Kevin Steil a écrit :
I use VMWare, but will start testing XEN...I use VMWare to slice up some
nice big servers to provide dedicated hosted PBXes. We also use the VMs
for easy deployment and is a vital part of our DR Plan...
Which version of VMWare are you using? Are you having any issues
The XML minibrowser is available in a special image only.
http://www.snom.com/minibrowser/firmware/
In future images it will be in by default.
On Friday 27 January 2006 16:15, Colin Anderson wrote:
Aha, I see it's 4.1, cool. So I just have to do a straight upgrade to 5.0
and I have this new
On Sun, Jan 29, 2006 at 05:24:12PM +1000, Rob Thomas wrote:
The question is somewhat ludicrous, and I'm slightly surprised that
no-one has sat down and done the maths about bandwidth utilization. So I
did.
To handle 5000 calls coming in over a PRI, you'd need 210 or so T1s or
170 E1's.
Hello,
How many digium cards is supported per asterisk
server ?
Regards
Harry
___
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exceptionnels pour
What about starting such a thing on groups.yahoo.com?
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, January 28, 2006 6:32 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] english snom
I put a .call file under asteriskserver:/var/spool/asterisk/outgoing via scp
(with keys)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darrell Long
Is anyone using Asterisk (and Festival) to make calls to
appropriate persons (techs, etc. )
Hi..
I have one intel 536 EP. Does it possible use it as x100p clone for
asterisk? I tried today with no luck :(..
Here is what I did :
- plugged the card
the card is recognised as (lspci -vv):
00:09.0 Communication controller: Intel Corp. 536EP Data Fax Modem
Subsystem: Intel
hi
I know people in here use 4 boards, but I believe the only real
limitation is the number of PCI slots in your computer.
Obviously, with 4x4xE1/T1 boards (480 B channels) you need a pretty fast
monster to drag it around due to the way the zaptel/asterisk works. So
the actual limit is
Hello!
I am considering mass deployment of Budgetones 102. According to their
website, remote provisioning (configuration via TFTP) is possible.
Anyone has experience with this? Is this really working?
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That INFO must be inside the extsting dialog, maybe that was the
problem.
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Phil Blundell
Sent: Monday, January 30, 2006 10:16 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
On Mon, Jan 30, 2006 at 10:08:00AM +0100, Mimmus wrote:
I put a .call file under asteriskserver:/var/spool/asterisk/outgoing via scp
(with keys)
There are chances asterisk will read the file before the sshd has
completed writing it. Normally you need to move it into the spool
directory.
--
On Mon, Jan 30, 2006 at 10:39:11AM +0100, [EMAIL PROTECTED] wrote:
hi
I know people in here use 4 boards, but I believe the only real
limitation is the number of PCI slots in your computer.
Is this 4 E1 boards?
Obviously, with 4x4xE1/T1 boards (480 B channels) you need a pretty fast
Hello,
How many digium cards is supported per asterisk
server ?
hi
I know people in here use 4 boards, but I believe the only real
limitation is the number of PCI slots in your computer.
Obviously, with 4x4xE1/T1 boards (480 B channels) you need a pretty
fast monster to drag it around
Is anyone using Asterisk (and Festival) to make calls to
appropriate persons (techs, etc. ) when Nagios generates a
particular type of alert?
If so, I would love to hear how people are doing it.
yeah...
do that on ALL the services and hosts, and see how long it takes
before your
Hello,
Is there an app_snmp for asterisk-1.2.3 ?
Harry
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! Découvez les tarifs exceptionnels pour appeler la
France et
On Mon, 30 Jan 2006, Sven Fischer (support) wrote:
The XML minibrowser is available in a special image only.
http://www.snom.com/minibrowser/firmware/
In future images it will be in by default.
Is the xml minibrowser firmware equivalent to 5.0+xml, 5.2+xml? It's not
obvious.
Also, might
Hello to all
Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know
the best way of implement a centralized address book system.
Maybe the solution is LDAP, but these clients doesnt seem to support
LDAP.Who should contact the LDAP directory? the SIP clients or the SIP
server?
