMarc Storck
MS Networks SA [EMAIL PROTECTED]
IT Service Providerhttp://www.msnetworks.lu
15, route d'Esch Phone: +352 2727 3030
Klaus Darilion wrote:
Hi!
I'm trying to understand how asterisk handles the TON (using the
pridialplan=... directive).
Setting the TON for outgoing calls using pridialplan and
prilocaldialplan works fine. But how can I query and process the TON for
incoming calls?
e.g. in the follwing
Eric Wieling aka ManxPower wrote:
Klaus Darilion wrote:
...
e.g. in the follwing scenario:
PBX--- asterisk PSTN
1. The PBX sends SETUP messages with the appropriate TON. I want to
rewrite the called number into a common format to make an ENUM lookup.
Thus, I need to query the TON sent by
SAN JOSE, Calif., April 26, 2005 - Cisco Systems® today announced a
definitive agreement to acquire privately-held Sipura Technology, Inc.
This represents Cisco's first acquisition for its Linksys division, the
leading provider of wireless and networking hardware for home, Small
Office/Home
Why would you expect a bunch of fax modems to work any better than
spandsp? If spandsp doesn't work reliably your system is very likely broken.
I have had hundreds of complaints about spandsp reliability. I have
analysed at least 50 or 60 audio logs. I have found maybe 5 or 6 which
has real
Can anyone recommend any free IP SoftPhones that are maybe open source?
Mine is not open source, but it's free for non-commercial use. Give it a try
http://www.marccharbonneau.com/asterisk/mediaxphone.php
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try with
type=peer
good luck
Edgar
On Tue, 2005-04-26 at 13:08 +0200, Alessio Focardi wrote:
Hi,
I'm integrating cisco call manager with asterisk
this is what I have in sip.conf
[callman]
type=friend
nat=no
insecure=very
context=dialplan
host=172.16.4.82
port=5060
You might look into the Lucent TNT's, they do SIP/MGCP (with the Hash
codes, os 10.1.xx+), and also terminate modem calls. They are cheap
(check ebay, www.qualitek.net) and their are loads of them out there.
One TNT will handle your requirements easily, their is an example on the
wiki on how to
Hi, I just wanted to know if Digium support ETSI ISDN?
Thanks.
Cheers,
Angelo
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I'm also looking for numbers from
HongKong,
Taiwan,
Japan and
Singapore
So if someone has some DIDs from this areas, I'm very interested to get
one or another from those DIDs.
Best Regards,
Marc
Marc Storck wrote:
Hello,
does someone offer DIDs from the areas of shanghai and/or bangalore.
Many
Hi Nathaniel,
ETSI ISDN is used by 99% of the world's ISDN E1s, so you can guess the
answer. :-) ETSI ISDN is also known as CTR4, Net5 and most commonly
EuroISDN. It is known as EuroISDN in the * config files.
Regards,
Steve
Nathaniel Angelo A. Torres (247talk) wrote:
Hi, I just wanted to know
I have just installed a release version 1.0.7 of asterisk: I already
installed in past asterisk and in my previous installation I may find
the dial command on CLI that now I haven't found: it is possible?
The lack of dial CLI command is an upgrade?or Is there some problem in
my installation?
Thanks Steve.
Cheers,
Angelo
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: Tuesday, April 26, 2005 8:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Digium for ETSI ISDN
Hi
On April 26, 2005 08:12 am, Steve Underwood wrote:
Why would you expect a bunch of fax modems to work any better than
spandsp? If spandsp doesn't work reliably your system is very likely
broken.
I've had spandsp crash out on some kind of floating point error about a half
dozen times over
Dear All,
I am new to this mailing list , I have bought some digium cards to play with , Installed it and configured asterisk . I was able to test voicemail IVR , I succeeded also to use xlite from a windows machine to call another phone through a PSTN line. and call the xlite client from a
On Tue, 26 Apr 2005, Klaus Darilion wrote:
Anyway, if I set TON to unknown, I have to send the number according to
the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the
PBX does not use UNKNOWN, I have to translate the numbers out of their
original TON to ton=unknown.
Hi Steve,
I sent a mail to this list a week ago regarding exactly this issue.
Spandsp doesn't work for me (getting 200rows tiffs), but sending and
receiving faxes through a FXS-FXO bridge (a TDM11B) works without
problems.
My motherboard is based an Aopen AK33 (VIA686a chipset, KT133, 700Mhz
I have just installed a release version 1.0.7 of asterisk: I already
installed in past asterisk and in my previous installation I may find
the dial command on CLI that now I haven't found: it is possible?
The lack of dial CLI command is an upgrade?or Is there some problem in
my installation?
I'd have to second this, it works flawlessly for us, the issues we do
have are with devices not properly turning off echo cancellation...
