On Jul 9, 2006, at 10:51 PM, Vincent Delporte wrote:
snip
Thanks for the info. This little experiment is getting expensive ;-)
LOL! I know that feeling... I actually thought I would save money with
VOIP, what a joke! Actually I am happy with my setup, but spent an
intial $75(us) thinking
What are the min cpu requirements for ppciax? Has any one tried ppciax
with Cingular 8125?
On 7/9/06, Administrator TOOTAI [EMAIL PROTECTED] wrote:
Attilla De Groot wrote:
Hi all,
I have two pda's and I want to be able to make calls, but I need a
client for this. The only problem is
ooops,
sorry, you right, forgot to mention it...
It was to be compared with AMD 64.
Olivier
C F a écrit :
Olivier can you please do a cat /proc/cpuinfo and post it here? I
think you have a 64 bit cpu.
On 7/9/06, olivier.taylor [EMAIL PROTECTED] wrote:
Fyi,
Double Intel Xeon 3Ghz
On 7/10/06, Ryder Brook [EMAIL PROTECTED] wrote:
and learning a lot and the stupid mistake was that the telephone that I was
calling from has caller id blocked. Well, the only satisfaction is that I
You should always have a way to test with a call that you know is
working, such as a cell phone
Dear
I want to make billing recharge through receiving digits
from IVR through dtmf and store it on a text file ,
How can I do that ?
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of
Martin Joseph wrote:
On Jul 9, 2006, at 10:51 PM, Vincent Delporte wrote:
snip
Thanks for the info. This little experiment is getting expensive ;-)
LOL! I know that feeling... I actually thought I would save money with
VOIP, what a joke! Actually I am happy with my setup, but spent an
Dear
I am in need urgently to upgrade my [EMAIL PROTECTED] from 2.6
to trixbox or 2.8 how can I do that .
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party
On Jul 9, 2006, at 11:44 PM, Rich Adamson wrote:
Martin Joseph wrote:
On Jul 9, 2006, at 10:51 PM, Vincent Delporte wrote:
snip
Thanks for the info. This little experiment is getting expensive ;-)
snip
Rumor has it that Digium will be announcing some new cards in the near
future that
On Sun, Jul 09, 2006 at 05:56:49PM -0400, C F wrote:
Thanks for that Tzafrir. Why does it ignore the secend CPU?
'show translations' is done by a loop that for each pair of codecs
meassures the time it takes to convert a relatively short ammount of
data between the two.
Thus each conversion is
I've been using *8# on my 7960's to pickup ringing phones in the office.
Anyone been able to do call pickup from a spa941?
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On Mon, Jul 10, 2006 at 09:47:34AM +0300, Khaled Chehab wrote:
I am in need urgently
Hire someone to do that if it is that urgent?
to upgrade my [EMAIL PROTECTED] from 2.6 to trixbox or 2.8
how can I do that .
Ask in the [EMAIL PROTECTED] / trixbox mailing list(s)?
--
Tzafrir Cohen
Have a look at cyber-telecom.net. CT-GSM-1000
seems to be one of the cheapest GSM Gateway that you can buy right now.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Tuesday, June 27, 2006 11:41
PM
To: 'Asterisk
Users Mailing List -
Rich Adamson wrote:
I've been using *8# on my 7960's to pickup ringing phones in the office.
Anyone been able to do call pickup from a spa941?
Disregard; dumb mistake on my part. Forgot to add pickup to the sip.conf
definitions for the extension.
One of my customers decided to allow me to make a test system for a fax server.
So far I have searched the wiki and came up with Hylafax(standalone or
with IAX) and astfax (integration with asterisk).
Scenario:
Customer has windows machines (500+) and we want to try a fax server
in the
On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote:
AS with Hylafax, it seems that I need to install an IAX modem in every
machine (arrrggg) or define a printer driver.
You need to install an iaxmodem on the machine where the hylafax server
is installed. Which can probably be the
So I install a machine with linux+hylafax+asterisk+iaxmodem+TDM400 (4fxo)
and then how the windows clients send email-to-fax to the above machine?
On 7/10/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote:
AS with Hylafax, it seems that I
Hello
If you look at hylafax.org, you can find several windows clients for Hylafax.
http://www.hylafax.org/content/Desktop_Client_Software
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Erick Perez
Sendt: 10. juli 2006 09:52
Til: Asterisk Users
We used to put one of the hylafax printer drivers on each windows box -
which is not much fun.
