try
http://www.digitnetworks.com/X100P_FXO_PCI_Card_p/x100p.htm
On Wed, 3 Jan 2007, Tzafrir Cohen wrote:
> On Tue, Jan 02, 2007 at 10:25:45PM -0500, cb wrote:
> > On Jan 2, 2007, at 10:15 PM, blackwater dev wrote:
> >
> > >I have trixbox working how I want. How do I now (cheaply as
> > >pos
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Richard Lyman
> Sent: Saturday, December 30, 2006 3:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Dialed Number missing from the CD
On Wed, 3 Jan 2007, Tzafrir Cohen wrote:
P[ 0] -- mISDN Channel Driver Registred -- (BE AWARE THIS DRIVER IS
EXPERIMENTAL!)
..Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10603 setup_zap: Unknown
signalling method 'bri_cpe_ptmp'
our Asterisk is not bristuffed. And you don't expect to use ZapBR
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Trevor Peirce
> Sent: Tuesday, January 02, 2007 5:57 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Double quotes in CDRUserField?
>
>
On Tue, Jan 02, 2007 at 10:25:45PM -0500, cb wrote:
> On Jan 2, 2007, at 10:15 PM, blackwater dev wrote:
>
> >I have trixbox working how I want. How do I now (cheaply as
> >possibly) get a phone number so people can call it from any
> >number? I am just doing a prototype so just want it done
On Tue, Jan 02, 2007 at 11:10:56PM +0100, Remco Barendse wrote:
> On Fri, 29 Dec 2006, Julian J. M. wrote:
>
> >It's not necessary to recompile the kernel for mISDN support. Check
> >http://www.laimbock.com/asterisk/
> >
> >Grab the mISDN source rpm, and build it.
> >
> >$ wget
> >http://www.xs4a
I agree with this... The cheapest way is to do this without anymore
hardware. Grab a pay-as-you-go VoIP provider (VoIPJet, Unlmitel, Gizmo
Project, etc.) and setup a trunk. They'll give you a number callable from
the PSTN, and that's all you need. The setup you have already can handle a
voip trunk
Your best option will be to contact a local Asterisk integrator - and
get them started on the work.
PaulH
On Tue, 2007-01-02 at 12:12 -0800, Khaled wrote:
> Dear Guys
> Merry Christmas and happy new year .
>
> Please do any one knows from where I can buy a full pbx corporate cd and
> integrate
To go nice and cheaply, you could just get a free number from
IPKALL.com or Stanaphone.com.. And do it all over IP...
-t-
On Jan 2, 2007, at 10:15 PM, blackwater dev wrote:
Ok,
I have trixbox working how I want. How do I now (cheaply as
possibly) get a phone number so people can call
From: "blackwater dev" <[EMAIL PROTECTED]>
Ok,
I have trixbox working how I want. How do I now (cheaply as possibly) get
a
phone number so people can call it from any number? I am just doing a
prototype so just want it done cheaply so I can demo it to my supervisors.
Thanks!
Add an FXO to
On Jan 2, 2007, at 10:15 PM, blackwater dev wrote:
I have trixbox working how I want. How do I now (cheaply as
possibly) get a phone number so people can call it from any
number? I am just doing a prototype so just want it done cheaply
so I can demo it to my supervisors.
I just went thr
DiegoF wrote:
> chan_vpb.o:chan_vpb.cc:(.text+0x4da6): first defined here
> /usr/bin/ld: Warning: size of symbol `load_module' changed from 3274 in
> chan_vpb.o to 3926 in chan_vpb.oo
> collect2: ld devolvi el estado de salida 1
> make[1]: *** [chan_vpb.so] Error 1
> rm chan_vpb.o
> make: *** [chan
Its Called Off Hook Call Announcement And Asterisk In 1.2 Can Not Do
This I Dont Know About 1.4
This would indeed be off-hook announcement - doesn't call waiting use this
feature already? Can't see much difference from the description - even the
big boss won't like forced barge-in if his hors
Ok,
I have trixbox working how I want. How do I now (cheaply as possibly) get a
phone number so people can call it from any number? I am just doing a
prototype so just want it done cheaply so I can demo it to my supervisors.