W. Kevin Hunt
CCIE #11841
MCSE, Linux+ SME
www.huntbrothers.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Title: Message
This
maybe an out of place comment but it would appear Digium show little to no
interest in non-North Americanimplementations, do we know if they are ever
going to resolve this issue? or indeed how much it would cost? Based on my
experience I'm sure there are a number of UK
yes *71 will disable call waiting on any phone used
with [EMAIL PROTECTED]
see the Handbook for more info
http://asteriskathome.sourceforge.net/handbook/
--- Anton Krall [EMAIL PROTECTED] wrote:
How? You mean if you use [EMAIL PROTECTED] right?
|-Original Message-
|From: [EMAIL
Anyway, if I set TON to unknown, I have to send the number according to
the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the
PBX does not use UNKNOWN, I have to translate the numbers out of their
original TON to ton=unknown. Therefore, I need to process the incoming
TON. How
Hi Andrew,
If you can catch one of these events, and get a traceback of the stack,
I will take a look. This is not happening to most users, so it must be
some specific combination of things on your machine. I have reports of
high volume faxing running for extended periods from some users.
Hi Julian,
Sounds like a frame slip problem if the result depends on the source.
Most people, including me, have trouble with the TDM cards. They worked
without problem when I was first developing the FAX software in spandsp,
so I assume the TDM driver has gathered bugs since that time.
I was afraid you would say that.
Does anyone out there have the latest firmware for the
Soundpoint IP 4000?
Thanks,
Wiley
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
HalesSent: Monday, April 25, 2005 7:45 PMTo: 'Asterisk
Users Mailing List - Non-Commercial
Yep. Or if you hand code the feature into your dial plan too
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Monday, April 25, 2005 6:13 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
Hello
I have just setup my first Asterisk box and Im having
a great time.
I am having a little trouble getting incoming calls to
answer. This is what I see on the console:
Apr 25 17:01:07 NOTICE[3514]: chan_zap.c:5374 ss_thread: Got
event 2 (Ring/Answered)...
-- Executing
As Steve has mentioned several times, it seems the TDM-fxo boards
have an issue with missed frames that no one is addressing. Very few
(if any) TDM users have been able to make spandsp function correctly,
and the few that might have it working don't know why.
Having played around some with zttest
On Tue, 2005-04-26 at 08:57 -0400, Andrew Kohlsmith wrote:
On April 26, 2005 08:12 am, Steve Underwood wrote:
Why would you expect a bunch of fax modems to work any better than
spandsp? If spandsp doesn't work reliably your system is very likely
broken.
I've had spandsp crash out on some
Does anyone know of a way to pass a value back to the dial plan after
calling a macro from the dial app in the 1.0 release? I think this
should be pretty simple, but I can't quite figure out how.
The example would work except that the modified value of found is not
usable when Dial ends. I
If this has already been posted I apologize for the redundant post.
http://newsroom.cisco.com/dlls/2005/corp_042605.html?DCMP=BAC-TS01
--
Cory Andrews
Senior Partner
VOIPSupply.com
+
V: 800.398.VOIP X22
F: 716.630.1548
E: [EMAIL PROTECTED]
On Tue, 2005-04-26 at 14:11 +0100, Julian J. M. wrote:
Hi Steve,
I sent a mail to this list a week ago regarding exactly this issue.
Spandsp doesn't work for me (getting 200rows tiffs), but sending and
receiving faxes through a FXS-FXO bridge (a TDM11B) works without
problems.
My
Hi Peter!
Peter Svensson wrote:
On Tue, 26 Apr 2005, Klaus Darilion wrote:
Anyway, if I set TON to unknown, I have to send the number according to
the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the
PBX does not use UNKNOWN, I have to translate the numbers out of their
That's what I figured.. Im not using [EMAIL PROTECTED] ... Plain ol' good
asterisk...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|[EMAIL PROTECTED]
|Sent: Martes, 26 de Abril de 2005 08:28 a.m.
|To: Asterisk Users Mailing List - Non-Commercial
Marc Storck wrote:
Anyway, if I set TON to unknown, I have to send the number according
to the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if
the PBX does not use UNKNOWN, I have to translate the numbers out of
their original TON to ton=unknown. Therefore, I need to process the
On 4/26/05, Nathaniel Angelo A. Torres (247talk) [EMAIL PROTECTED] wrote:
Hi, I just wanted to know if Digium support ETSI ISDN?
Yes
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Hi folks,
I'm curious; What does everyone do for failover? I have two servers,
same os/compilation. I designate one the master, the other the slave,
and I rsync the config files once an hour and trigger a restart when
convenient command on the console. These two servers are setup in the
Guys.
Im using YAC to send callerid info to PCs and I was wondering if there is a
way to get the IP of a certain SIP or IAX client/technology when a dial
command is issued.