On Mon, 2006-01-30 at 11:42 +0200, Dmitry Ivanov wrote:
I am considering mass deployment of Budgetones 102. According to their
website, remote provisioning (configuration via TFTP) is possible.
Anyone has experience with this? Is this really working?
It does work, yes, though I think you
Hello Joao,
I'm using SER and Asterisks-based system, with centralized LDAP backend.
To access LDAP I use SOAP and DSML.This is now used for every
provisioning/management/billing/ivr activity in the system. In future I
plan to have centralized phonebook based in LDAP.
I think that having
Hi,
Does anyone know how to play music to a caller while you dial a second call?
Once the second calls has answered, i'd like to music to stop, and the calls to be bridged.
Thanks
Dan Journo
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Heya,
I upgraded from 1.2.0 to 1.2.3, and when a new call comes in on a
realtime queue, the queue settings and members are not updated anymore!
Only a reload of Asterisk seems to update the settings. Is this a bug or
is there some way to solve this?
Cheers,
Frank
Yes its how I started (ldap+radius(
But it depends what you want to do.
1) If you want to have nice display in softphone (or hardware phone with
LCD) of global system phonebook and/or private phonebook - I'm sorry, no
vendor is supporting this.
I was trying to convince few videophone
Using G711A (ie, worst case bandwidth wise):
it's 64kbit/s not 64Kbyte/s
so it's 320Megabits per seconds
That will only do if you talk a lot with your mother in law! ;-)
For the rest of the conversation (those with both speaking):
5000 * 64k * 2 = 640M
It should in theory work with a
Matt:If you use centralized configuration of your Polycom, using XML files; you can disable the call forwarding by setting the
divert.fwd.x.enabled attribute of the fwd XML element to 0.For
more information and extra attributes you can check the 4.6.2.3.1
section of the IP 300 Admin Guide,
Hi,
the server play the standard music on hold while you are calling the
second person. Bridging the calls is an option of the phone, although
once bridged, the music will stop straight away.
Dan Journo wrote:
Hi,
Does anyone know how to play music to a caller while you dial a
second
That manager looks really awesome!
There is 1 problem.. I only took 1 semester of German 15 years ago.
Looked all over the page for the English button, but I could not find one.
I did wake up 10 minutes ago, so I could still be blind.
I will rephrase the statement..
AMP hands down is STILL
Sorry my vote goes to AMP for sure.
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan-Michael.
Guenther (in-put GbR)
Sent: Monday, 30 January 2006 2:58 AM
To: asterisk-users@lists.digium.com
Subject: Re: Re: [Asterisk-Users] Web interface
I wonder if anyone can tell me which version of spandsp is used in iaxmodem?
I would like to use the app_rxfax application WITH iaxmodem.
Regards Allan Gee
Phone: +27 21 4644400 Ext. 103
www.equation.co.za
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
On Mon, Jan 30, 2006 at 06:05:57AM -0600, Zac Amsler wrote:
That manager looks really awesome!
There is 1 problem.. I only took 1 semester of German 15 years ago.
Looked all over the page for the English button, but I could not find one.
I did wake up 10 minutes ago, so I could still be
On Mon, Jan 30, 2006 at 12:49:15PM +0100, [EMAIL PROTECTED] wrote:
Using G711A (ie, worst case bandwidth wise):
it's 64kbit/s not 64Kbyte/s
so it's 320Megabits per seconds
That will only do if you talk a lot with your mother in law! ;-)
For the rest of the conversation (those with
On Monday 30 January 2006 13:03, Phil Blundell wrote:
Personally I'd be a bit wary of mass Budgetone deployment for other
reasons, but the remote configuration stuff shouldn't be a problem.
What reasons do you mean?
Grandstream use basically the same configuration file system for the
On 1/30/06, Strain Jer [EMAIL PROTECTED] wrote:
I was curious which one is best suited for asterisks. Thanks
The 'best' depends on your personal flavor I guess.