Thanks!
___
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Chris how would I use 'verbose' in a dialplan context?
A sample line?
Larry
Chris Tooley wrote:
If you mean in the dialplan, you can use NoOp or verbose (verbose being
something that will get logged too), and if you mean in the asterisk code,
there are logging examples all over the place.
-
Michael Collins wrote:
Question: I'm trying to put a double quote into the CDRUserField. What
I end up with is a pair of double quotes. Example:
exten => s,n,SetCDRUserField(data)
exten => s,n,AppendCDRUserField(")
exten => s,n,AppendCDRUserField(moredata)
My record will look like this:
data""
Haven't yet. Gotta wait until the calls stop flowing in/out. It's a
production system. That's on the list of tonight.
Bill
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore
- Aspendora
Sent: Tuesday, January 02, 2007 8:36 PM
To: A
On 1/2/07, Bill Gibbs <[EMAIL PROTECTED]> wrote:
Echo cancel: yes (and zap show channel confirms it's enabled)
I would think if echo cancel was the problem incoming faxes would fail as
well?
This is only a guess. The Sangoma is detecting the fax when it receives it,
and is turning off echo
Its Called Off Hook Call Announcement And Asterisk In 1.2 Can Not Do
This I Dont Know About 1.4
On 1/2/07, Carlos Chavez <[EMAIL PROTECTED]> wrote:
I have a customer that is asking for a feature called On Hook Call
Announcement. The way he explains it is that when someone is on another
Vulpes Velox wrote:
Jan 3 08:05:03 NOTICE[66269]: app_dial.c:1056 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
I end up getting this when I call from 2000 to 2001.
2000, 2002, and 2001 all exist in sip.conf and I connect using them.
I have all thr
Remove from zapata.conf the lines re bristuff (bri_cpe_ptmp, etc).
Setup misdn:
/etc/init.d/misdn-init config
vi /etc/misdn-init.conf(check it's ok, NT or TE, PTP or PTMP...)
/etc/init.d/misdn-init start
chkconfig --add misdn-init
Setup chan_misdn, in /etc/asterisk/misdn.conf. At the end:
[t
Question: I'm trying to put a double quote into the CDRUserField. What
I end up with is a pair of double quotes. Example:
exten => s,n,SetCDRUserField(data)
exten => s,n,AppendCDRUserField(")
exten => s,n,AppendCDRUserField(moredata)
My record will look like this:
data""moredata
What I want is
Nikola,
Check the maxlen parameter for the queue... Also check the sample
queues.conf distributed with Asterisk source, which somehow includes
queue parameter documentation.
If set, maxlen will limit the number of calls in the queue.
Cheers,
--
Ex Vito
On 1/2/07, Nikola Ciprich <[EMAIL
Jan 3 08:05:03 NOTICE[66269]: app_dial.c:1056 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
I end up getting this when I call from 2000 to 2001.
2000, 2002, and 2001 all exist in sip.conf and I connect using them.
I have all three setup to use the fr
I have a customer that is asking for a feature called On Hook Call
Announcement. The way he explains it is that when someone is on another
call you can sort of break in into their conversation but only the local
person hears you and not the external caller.
Basically he wants to
Hi, Anyone knows where to get the admin (not the end user) manual for
the linksys spa2102. This model is the 2 analog port+router.
There are a lot of advanced options that I would like to see what they do.
Thanks,
--
Erick Perez
Panam
On Fri, 29 Dec 2006, Julian J. M. wrote:
It's not necessary to recompile the kernel for mISDN support. Check
http://www.laimbock.com/asterisk/
Grab the mISDN source rpm, and build it.
$ wget
http://www.xs4all.nl/~pjl/downloads/asterisk/srpms/mISDN-cvs20061107-2_fc6.lc.src.rpm
$ rpmbuild --re
On Tue, 2 Jan 2007, Colin Anderson wrote:
ASUS motherboards, in particular, have worked for me perfectly, everytime
with both Digium and Sangoma cards. They are also easy to work with and well
documented.