For example, if the dialplan has a dial sip/client or iax2/client, is there
a way to get the current clients IP so I can
Trying to make a call via our PRI: (CVS everything,
CVS-NHEAD-04/23/05-16:08:12)
-- Executing Dial(IAX2/[EMAIL PROTECTED],
Zap/R2/2815699900|30) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called R2/2815699900
-- Channel 0/19, span 2 got hangup
-- Channel
On Tue, 26 Apr 2005, Klaus Darilion wrote:
You have two options:
1) Use the CALLINGTON variable in the dialplan. This is only for the
calling party number, not the called party number.
Bad thing. I guess this is an important feature when interacting with
existing PBXs. How are
On 4/26/05, Michael Baird [EMAIL PROTECTED] wrote:
You might look into the Lucent TNT's, they do SIP/MGCP (with the Hash
codes, os 10.1.xx+), and also terminate modem calls. They are cheap
(check ebay, www.qualitek.net) and their are loads of them out there.
One TNT will handle your
-Original Message-
snipped
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Sampson
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Hi All,
Can anyone help me out here? I'm having some issues configuring my IPTables
firewall to properly NAT SIP and RTP packets to my asterisk server hiding
behind it.
Here are my current rules:
#Inbound SIP to HERMES
$IPTABLES -A PREROUTING -t nat -i $EXTIF -p udp --dport 5060 -j DNAT --to
Hello all,
I am new to Asterisk and I tried a basic configuration with 2 SIP phones
(FCI IP Ranger), and it works very well !!
I wanted to understand furthermore the asterisk product, his technical
architecture, and after that try to understand how to add
functionnalities thanks to te API. I have
On Tue, 2005-04-26 at 08:34 -0600, Rich Adamson wrote:
As Steve has mentioned several times, it seems the TDM-fxo boards
have an issue with missed frames that no one is addressing. Very few
(if any) TDM users have been able to make spandsp function correctly,
and the few that might have it
Hello,
I've got an older machine that is being locked/hung
by what appears to be the X100P card. I'm running
the following:
[9:[EMAIL PROTECTED]:[/home/jesse]# qpkg -v -I zaptel
net-misc/zaptel-1.0.7 *
[9:[EMAIL PROTECTED]:[/home/jesse]# qpkg -v -I asterisk
net-misc/asterisk-oh323-0.6.5 *
Anybody would have the japanese voice files for *?
I need now the number's recording at least.
Thanks,
Isamar
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Time Bandit wrote:
Can anyone recommend any free IP SoftPhones that are maybe open source?
Mine is not open source, but it's free for non-commercial use. Give it a try
http://www.marccharbonneau.com/asterisk/mediaxphone.php
I´m using X-lite on windows and linux, looks pretty well.
On April 26, 2005 09:47 am, Adam Goryachev wrote:
I was under the impression that pretty much all of these problems were
usually traced to the version of libtiff that was in use... Perhaps you
should try to track it down/solve the problem rather than patch it over?
Nope; it's not a tiff issue;
Title: Message
I just bought a DigitNetworks card called "DigitNetworks
X100P - FXO PCI card" which supposedly is compatible with the discontinued
Digium X100P card. This is a single port FXO card. Tell me how to test
forthe TDM400 problem and I'll perform a test and post my results back to
I have just installed a release version 1.0.7 of asterisk: I already
installed in past asterisk and in my previous installation I may find
the dial command on CLI that now I haven't found: it is possible?
The lack of dial CLI command is an upgrade?or Is there some problem in
my installation?
Having played around some with zttest (modifying the code to better
understand the issues), it would appear the TDM card consumes about
1.02 seconds to obtain one second of data. That would suggest the
I would like to chime in with my experience:
We are trying to use SpanDSP off of a PRI to
On Tuesday 26 April 2005 9:58 am, Anton Krall wrote:
Guys.
Im using YAC to send callerid info to PCs and I was wondering if there is a
way to get the IP of a certain SIP or IAX client/technology when a dial
command is issued.
For example, if the dialplan has a dial sip/client or
We have terrible problems sending faxes via the TDM cards. Not even
using SpanDSP. Just TE110P for the telco side and TDM400P for the fax
machine.
Steve Underwood wrote:
Hi Julian,
Sounds like a frame slip problem if the result depends on the source.
Most people, including me, have trouble
On Tue, 26 Apr 2005, Marc Storck wrote:
I have a Digium E100P card, with an EuroISDN PRI E1. On incoming calls
the CALLINGTON variable is empty. I have the latest stable version of
asterisk. Do I have to use another variable or is the TON only support
in CVS?
CALLINGTON was not populated
On Mon, 25 Apr 2005, Wiley Siler wrote:
Call waiting can be disabled in Asterisk via *71 regardless of the phone
used.
Cheers,
Wiley
Well, this is part of a larger problem I'm having.
I can't get CheckGroup/SetGroup to work as I think it should for my
dynamically added ACD agents.
The
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