PaulH
On Mon, 2006-07-10 at 02:52 -0500, Erick Perez wrote:
So I install a machine with linux+hylafax+asterisk+iaxmodem+TDM400 (4fxo)
and then how the windows clients send email-to-fax to the above
Hi all,
I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also have 2 Digium FXO cards, and
I have premicells connected to the FXO's . Calls come in off the Sangoma E1 cards, from a Philips
PABX. The problem I have is that the user, when he dials from his desk phone, does not
Anybody who knows a good source of AGI tutorials on the net? plz share-- RegardsRizwan HishamSoftware Engineer
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We used?
what are you doing different now?
On 7/10/06, Paul Hales [EMAIL PROTECTED] wrote:
We used to put one of the hylafax printer drivers on each windows box -
which is not much fun.
PaulH
On Mon, 2006-07-10 at 02:52 -0500, Erick Perez wrote:
So I install a machine with
A different job
PaulH
On Mon, 2006-07-10 at 03:34 -0500, Erick Perez wrote:
We used?
what are you doing different now?
On 7/10/06, Paul Hales [EMAIL PROTECTED] wrote:
We used to put one of the hylafax printer drivers on each windows box -
which is not much fun.
PaulH
On
Hi,
Does Any one has experience or know any open source client for AstManproxy?
My main goal is to monitor every started and hung up call into a CDR,
but with particular features:
- Every call is started via AMI with Originate command.
- I wanna keep record of the brigded call and both calls:
Hi,
I'm using res_perl with asterisk 1.0.0.
And after running asterisk a couply of months, I see that the process asterisk
take a lot on memory.
And asterisk will freeze.
If I look in the logging I see that the
last command asterisk perfomed is a call to a perl program.
So I think
On Jul 10, 2006, at 1:23 AM, yusuf wrote:
Hi all,
I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also
have 2 Digium FXO cards, and I have premicells connected to the FXO's
. Calls come in off the Sangoma E1 cards, from a Philips PABX. The
problem I have is that the
Hehe, ok.
Thanks,
On 7/10/06, Paul Hales [EMAIL PROTECTED] wrote:
A different job
PaulH
On Mon, 2006-07-10 at 03:34 -0500, Erick Perez wrote:
We used?
what are you doing different now?
On 7/10/06, Paul Hales [EMAIL PROTECTED] wrote:
We used to put one of the hylafax printer
Hi,
Maybe this is dirty but this is how I did it (with capi but you can
probably do it with anything you want):
***Suppress the Hisax drivers in conflict with capi:
[EMAIL PROTECTED]:~# mv
/lib/modules/2.6.12-9-386/kernel/drivers/isdn/hisax/hisax.ko
On Mon, Jul 10, 2006 at 10:59:20AM +0200, Arjan Kroon wrote:
Hi,
I'm using res_perl with asterisk 1.0.0. And after running asterisk a
couply of months, I see that the process asterisk take a lot on memory.
And asterisk will freeze.
If I look in the logging I see that the last
After using the trunk versions as below, it all compiled ok, and the polycom
acd is working great, but the music on hold and meetme will now work. I do
not have any digium cards, is the ztdummy installed with the truck version?
Or is there any thing I need to change?
Thanks,
Dean.
-Original
Thanks,
Can you maybe give me an example of such a build-in option sebuuging.
Arjan Kroon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: maandag 10 juli 2006 11:19
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
Hi list,
I get this message sometimes ( randomly ) when queues are calling agents:
Jul 10 11:26:46 ERROR[8856]: app_dial.c:1481 dial_exec_full: Could not
stop autoservice on calling channel
I'm trying to see where it comes from ...
Does someone has an idea ???
Thanks in advance !
Erick Perez wrote:
AS with Hylafax, it seems that I need to install an IAX modem in every
machine (arrrggg) or define a printer driver.
This is incorrect.
Check out http://iaxmodem.sourceforge.net
Doug
-- Ben Franklin quote: Those who would give up Essential Liberty to
purchase a
Hi,
Is it possible to encrypt the conversation between two parties on SIP,IAX or ZAP channels?
Zeeshan
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Hi,
I am trying to setup fax on my phone system. Which fax-to-email and email-to-fax solution really works on IAX and SIP?