However I'm impressed by voiceone (http://www.voiceone.it/), didn't
use it yet, but will surely look into it sooner or later.
cheers
Hello, I would like to find an appropriate solution for SIP to H323
translation (vice versa would be great too!), in an environment where there's going to be 100+
concurrent calls: has anyone succesfully implemented such a translator/gateway,
e.g. using Opal+OpenH323/Asterisk or anything else? Any
Allan Gee wrote:
I wonder if anyone can tell me which version of spandsp is used in iaxmodem?
IAXmodem version 0.0.8 uses spandsp development snapshot 20051220... so
it's 0.0.3, more recent than pre6.
I would like to use the app_rxfax application WITH iaxmodem.
This will be possible
Seems like backgrounddetect is missing some checks (perhaps)?
Using an analog TDM04B card how do I tell if the line is busy?
The call (from a call file) is always given a state of answered
immediately after calling even though the person has not yet
answered the call.
My message to the person
Asterisk talks SIP-UDP and LCS talks SIP-TCP. A guy called hjlee or
Hyoungjoo Lee has been working on a TCP-patch for Asterisk but this patch
is not yet in the official Asterisk and he cannot work on it any longer.
Hoe good is it ? Also there is SER or SIP Express Router from iptel.org,
does this
All,
Thanks for the help. Checking on and changing the route based on
dialstatus is the way to go.
Thanks,
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On Mon, 2006-01-30 at 15:14 +0200, Dmitry Ivanov wrote:
On Monday 30 January 2006 13:03, Phil Blundell wrote:
Personally I'd be a bit wary of mass Budgetone deployment for other
reasons, but the remote configuration stuff shouldn't be a problem.
What reasons do you mean?
Just that, from
Well, skype. but i was tweaking some code. This is more a question for
lab usage.
On 1/28/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
Erick Perez wrote:
Besides the codecs that * supports. Is there any ISAC implementation
for asterisk available?
This is to be used mainly with
I have found * with the ooh323 channel to be best for this.
On Mon, 2006-01-30 at 15:23 +0200, [EMAIL PROTECTED] wrote:
Hello,
I would like to find an appropriate solution for SIP to H323
translation (vice versa would be great too!), in an environment where
there's going to be 100+
64kb/s channel will require a bandwidth of about 90kb/s using ONE voice packet
per UDP packet. The overhead is a bit low if you put TWO voice packets in ONE
UDP packet.
check this out.
http://web1.egvrn.net/tokata/VoIP%20Bandwidth%20Consumption.pdf
-Original Message-
From: [EMAIL
I have a DrayTek Vigor3300V gateway with 8 FXO ports. I am trying to
configure asterisk to dial out on the gateway. I have one of the FXO ports
configured on sip account 100. If I dial the sip account then the router
gives me dial tone, with which I can dial a number. Unfortunately this is
not the
Title: Manage api- Matching 'Newchannel' event with the 'Originate' command
Hi all,
When the 'Originate' command is issued with 'Async' open set to 'yes', I got the response right away with the correct 'ActionID'. What follows is the 'Newchannel' event with a 'Channel' ID, but their is
5k+ simultaneous calls (in/out) are becoming normal with the kind of
call centers being opened in my country during the past 24 months
(Panama, Central America).
Take Dell Corp. for example. the call center they have here is about
3k people taking/making calls (internal, to/from US, Europe,
64kb/s channel will require a bandwidth of about 90kb/s using ONE voice packet
per UDP packet. The overhead is a bit low if you put TWO voice packets in ONE
UDP packet.
And what is 'one voice packet'?
Jan
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Hi,
I have a problem with setting outgoing caller id to nothing (secret)
on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID
seems to work fine when connecting the same line to a Ericsson PBX - so
something must be wrong in my settings, but I don't know what.
I've tried:
One voice packet is about 20ms of sampling depending on the codec.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Monday, January 30, 2006 9:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Check with your service provider.
Hälsningar
/Olle
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Hello,
It's kind of sad, but the only way to have the events of a specific
call in EVERY instance have a unique tag to follow is to use the
CallerIDname field.