I'd second that. I've been using Asus motherboards for over 10 years now
in various LAMP
Hello,
I'm using asterisk queues, for reception phone, and I have small problem: I
have only one phone as queue member, and the problem is, that ALL channels
waiting in queue are ringing on it. And if there are too many people ringing on
it, it's not possible to use attended transfer then...
Is
Do you know If its possible to do the same with Dock and Talk and an
ATA GrandStream HandyTone 386?
Thanks
Joao Pereira
Jonathan Attwood wrote:
I use a Dock-n-Talk in conjuction with a Sipura SPA3000 & Asterisk.
Because I'm using Asterisk, I cannot use voice dialling, however
inbound & out
Follow up:
I used my Cisco 3660 that's a hop away and connected to a different PRI
provider.
Faxes work _fine_
>From the ATA box
I faxed a DID that would come back into the Zap enabled Asterisk server, then
talks BACK to the server that the ATA box is regstered via IAX2 (or SIP, I
found they bo
- Lee Jenkins <[EMAIL PROTECTED]> wrote:
> Thanks to all for the feedback. I have created a wiki page here:
> http://www.voip-info.org/wiki/view/DTSwift+Cepstral+AGI+Wrapper
> http://preview.tinyurl.com/yl9utq
>
> I will host it on my company website for now. Seems like a small
> project to
Thanks to all for the feedback. I have created a wiki page here:
http://www.voip-info.org/wiki/view/DTSwift+Cepstral+AGI+Wrapper
http://preview.tinyurl.com/yl9utq
I will host it on my company website for now. Seems like a small
project to bother with SF.net.
--
Warm Regards,
Lee
Wow Doug thanks for the specs. This has really helped.
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ASUS motherboards, in particular, have worked for me perfectly, everytime
with both Digium and Sangoma cards. They are also easy to work with and well
documented.
-Original Message-
From: Doug [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 02, 2007 1:04 PM
To: Asterisk Users Mailing Lis
Bob,
It looks like the gnutls development package is called gnutls-devel:
'yum install gnutls-devel' should get the package installed.
Yah, I thought that would be it. I have that installed, as well as
gnutls. (I basically installed both packages you can find with yum
search gnutls). Any other
I am able to get this script to dial, but I am unable to talk or hear
anything. The script asks for the number to call and the the caller id to
display (if user is not at their normal extension). Once submitted, the
external extension receives a call, once answered the call is then placed to
the d
On 2006-12-21 13:29:47 -0800, cb <[EMAIL PROTECTED]> said:
Has anyone used either the 8 port or 4 port FXO device from
Grandstream? (GXW-4108 or 4104).
They seem to be the lowest cost multi port FXO devices that I can
find, so I'm getting ready to buy the 8 port version. I just want to
se
At 07:20 1/2/2007, Mark Greene, wrote:
Hey guys,
In your experience what is the best way to go for a production
asterisk box in your offices? With desktop prices so cheap you might
think that you should just buy them off the shelf, but is that
really a reliable machine? Anything you can tell
Hi Thiru -
Please clarify one more doubt in extensions.conf file...
is the following dial plan is right way to call another server(frome
serverA to serverB)
exten => _5X,1,Dial(sip/[EMAIL PROTECTED]:6030,15,tr)
exten => _5X,2,Hangup
You can dial either via IP or by sip device nam
2 Asterisk servers 1.2.12.1
Connected via IAX2, same switch, GigE, no packet loss, etc
1 with a Sangoma A101 for a PRI to the PSTN
Ulaw
QoS enabled
NAT for the registered ATA boxes, no nat between the * servers
Faxing inbound:
Call from PRI hits the first Asterisk server
Then talks to th
Hi all,
Special thanks to David and Noah for the earnest efforts...
Dan
On 02/01/07, Dave Schardin <[EMAIL PROTECTED]> wrote:
Your best bet is to contact Sysmaster support at [EMAIL PROTECTED] or
1877-900-3993.
I was talking to one of our contacts there and he said that it would be best
t
> I believe I am going to start out with some refurbished Dell Poweredge
> servers. They have had a high success rate with a friend.