Zeeshan
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Rizwan Hisham wrote:
Anybody who knows a good source of AGI tutorials on the net? plz share
How about the Asterisk Wiki?
Flynn
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It really depends on the programming language you plan to use. I'd have a
look at the PHPAGI first, but there is not much as to AGI per se as with
the underlying programming language on one side and understanding Asterisk
on the other
Hope this helps
l.
On Mon, 10 Jul 2006
Hi,
Is it possible to encrypt the conversation between two parties on SIP,IAX
or
ZAP channels?
Sure, setup a VPN.
You can get a Linksys VPN router for less than $100 and run whatever
protocol you like over your VPN.
--
Henry J. Cobb
http://www.io.com/~hcobb/
Hi,
I was wondering whether anyone has any input into the reliability of
faxing (over a PRI) using spandsp and rxfax.
99% of times this is a reliable combination - we use it almost
exclusively, but there seem to be certain fax devices which have
problems talking to us. Most notably fax modems,
On Mon, Jul 10, 2006 at 03:14:27PM +0800, Sam Tam wrote:
Have a look at cyber-telecom.net. CT-GSM-1000 seems to be one of the
cheapest GSM Gateway that you can buy right now.
Which is biz, and Sam works for Cyber-Telecom ...
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20
I believe that these sip phones only work with cisco call manager. Only
the 7950 and 7960 have an open sip stack
Per Møller wrote:
After google’ing extensively, I now have sip firmware (8.0.2SR1/8.0.3)
running on the 7941, 7961 and 7971 and I even have a SEP.cnf.xml that
seems to have
My understanding is that asterisk will read the file before it is finished
being written.
The proper method for NFS would be to write into another folder on the same
file system, then have a script move the file to the
proper call file location.
The script would run every 5 seconds or so.
The
And you might as well sell them Internet access as well, because some of them
may try to use dialup Internet over that VOIP
connection, which will fail miserably. (see all of the fax threads for
reference.)
--
--
Steven
http://www.glimasoutheast.org
Cory Andrews [EMAIL PROTECTED] wrote
Actually this seems to have fixed it!!
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Sawa
Sent: Sunday, July 09, 2006 10:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Choppy MOH (Cisco
The metermaid changes in head are very different, but there is a working
1.2.7.1 patch in the bug tracker.
http://bugs.digium.com/view.php?id=5779
I believe that the 1.2.7.1 patch also works with 1.2.9.1.
--
--
Steven
http://www.glimasoutheast.org
Matt [EMAIL PROTECTED] wrote in message
Hi,
Is it possible to encrypt the conversation between two parties on SIP,IAX
or
ZAP channels?
Sure, setup a VPN.
You can get a Linksys VPN router for less than $100 and run whatever
protocol you like over your VPN.
Agreed. I have seen and heard of a lot of attempts to bring SRTP support
Hi Marnus,
That is a good idea, I didnt think
of thatJ
thanks
Dinesh.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marnus van Niekerk
Sent: Thursday, July 06, 2006 4:59
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re:
I am not for the billing part, as its sip based, and its educational calls
only. I mean between sip.edu community and my educational institute. So
practically any sip uri should be able to be dialed from the website. I
dunno I am just asking the ideas for the group.
Regards,
Dinesh.
And of course I just found this article
http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3
Hope this helps some other people out as well!
Bill
-Original Message-
From: Bill Gibbs
Sent: Monday, July 10, 2006 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial
Has anyone here tried to use zphone with SIP soft phones and Asterisk?
Michael
On Mon, 10 Jul 2006 07:35:34 -0400, Raymond McKay wrote:
Hi,
Is it possible to encrypt the conversation between two parties on SIP,IAX
or
ZAP channels?
Sure, setup a VPN.
You can get a
On 7/10/06, Doug Lytle [EMAIL PROTECTED] wrote:
Any pointers on how to diagnose or improve this would be appreciated.
Install HylaFAX and iaxmodem on your Asterisk box.
Thanks, I will do.
I assume that iaxmodem talks to the PRI, and then HylaFax talks IAX to
asterisk? Any
Maybe in Asterisk 1.4 SecureRTP application would do that.
Regards
Henry J. Cobb escribió:
Hi,
Is it possible to encrypt the conversation between two parties on SIP,IAX
or
ZAP channels?
Sure, setup a VPN.