Call variables(that you can set at Origination) and variable
inheritance have gotten much better in the last year, but they still
don't
We do this routinely as a service for our customers. How many phones do you
need to provision? Do you already have the phones? Contact me off list and
I can help you out. -Mike
Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
-Original
Hi,
I'm wanted to do working anonymous calling with my
sip provider.
To do it, I use SetCallerPres(prohib).
The problem:
The "fromuser=" parameter overide the value of
"CallerID(number)" and do it don't working.
Anyone had an idea?
Tank's
Loic Foucault
I had a problem with the scripts you can bulk generate, they are linked
to the MAC address you initially put in, so if the phone packs in you
can't just rename the file.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil
Blundell
Sent: 30 January
How many free intreuppts do you have, its hard to get more than two to work
in most systems. How many PSTN lines are you trying to support?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, January 30, 2006 1:00 AM
Hello,
I have a situation where I need to differentiate between registrations
by users where there might be clashes on the left hand side (username)
portion of the SIP From URI. (for a multi-domain virtual hosting system)
It seems that only the username portion is used for SIP
I've some DID's that I'm using for in-bound faxing, but I'm having some
trouble with getting that working perfectly on my T1. So I'm thinking of
pointing them to an analog line. Will the DID's simply come in over the
analog, presumably sending the DID digits via DTMF? Or is that not
something
Also there is SER or SIP Express Router from
iptel.org, does this do what I need and how do I do it ?
Yes.
Converting from TCP-to-UDP is simple; in ser.cfg put:
# Forward to Asterisk
if (method == INVITE) {
if (uri =~ sip:[EMAIL PROTECTED]) {
log(1, Forwarding
Hi all. I am
trying to find out what the most popular soft phone for Windows is for use with
Asterisk. SIP or IAX?
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com
Tel: (519) 963-3020
Fax: (519) 451-6615
Lead, follow or get
Dearest List,
I guess I missed this point: Is it true that if you change the echo canceler
in zconfig.h, and then recompile/install your zap modules, that for this to
be taken into effect by * you must then recompile/install *?
I would have figured that the zap echo cancellation method was
Trying to compile asterisk (again) from scratch.
I seem to be still experiencing the effects fro Jan 25 where I get no sip
to sip audio.
I have tried upgrading to 1.2.3 which has made no change in the
problem.
I am starting over and now trying to compile/install /trunk
zaptel
libpri
asterisk
How many TDM2400P cards can I safelly install in one PC? I'm loking for
answers from whoever has a working scenario with * and a number of cards
higher than one.
Thx,
Juan
__
Mensagem enviada usando o Webmail da ViaLink ver. 2.7.8
One thing I was pondering: you are not, by chance, using the same
sip.cfg between version 1.4.1 and version 1.6.2 are you? The file has
changed significantly between these versions, and certain acoustic
settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention
that
Hi Harry,
How many IRQ do you have ?
Be carefull for power supply is it is several TDM2460E (all FXS ports) !
It is better to use a seconf power supply...
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
[EMAIL
Hi there,
Don't think you have to recompile * if you've already compiled * with zaptel
before. (chan_zap.so exists)
Should just have to rebuild zaptel, install the module, and do a ztcfg
Good luck,
- Original Message -
From: Brent Torrenga [EMAIL PROTECTED]
To:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Upgrading to the correct sip.cfg fixed the problem. The Polycoms are
back to their great speakerphone-ness. A gotcha is that the new
sip.cfg now contains ntp settings. You'll need to modify these to fit
your
Check http://www.voip-info.org/wiki/view/Asterisk+at+large
Or sipedu http://mit.edu/sip/sip.edu/
Plenty of examples
/Vel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sharon
Sent: Friday, January 27, 2006 4:41 PM
To: Asterisk Users Mailing List -
Your settings are fine. Debug PRI to make sure SETUP message is OK (which
probably is) and then check with you PRI provider that callerid is enabled
on that PRI - E1.
/Vel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday,
Juan Carlos Castro y Castro wrote:
How many TDM2400P cards can I safelly install in one PC? I'm loking for
answers from whoever has a working scenario with * and a number of cards
higher than one.