One word of caution: some have had various hardware issues getting
certain telephony cards to work with certain Dell PowerEdge servers. If
you aren't going to hav
Has anyone made this combination work together? I've tried everything
and can't seem to get it work right. It all compiles fine, but when
rxfax is called, I get an unknown symbol error. From my reading,
everything points to me having multiple copies of spandsp and it's maybe
calling the wron
Bruce Reeves wrote:
After skimming over your readme file I thought I would ask, how does
this app differ from passing the parameters to the swift program using a
System dial plan command? You mention having cepstral play back a text
file in a certain voice, which I have done from the dialplan w
Mark Greene<[EMAIL PROTECTED]> Wrote on: 1/2/2007 12:58 PM:
> I believe I am going to start out with some refurbished Dell Poweredge
> servers. They have had a high success rate with a friend.
I was going to go that route as well. But, depends on the model. I have
several of the Poweredge 2300/
I am a a little confused on how to get h323 working on asterisk. Could you
please point me towards specific resources you used? voip-info.org seems to
keep me in a loop of info.
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asterisk-users m
I believe I am going to start out with some refurbished Dell Poweredge
servers. They have had a high success rate with a friend.
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Your best bet is to contact Sysmaster support at
[EMAIL PROTECTED] or 1877-900-3993. I was talking to one of our
contacts there and he said that it would be best to have you contact
them. In order to get it to work for you they need to know the exact
configuration you are trying to set up
> Also, anyone have suggestion on licensing? LGPL? FreeBSD?
One advantage of LGPL over GPL is that GPL is 'viral' whereas LGPL is not. For
a more in depth discussion please see:
http://www.ugcs.caltech.edu/manuals/devtool/autotoolset-0.11.4/toolsmanual_87.html
In short, if you want anyone to
I would like to show a remark that would show call progress
and appear on the CLI screen.
The remark should be in the code of a sip [channel] or extentions [context]
If I can't send my own remark, what little used 'show' command could I
insert in the code?
Can this be done?
--
Larry Alkoff N2
On Tue, 2 Jan 2007, Mark Greene wrote:
Hey guys,
In your experience what is the best way to go for a production asterisk box
in your offices? With desktop prices so cheap you might think that you
should just buy them off the shelf, but is that really a reliable machine?
Anything you can tell me
--- Olivier <[EMAIL PROTECTED]> wrote:
> Thanks !!
> I've never heard of this one (I mean : I've never
> heard of OptiPoint phones
> to support 802.1x).
>
> Have you used the SIP version with Asterisk and
> 802.1x ?
we have several Optipoint410/420/600 configured with
Asterisk and they seem to w
--- Mark Greene <[EMAIL PROTECTED]> wrote:
> Hey guys,
>
> In your experience what is the best way to go for a production
> asterisk box in your offices?
(In the US) I have had very good luck with Opterons in Tyson rackmounts
bought from Newegg.
Thanks !!
I've never heard of this one (I mean : I've never heard of OptiPoint phones
to support 802.1x).
Have you used the SIP version with Asterisk and 802.1x ?
Am I correct to think that using 802.1x isn't directly of Asterisk concern ?
2007/1/2, richard Coco <[EMAIL PROTECTED]>:
***
This
On Sun, Dec 31, 2006 at 12:44:48PM -0500, Lee Jenkins wrote:
>
> Hey all,
>
> After figuring out a problem with AGI and freepascal, I have finished
> writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a
> small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt.
Hi,
http://www.communications.siemens.co.uk/enterprise/products/optiPoint_410s.htm
rich.
--- Olivier <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Is anyone aware of a wired sip hardphone supporting
> 802.1x authentication ?
> I've been told some Avaya and Alcatel ip phones
> supported 802.1x.
>
> As
I would post it to some site of yours (or Sourceforge if you plan to have
shared CVS) plus a page on the wiki, so people can find it. I have been
working on a few projects on sourceforge and never had problems with it.
With licence, you choose. GPL is usually a good starting point for
lic
After skimming over your readme file I thought I would ask, how does this
app differ from passing the parameters to the swift program using a System
dial plan command? You mention having cepstral play back a text file in a
certain voice, which I have done from the dialplan with the options provide
Hi,
Is anyone aware of a wired sip hardphone supporting 802.1x authentication ?