You can get a Linksys VPN router for less than $100 and run whatever
protocol you like
Hi Raymond,
Raymond McKay wrote:
Agreed. I have seen and heard of a lot of attempts to bring SRTP
support into Asterisk but the idea of SRTP just doesn't make sense to
me. Asterisk, and VoIP servers in general, are meant to be
communications services not security services. In my mind at
You may need to recompile now that you've got zaptel/ztdummy
installed so that your install sees that the proper zaptel exists now.
On 7/10/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote:
After using the trunk versions as below, it all compiled ok, and the polycom
acd is working great, but the
Hello
Can any one may send me log when channel bank is work
Best regards
Viktor Tatianin
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Joe Baptista wrote:
On Sun, 9 Jul 2006, Andrew D Kirch wrote:
To some extent I see your point and have been on the receiving end of
one of Jeremy's tirades.
I've since decided that NuFone is an interesting study in whether your
business can survive
with only clueful customers.
Some
Is this correct:
zaptel: make clean; make; make install
asterisk: make clean; make; make install
Will this recompile everything needed? I tried, but the meetme app still
does not get compiled (and no music)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
Are you talking about ZiPhone a USB device
?
Mike
Simple Simon
http://www.simplesimon.com
- Original Message -
From:
Michael Graves
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, July 10, 2006 6:44 AM
Subject: Re: [asterisk-users]
Hi,
I set the sip.conf parameter call-limit=1 to limit outbound calls and
'disable' call waiting.
But internally, I want to enable transfers. If the call-limit=1, the
transfers fails.
Any help ?
Thanks all,
Alexandre
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On Mon, 2006-07-10 at 07:34 -0500, Mike Bates wrote:
Are you talking about ZiPhone a USB device ?
Mike
zphone is phil zimmermans (creator of pgp) encrypted rtp system. Unlike
SRTP this does not rely on the server itself to provide the encryption.
It also lets you be reasonably assured
Hi friends,At present, I am making outgoing calls using Teliax service with Asterisk. But, I am unable to receive calls. My DID number is: 3031234567. I am using SIP Server (Asterisk) setup, which is provided on Teliax website support. I have replaced my DID number i.e., 3031234567 in YOURNUMBER.
I think it's the same,
10 calls in 200ms = 50 calls in 1s
because 1s = 5 x 200ms
IMHO, is better to use seconds as period, because is more ease to compare
rate speeds of each codec that are in bits per second.
fabio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL
I can confirm that the 1.2.7.1 patch works with 1.2.9.1 as well.
On 7/10/06, Steven [EMAIL PROTECTED] wrote:
The metermaid changes in head are very different, but there is a working
1.2.7.1 patch in the bug tracker.
http://bugs.digium.com/view.php?id=5779
I believe that the 1.2.7.1 patch also
Steve Davies wrote:
I assume that iaxmodem talks to the PRI, and then HylaFax talks IAX to
asterisk? Any downsides/gotchas to this that I should be aware of?
No,
iaxmodem gives HylaFAX software modes that can also communicate with
Asterisk.
From the iaxmodem home page:
IAXmodem is a
On Friday, June 23, 2006 4:08 PM Steven wrote:
Exchange changes
http://www.microsoft.com/exchange/techinfo/tips/mailtip01.asp
Looks promising and helps a bit. Still no use of precedence bulk etc. though.
Very poor detection of lit mails.
___
Hi
first:
exten = 3031234567,1,Answer()
exten = 3031234567,2,DIAL(SIP/user,20)
if this still don't work try
exten = _3031234567,1,Answer()
exten = _3031234567,2,DIAL(SIP/user,20)
second:
You have in sip.conf [teliax] configured, did You specify context=
?
if yes, then
Haven't read this whole thread (got way behind in this list :) )
Polycom has a softphone with video support also. Not sure if it is good
or not, just downloaded the trial version to test it out.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Hi,
I have configured digium tdm04b card with asterisk
on debian. Incoming call is ok. But outgoing call has problem. Would you give me
advice ?
Here is my config files.
zaptel.conf
fxsks=1fxsks=2fxsks=3fxsks=4
loadzone=usdefaultzone=us
zapata.conf
[channels]language=en
Are you sure they are sending you all 10
digits and not just the last four? Our provider just sends the last four digits
on DID. If this is the case you would have this:
exten = 4567,1,Answer()
exten = 4567,1,DIAL(SIP/user,20)
Hope this helps.