Depends on the specs of the server. For example, a quad Xeon will be
able to service many more
I would love to run Asterisk on an old laptop, in a mostly solid state
configuration, with no HD. The laptop is slow (Pentium 233), and I
need PCMCIA support (for my network card). Are any of you aware of a
live CD that might work?
Thanks,
Dave
___
Hi,
Can someone explain to me or point me in the direction of documentation
for the domain support feature in 1.2.x?
Specifically I need to be able to have sip users who are authenticated
as [EMAIL PROTECTED]
Thanks..
--
-Barry Flanagan
___
I've some DID's that I'm using for in-bound faxing, but I'm having some
trouble with getting that working perfectly on my T1. So I'm thinking of
pointing them to an analog line. Will the DID's simply come in over the
analog, presumably sending the DID digits via DTMF? Or is that not
Ken,
Analog DID's are a bit backwards compared to normal POTS lines. I don't
know about outside the US, but here (in California, specifically) I've
done a few analog DID installs on some NEC PBX equipment. The trick is
that with an analog DID line, the CPE provides the battery to the telco
Hi,
In Brazilian POTS, the caller holds the callee line hijacked - the call
is not droped until de caller hungs up. I really don't know the right
english term to describe this behavior.
My problem is that its most standard here, and there are a number of
hacks that count on this. For
I prefer IDEFISK.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
MorrowSent: Monday, January 30, 2006 8:35 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] Most Popular FREE SoftPhone for Windows
Hi
In article [EMAIL PROTECTED],
Steve Gladden [EMAIL PROTECTED] wrote:
I am starting over and now trying to compile/install /trunk
zaptel
libpri
asterisk
following the instructions to grab the source trees:
# svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
# svn checkout
Michael Collins wrote:
Analog DID's are a bit backwards compared to normal POTS lines
If you point a DID number at a regular POTS line then when a call rings
in it simply rings in like a regular analog phone line. Unless the
carrier can provide some form of DNIS on an analog line I
Hi All,
I want to setup the interconnectionm between two servers, both having sip
clients behind firewalls. I want the calls from any of the servers to land
on any of SIP clients on the other. I am looking for dial out plans with the
sample configuration files .
Thanks,.
satish
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio
Sent: Monday, January 30, 2006 9:22 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] DID over analog?
I've some DID's that I'm using for in-bound
Im trying to detect before entering in Meetme , which dtmf has been entered.
I did a Background(file) and go to a context where i define a exten =
_X.,1,Meetme()
I have detected that with (1.2.1) when 1 is entered and conference 1
must be created, extensions say it is not possible and
Barry Flanagan wrote:
Hi,
Can someone explain to me or point me in the direction of documentation
for the domain support feature in 1.2.x?
Specifically I need to be able to have sip users who are authenticated
as [EMAIL PROTECTED]
For authentication, we only look at
1) Whether the domain is
Hi
all,
has anyone tryied to
configure asterisk with Kirk IP600 Dect-IP gateway?
Could it works using
the skinny channel ?
Thanks
Giordano
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Barry Flanagan wrote:
Hi,
Can someone explain to me or point me in the direction of documentation
for the domain support feature in 1.2.x?
Specifically I need to be able to have sip users who are authenticated
as [EMAIL PROTECTED]
For authentication, we only look at
1) Whether the domain is
And you should consider how many FXS's you're running. More than one card with all FXS's will require a turbo fan to cool and if they all ring you'll need a decent power supply to handle the power draw.Rob
On 1/30/06, Steven Ringwald [EMAIL PROTECTED] wrote:
Juan Carlos Castro y Castro wrote: How
Hi!
I'm trying to install the RPMS, in the installation document the following
module is not mentioned: perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm
But the RPM is in the CentOS 4 directory.
On CentOS 4 the rpm is even already present albeit an older version:
[EMAIL PROTECTED] rpms]# rpm -qa |
I have a 7940 trying to connect to an existing running system.
tftp is configured and running normal.
