Displaying 20 results from an estimated 600 matches similar to: "CIsco Gateway recommendation"
2004 Dec 18
4
Free World Dialup and Asterisk
Hi forum,
I have been fighting days and days configuring FWD and asterisk with NO success
I have the following scenario.
My sister in Spain with FWD dialup client
My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone.
Spain LAN
FWD
2004 Aug 14
3
7960 help
I have 4 7960's that I am trying to get working but 2 of them will not
update to the SIP image on my tftp server like the first ones did.
i keep getting the error on the phone 'Defaulting CM to TFTP server' like it
isn't seeing the *.bin on the server.
are you supposed to have on of those for each phone? would be like cisco et
al to do something like that.
TIA
Jason Kawakami
2004 Sep 26
4
IP Phones ?
Hi guys,
I know this isn't strictly about Asterisk, but it is related...
I am looking to buy a few IP phones, but I don't have a huge budged (hence
why I love Asterisk, its amazing and free !), so I was wondering if anyone
knew where I could get some cheap IP Phones ?
Ideally they should be no more then about ?50 ($90).
Thanks, Paul.
2003 Nov 05
12
Mediatrix 1204
I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following.
PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0
2005 Jan 28
3
FWD and IAX2
Hi,
I had a FWD account set up with asterisk (using SIP) and it was working
fine both ways. I switched to IAX2 and now I can't get incoming calls
from FWD. People who call my FWD number get a "480 - user is not online"
message without any traffic reaching my box. I can call FWD numbers fine
over IAX2.
It seems fwd isn't trying to place the call over IAX2 because it thinks
2005 Jan 14
2
Spandsp....And garble incoming fax
Hello:
I have successfully install spandsp and patch asterisk with it. But when
I received a Fax is garble or shrink. Does any one know why???... Am using a
PRI T100P card to receive the fax and save it to a tiff file... Any help
will be greatly appreciated. Here are the versions.
Latest csv from asterisk,
spandsp-0.0.1k.tar.gz
redhat 7.3
T100P has its own IRQ.
Any help will be greatly
2005 Mar 20
1
TAPI
I just installed tapi and some app called identapop pro. I havent tested
incoming calls yet but so far, I cant get calls out using outlooks.
I configured TAPI for asterisk inside outlooks and I set TAPI to these
configs:
TAPI connects using the manager to asterisk without problems.
As channels I configure this:
User channel: SIP/myphone and the phone actually rings when I tell outlook
to dial
2004 Sep 06
6
RES: Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual.
Gonzalo,
I have an APA III-4FXO and I tried using your configurations, I received the
message below:
-- Executing Dial("SIP/2010-edfc", "SIP/2217008@Mediatrix") in new stack
Sep 6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x814bf0c (len 774) to 192.168.199.5 returned -1: Operation not permitted
-- Called 2217008@Mediatrix
Sep 6 16:54:54
2004 Jun 17
3
SJphone regestration problem - Help!
I am having a problem with SJphone registration, having read the list
and wathced it for a while for similar problems. I just can't seem to
figure out the problem.
I tryed to follow a tutorial from
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone,
but in SJphone (SIP tab), I can't find the following setting.
Use local outbound proxy - checked.
Proxy IP Address:
2010 Apr 11
2
(no subject)
Hi Every one,I am a new user for wine. I am trying to log into the winehq forum, but can not login with my mail address and password, and I do not see any kind of link which indicates me register or such things! How can I log into the forum to look for topics? Thanks a lot!
_________________________________________________________________
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2019 Jan 11
2
Solr -> Xapian ?
<!doctype html>
<html>
<head>
<meta charset="UTF-8">
</head>
<body>
<div>
I would recommend making this a standalone plugin for now instead of trying to keep it in core fts.
</div>
<div>
<br>
</div>
<div>
Aki
</div>
<blockquote type="cite">
<div>
On 11
2019 Jan 11
0
Solr -> Xapian ?
I managed to deal with the namespace issue (updated makefile.am)
However, I reach :
../../../src/lib/compat.h:207:19: error: conflicting declaration of
'ssize_t i_my_pread(int, void*, size_t, __off_t)' with 'C' linkage
# define pread i_my_pread
^~~~~~~~~~
../../../src/lib/compat.h:210:9: note: previous declaration with 'C++'
linkage
ssize_t i_my_pread(int fd, void *buf,
2019 Jan 11
0
Solr -> Xapian ?
There is no point into a separate plugin, the purpose is to replace
squat as the default fts (solr being a nightmare)
On 2019-01-11 18:23, Aki Tuomi wrote:
> I would recommend making this a standalone plugin for now instead of trying to keep it in core fts.
>
> Aki
>
>> On 11 January 2019 at 18:40 Joan Moreau via dovecot < dovecot at dovecot.org> wrote:
>>
2004 Sep 28
1
Help with Call Waiting!
Help! I am having a problem getting wall waiting to work.
I have a X100p and a TDM400 card with and analog phone attached (as well as
a number of SIP phones) running on RedHat 9.0 box. The PSTN line has the
call waiting feature. If I am on the phone, I get a beep indicating a second
call is waiting. However, there is no way, no how, that I can get it to
switch to the second line! Flash button,
2019 Jan 11
2
Solr -> Xapian ?
Also,
1 - WHat does represent "subargs" in mail_search_args
2 - I made my first code, and the error I get compiling within the
dovecot architecture is
"In file included from fts-xapian-plugin.c:4:
fts-xapian-plugin.h:6:1: error: unknown type name 'using'; did you mean
'uint'?
using namespace std;"
if I remove this, the Xapian library is also complaining
2006 Apr 21
5
installing hieraki help
Ahoy,
I tried to install hieraki and it was a big mess.
1) upload_progress module needed to be installed
2) no test/development server configured in the config file so i had to
add those
3) no log directory, had to add that
4) had to install 2 gems (ok fine)
5) once the server started, go to homepage and
" NameError in Wiki/pageController#index
uninitialized constant Node
RAILS_ROOT:
2006 Apr 24
6
Hi...Please help me
Hi Friends,
I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to
2006 Jan 19
3
Processor Size
Can someone give me an idea of the processor power I will need for 1 x
TDM240 with 2xquad FXO's and 8 sip phones/ATA's on a quite 100Mbit LAN.
The machine we have available of hand is a P4 1GHz with 768MB RAM.
Tx
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093
2004 Sep 23
3
Help with strategy for echo cancellation.
I have just installed * (RH9, P4 3.0GHZ, 1G RAM) in a small office,
using three TDM400's with 4 FXO's each for incoming calls. Outgoing
calls are (for the moment) routed via VoicePulse. Phone sets are Cisco
7940G's using SIP. I'm getting intermittent echo on outgoing calls, and
my understanding, based on reviewing the wiki and several posts here, is
this:
>>>> The
2004 Jun 07
2
Mediatrix 1204 Configuration
I added those lines to my configuration, and i just see with ethereal that my client dial
and the 1204 led turn on and they started to interchange packets, im newbie with asterisk
i have been trying another sip server with mediatrix that work so well, but i dont know how to set it up?
could u send me all the configuration i need step by step?
----- Original Message -----
From: "Wojciech