From:
[EMAIL PROTECTED]
I'm not a a guru, but
Check this line:
exten = _9.,2,Dial(Zap/g1/${EXTEN})
do you really want to dial digit 9 through your ZapLine? are you
connected to another pbx?
If you don't want do dial 9 to PSTN line , but you want your users to
dial 9 to place outgoing calls, try this:
exten =
I am using asterisk 1.2.9.1.
I had been using the option D to send some dtmf tones after the call is
answered.
This doesnt seem to be working for me now.
I am using and IAX2 connection from one machine to another.
my extensions.conf has:
exten= 57,1,Dial(IAX2/boxa_to_boxb/597,,tD(101))
Hi all,I'm stuck here! I am trying to get DUNDi to work and it seems DUNDi is working accept the IAX part I think. I'm trying to let an extension from Trixbox1 call an extension on Trixbox2 with the use of DUNDi.
Ext 1301 * TrixBox1 * ---IAX2 * TrixBox2 *---Ext
Hi List,
Just a little question about QueuePauseMember()
I use it in the manager with the following action:
Action: Command\r\n
Command: PauseQueueMember(|Agent/$id)\r\n\r\n;
where $id is the agent's ID
but the agent is still taking calls...
Do I have to use the phone number agents
Hi
I've setup asterisk as client, it was working very fine 2 months
before, but now there is no audio on both side, as i'm on same network
as provider.
My IP = 111.111.20.*
Provider Host IP = 111.111.10.10
sip.conf
[general]
context=sip-incoming
bindport=5060
bindaddr=0.0.0.0
insecure=very
Hi everyone,
I have Asterisk SVN-trunk-r7498 running for a few months and
I'm quite
happy with it. However, I am experiencing a quality issue
with my AVM
Fritz!card PCI which is used with chan_capi. When somebody
calls me on
this line he hears a lot of noise
Sorry Misreading of the wiki,
I have to use instead
Action: QueuePause
Interface: Agent/ID
Pause: 1|0
I write it to the mailing list so that people will be aware of the way
to use this manager command ;)
Tristan a écrit :
Hi List,
Just a little question about QueuePauseMember()
I use it
Martin Joseph wrote:
On Jul 10, 2006, at 1:23 AM, yusuf wrote:
Hi all,
I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also
have 2 Digium FXO cards, and I have premicells connected to the FXO's
. Calls come in off the Sangoma E1 cards, from a Philips PABX. The
problem
hI,
I've got problem with zaphfc kernel module. After I load into kernel i
receive something like that into syslog:
0 16:56:42 viperprint-de kernel: zaphfc: empty HDLC frame or bad CRC
received (framelen = 5, stat = 0xff, card = 0).
Jul 10 16:56:42 viperprint-de kernel: zaphfc: empty HDLC frame
I have an image in CIP format that i'm trying to load onto a 7960 phone using
sccp. I don't know where to reference the data in which config file. Any
help available?
--
Edward F. Klimowicz
Voicenet Systems Administration
[EMAIL PROTECTED]
215.259.2131
Hi Marcin,
Marcin J. Kowalczyk wrote:
hI,
I've got problem with zaphfc kernel module. After I load into kernel i
receive something like that into syslog:
0 16:56:42 viperprint-de kernel: zaphfc: empty HDLC frame or bad CRC
received (framelen = 5, stat = 0xff, card = 0).
Jul 10 16:56:42
Title: Message
I want to allow a
SIP caller to place multuiple consecutive calls -
So a caller connects
to Asterisk and gets routed to a destination with the Dial
command.
After the call
completes I would like to let them optionally enter a new destination.
Currenty the call always
I'm trying to provide dial tone on EM Wink type trunks. I found where
in source, 'chan_zap.c' where I believe the code needs to be added.
Basically I believe I can copy parts used for PRI in to EM and EM Wink
signal types.
However with my attempts, it fails to compile at chan_zap. And I'm not
Zeeshan Zakaria wrote:
I am trying to setup fax on my phone system. Which fax-to-email and
email-to-fax solution really works on IAX and SIP?
Due to the way that your question has been asked I assume that you're
talking about IAX or SIP over a latent internet connection to some VoIP
Yes, I am using the 1.2.7.1 patch on 1.2.9.1. It seemed to work fine.