(NOTE: I know there is a later SIP version but this is the one I have)
I see the phone bootup and ask for OS79XX.TXT which has
[EMAIL PROTECTED] src]# cat /tftpboot/OS79XX.TXT
P0S3-05-1-00
Hi!
Yes, it works (sort of) but I still have some issues. When using more than
2 handsets some of them do not always ring on an incoming call. This might
be because I use only 2 Kirk handsets and the rest are Siemens, maybe it's
the driver
I created a howto for it, you can find it here:
Ah -- for all intents and purposes, assume I can obtain the most kickass PC
server hardware in the known Universe. So -- any real-life experiences out
there?
How many TDM2400P cards can I safelly install in one PC? I'm loking for
answers from whoever has a working scenario with * and a number
A PICMG card equiped with Pentium M CPU permits to reduce dragsticaly the
power consumption.
As this, you can use more power from your PSU for the interface cards.
But, for several TDM2460E/B cards with a heavy traffic charge (many
simultaneous rings), I believe that it could be better to use a
I have 3 Grandstream Budge-Tone 100 with Firmware 1.0.7.11beta, and they
are working very well since more then 4 month now.
Guenther
Davao City, Philippines, Planet Earth, 32.1 °C
Phil Blundell wrote:
On Mon, 2006-01-30 at 15:14 +0200, Dmitry Ivanov wrote:
On Monday 30 January 2006
Hi,
I didn't have this gateway, But on welltech 4fxo gateway. You can just dial
SIP/[EMAIL PROTECTED] Even the gateway didn't register to the
asterisk server.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Corne
Vermeulen
Sent: Monday, January
I am experimenting with an asterisk setup in my office. The last bit I have to
test is working with analog lines. I have TE411p digium card, with an ISDN
line plugged into the first, a channel bank plugged into the second port, and
the last two ports empty. I have the following setup in my
I have an iaxy the is across a 802.11b link from my asterisk server.
signal strength is good and it has been working fine there for about a year.
Friday night lightning took out the x100p card in my asterisk server and
I just got it all working again last night. Lucky I had a spare.
Since I
Hi;
I'm trying to implement what is known by Cisco
Callmanager as regions: Specify that when phones from zone A call to
phones in zone B, use g729, but if they call to zone C, use g711. Any ideas on
how to achieve this?
Thanks!
Francisco Sedano
Hello Zac,
There is 1 problem.. I only took 1 semester of German 15 years ago.
Looked all over the page for the English button, but I could not find one.
I did wake up 10 minutes ago, so I could still be blind.
the language of the module is influenced by the language you choosed for
webmin.
hello all,
i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the
On Mon, 30 Jan 2006, Dmitry Ivanov wrote:
I have created dynamic CGI-like TFTP server so I will create config
files on-the-fly. Now we use this system (dynamic tftp server and Perl
CGI script) for country-wide Sipura 3000 configuration. BTW, if
anyone is interested I can send sources of this
On Mon, 30 Jan 2006, Phil Blundell wrote:
Budgetones as they do on the Handytones and the GXP-2000. Obviously you
need some way to make the files in the first place: when we deployed our
GXP-2000s I ended up writing a little Python script to create the
Grandstream config files (and the
At first glance it sounds like a routing issue. Are you IAX to your phones,
but SIP to your tisp provider? Any change on your asterisk box / firewall /
ISP / TISP since then?
My first guess is that when you replaced the card you changed a network /
iptables setting etc. on your asterisk box.
All the phones, the iaxy and the server are on a 192.168.3.* network and
the only outside interface currently is the x100p card. I do have an
account with a voip provider but I haven't got that setup up yet since
rebuilding the server on Sunday.
So there shouldn't be any routing issues coming
Remco Barende [EMAIL PROTECTED] wrote:
Hi!
I'm trying to install the RPMS, in the installation document the
following module is not mentioned:
perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm
But the RPM is in the CentOS 4 directory.
On CentOS 4 the rpm is even already present albeit an older
Anyone have one of these yet?
http://www.dlink.com/products/?pid=451
-Dan
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