I've been told some Avaya and Alcatel ip phones supported 802.1x.
As 802.1x is widely used with wireless hardphones, I'm wondering whether or
not, 802.1x could also be valuable for wired environments.
Regards
__
Happy Holidays!
Sourceforge provides free hosting for open source projects. That is where I
would put it if I were me.
For licensing.. I use the BSD license for my creations, but version 2 of
the GPL is "stronger" in my opinion.
Good luck,
Ejay Hire
-Original Message-
From: [EMAIL PRO
I believe there were some new prompts added for 1.4 for Directory Info.
These have now been added to http://www.tel.net
Have a good 2007.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekenn
I had this same problem. It was that I was missing the mysql-devel
package. I installed this on my Fedora Core 4 system with "yum install
mysql-devel". Once I installed this I redid the ./configure, make and
make install of the addons and voila it was there.
Fr
Noah is correct. We will install a trial system with 11 AP. The WiFi
terminal will hold a conversation when moving between APs. Initial tests
with Hitachi IP5000 were ok. We need to test as well PDA and cell/WiFi
phones.
Jorge Mendoza
Noah Miller wrote:
Roaming is irrelevant in VOIP. You just
add a wait before you dial the sip phone, keep in mind the callerid
information arrives later than the call setup info
On Dec 31, 2006, at 4:38 PM, David Sampson wrote:
For some reason something that seems like it should be simple is
leaving me a bit perplexed. I am receiving incoming Call
Hi guys,
How can i save sip debug command output to a file ??
Thanks.
--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
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Hey guys,
In your experience what is the best way to go for a production asterisk box
in your offices? With desktop prices so cheap you might think that you
should just buy them off the shelf, but is that really a reliable machine?
Anything you can tell me that would assist me in deciding the bes
Hi,all
Randomly my line drops and when I look in message log file I always see
the following notice:
NOTICE[14491] chan_zap.c : avoiding deadlock…
The situation appears with no obvious reason, the CLI shows nothing more
than the zaptel channel hanging up. I have a Asterisk 1.2.10 and Zaptel
Kenneth Padgett wrote:
I'm working from the docs here:
http://voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk
and getting an error doing the ./configure on the iksemel module:
checking for getaddrinfo... yes
./configure: line 20399: syntax error near unexpected token `,'
./configure:
Dovid B wrote:
> can u get me the info on the part ?
>
Hi Guys
I have found this. Have not tested as yet, but have asked them for some
more info.
Might be of some help.
www.its-tel.com
Cheers
Terry
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Dear Guys
Merry Christmas and happy new year .
Please do any one knows from where I can buy a full pbx corporate cd and
integrated with exchange server and life communication server .
Regards
*
No employee or agent is authorized to conclude any
Maybe, what is meant is handover.
Cheers
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slave,
have you install asterisk-addons yet?
if you installed and that error still happen, pls find that file, you can also
put them into that path to load.
regards,
osochebol
- Original Message
From: RdBSD <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussi
2006/12/29, Frédéric Marti <[EMAIL PROTECTED]>:
Hi all,
We are looking for information about Dynamic Realtime Asterisk, We have
install some Snom
phone 360 (SIP) for our customer , but we have a problem when we want to
register 2 accounts on the same phone and on the same Asterisk PBX.
The pr
Dear All,
I' I have a problem in installing asterisk 1.4.0. how can i compile
res_config_mysql.c in astersisk-addons dir. I've downloaded
asterisk-addons-1.4.0 compiling and installing it. But i can't found
shared object of res_config_mysql.so.
My system is :
Debian Linux 3.1
Kernel 2.6.8-11
as
Hi,
I configured openh323_v1_18_0, pwlib_v1_10_0 and
asterisk-oh323-0.7.3.
I can call inbound and outbound.
But early media is not working in outboubd.
Regards,
Jason.
oh323.conf
==
[general]
listenPort=1720
connectPort=1720
tcpStart=1
tcpEnd=2
u
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