Still curious if anyone has this working on an Aastra phone? I can't get it
to work but someone in the bug.digium.com list said they had it working on
an Aastra phone. Maybe I am missing something. I tried just about
Hi Everyone,
I was wondering if anyone had any ideas regarding this. I can see in the sip debug that music on hold is called when a person is put on hold, however I hear nothing. Any help would be appreciated.
Thanks
Julian
From: [EMAIL PROTECTED]To: [EMAIL PROTECTED];
Hi,
Has anyone used the cooker RPM for asterisk version 1.2.9? I would like to hear some feedback before I install it.
Thanks
Julian
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Metermaid phone compatibility Date: Mon, 10 Jul 2006 09:02:37
There is an old, very old document that I found somewhere that this
PoE switch was designed for NBX phones at that time.
Does anybody in this list is using this switch with non-3com NBX PoE phones?
--
Erick Perez
Panama Sistemas
Kai Fürstenberg napisał(a):
If you want to connect a telephone to the HFC card you need a crossed
cable to connect to an NTBA to which you connect the phones.
i'm connecting to ISDN-NTBA from T-Com.
As far as I know you don't need a crossed cable when you connect the
phone directly.
Have
Julian Varanini wrote:
Hi,
Has anyone used the cooker RPM for asterisk version 1.2.9? I would
like to hear some feedback before I install it.
I haven't, I find it just to easy to compile it under Mandriva.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to
shadowym wrote:
Yes, I am using the 1.2.7.1 patch on 1.2.9.1. It seemed to work fine.
Still curious if anyone has this working on an Aastra phone? I can't get it
to work but someone in the bug.digium.com list said they had it working on
an Aastra phone. Maybe I am missing something. I tried
Hi Doug,
I thought of that as well. I am not a total n00b with Mandriva, just enough to be dangerous. Do you have any walk throughs that I could use as a guide? Do you create an asterisk user for it to run under? What other software should I install in order for it to not only compile properly
I have a asterisk box up and running great. I have another house in my
backyard that also wants to use my asterisk box. I am running trixbox
now and have two POTS lines connected to digium TDM400P as well as 1
voip line for long distance. I would like to keep these two houses as
seperate as
You can place the phones at each house in a different context. Trunks, too.
On 7/10/06, Andrew Niemantsverdriet [EMAIL PROTECTED] wrote:
I have a asterisk box up and running great. I have another house in mybackyard that also wants to use my asterisk box. I am running trixbox
now and have two
Ariel Batista wrote:
Justin Johnson wrote:
Hi All,
I have centOS 4.3 installed and have attempted to install asterisk
separately. I have installed all the modules as suggested on Asterisk
downloads, more (via SVN) However, on the zaptel install I am getting
the following errors.
centosbug
Is that the standard way of doing things? I found a bunch of asterisk
hosting providers in my search on the best way to do this. Is this
what they are doing?
On 7/10/06, Tom Lynn [EMAIL PROTECTED] wrote:
You can place the phones at each house in a different context. Trunks, too.
On 7/10/06,
Yes that is correct.
Bill
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph
Sent: Monday, July 10, 2006 12:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway)
On Jul 10,
But my question is, those that mean that it will take 1 second to
convert 50 channels? if so do I get a 1 second latency when coverting
50 channels?
On 7/10/06, Fabio [EMAIL PROTECTED] wrote:
I think it's the same,
10 calls in 200ms = 50 calls in 1s
because 1s = 5 x 200ms
IMHO, is
I use stable 1.2.9.1 for my servers. How can I maintain my asterisk
1.2.9.1 updated with the patches produced for that release, in case a
patch fits a need?
what should I do in MANTIS to see patches applied to 1.2.9.1?
While looking at MANTIS I just (?) saw one entry for Product build
1.2.9.1
Julian Varanini wrote:
Hi Doug,
I thought of that as well. I am not a total n00b with Mandriva, just
enough to be dangerous. Do you have any walk throughs that I could
use as a guide? Do you create an asterisk user for it to run under?
What other software should I install in order for
On Jul 10, 2006, at 10:48 AM, Andrew Niemantsverdriet wrote:
Is that the standard way of doing things? I found a bunch of asterisk
hosting providers in my search on the best way to do this. Is this
what they are doing?
Yes,l I think that's what contexts are for... I am also relatively new
I have not done a lot of compiling in Mandriva. Did you createa directory for it, e.g. /data/asterisk?
Thanks
Julian
Date: Mon, 10 Jul 2006 15:23:22 -0400 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9
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