Hello,
I'm trying to fill CID data into the astdb using AsteriskWin32's
asterisk.exe, to no avail: The batch file stops after the first line, and
just waits:
rem c:\cygroot\mystuffimport.bat
rem
rem c:\cygroot\mystuffC:\cygroot\bin\asterisk.exe -rx
Hello,
In case there are other users of the AsteriskWin32 port...
I haven't really used the AGI feature of Asterisk to run an application
from extensions.conf. *Win32 supports Perl, which I don't know. Apparently,
it's also possible to write AGI applications as EXE's (there's a
eagi-test.exe
Hello
I'm using a Linksys 3102 as VoIP gateway to connect a POTS line to a PBX. I
also have an IP phone in a remote network across the Net. The PBX +
gateway, and the phone are both behind a NAT router.
I was wondering:
1. When a customer calls us through the POTS line and I pick up the
Hi
To investigate the UNREACHABLE issue I'm having, I need to have
confirmation that it's OK for the Asterisk server to be behind a NAT
router, and also have clients elsewhere on the Net behind their own NAT router?
I know that clients must use STUN to resolve their public IP and punch UDP
Hello
Before I order a Travla C156 case
(http://206.14.132.88/products/Travla/c156/C156.html), a Via mini-ITX
motherboard (either the fanless ME6000
http://idotpc.com/TheStore/pc/viewPrd.asp?idcategory=50idproduct=4 or the
fan-equipped M1
At 10:09 11/02/2007 -0500, Gordon Henderson wrote:
Check the processor spec. carefully. [...] Also make sure you compile
asterisk for an i586
OK, I'll make sure it has enough cache and I'll recompile the code myself.
I'm thinking of getting an ML 8000
At 10:09 11/02/2007 -0500, Michelle Dupuis Henderson wrote:
We use a lot of mini-itx pc's, including the pCI slot. I don't think any
of the systems have shared an irq with the PCI slot
Thanks for the tip. In that case, I have a couple of questions for you :-)
1. The smallest mini-ITX
Hello
If someone had the opportunity of trying those two analog cards, how do
they compare? Digium's sells for $150 while OpenVox's sells for $95.
Thanks.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
At 10:09 11/02/2007 -0500, Gordon Henderson wrote:
On the CN1000 boards I'm using, the PCI slot seems tobe locked to IRQ10.
The on-board USB hardware also seems to be wired to IRQ 10 :-) Using the
BIOS to reserve IRQ 10 caused the on-board USB hardware to move to IRQ5
on the old VIA 533MHz
At 10:09 11/02/2007 -0500, Michelle Dupuis Henderson wrote:
Yes on booting from a USB drive/memory stick - we setup the same way.
Thanks for the tip.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Hello
I'm having a problem with the Linksys 3102: With incoming PSTN calls, I
can hear the caller through the X-Ten softphone, but he can't hear me. The
problem is worse with Sjphone and the GrandStream 100 hardphone, as I get
no sound in either direction.
FWIW...
- the SIP client, the
At 15:12 23/08/2006 -0700, Ron Wellsted wrote:
Here are mine (with UK regional settings/A-law).
Thanks a bunch :-) After more search, it turns out that if you don't need
the router feature of the 3102 (I already have a router), the unit must be
connected to the LAN through its... WAN plug.
(I'm sorry to ask this question here, but I didn't get a reply in
VoIP-related forums and I figured there's a lot of people here who are
knowledgeable about VoIP and SIP, and could help me see the light. Please
replace Axon PBX server with Asterisk in SIP mode if you will :-) )
I finally got
Hi
I'm a little lost on how to set things up with the two FXO cards I have: I
want card #2 to dial a number when a call comes in on card #1. Using the
following configuration, card #1 picks up the line and remains silent,
instead of dialing out through card #2. Anybody knows what's wrong?
Thanks Noah for the help, but... no go :-/
From: Noah Miller
ONE: You should answer an incoming zap line before doing anything with it,
so do this:
exten = s,1,Answer
exten = s,2,Dial(Zap/2/014XX)
When I try this, instead of using the Zap/2 interface to ring the other
number,
Thanks Noah for the help, but... no go :-/
From: Noah Miller
ONE: You should answer an incoming zap line before doing anything with it,
so do this:
exten = s,1,Answer
exten = s,2,Dial(Zap/2/014XX)
When I try this, instead of using the Zap/2 interface to ring the other
number,
Hello
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
You have a problem unrelated to what you are trying to do. Fix the
problem with dialing out of Zap/2 first.
As an Asterisk newbie, I have no idea what you mean :-) Besides checking
that the two FXO cards seem to be loaded by Asterisk
Hello
From: From: John D. Coleman [EMAIL PROTECTED]
Correct me if I'm wrong but I think you would want to use the transfer
command instead of dial to get it to call out to a remote office.
The reason I used Dial() is because it's what I saw in Asterisk The Future
of Telephony.pdf : (page 88)
Hi
From: Noah Miller [EMAIL PROTECTED]
Sorry for the long delay in responding. I didn't see you message until
now due to the postfix problems on the mailing list.
No problem. I've decided to dump the rPath PoundKey linux distro because it
was still using Asterisk 1.2.5 and it was pointless
Hi
It looks like the X101P clones I bought from eBay are dogs, so I'll look
into buying some FXO-SIP box instead. Hopefully, I won't have the same
problems with static, or caller ID and call termination not being detected.
Still, considering the number of people having similar problems with
At 22:36 09/07/2006 -0700, Michael Graves [EMAIL PROTECTED] wrote:
Skip local FXOs altogether. Setup an account with somone who provides DIDs
via IP. Call forward your analog line to the IP based number. It will be
absolutely painless compared to the
troubles of small FXO interfaces.
I'll
Hello
I'm several days into configuring this thing, and I still have a couple of
issues. FWIW, I upgraded it to 3.2.10, and will only use this device to
handle incoming calls from the PSTN, ie. no phone plugged into the Line
1/FXS port, and no need for outgoing calls through the PSTN
Hello
I'm following instructions on how to install Asterisk on Fedora 5, but I'm
having a problem:
- the host is an older i686 athlon i386 GNU/Linux
- /etc/rpm/platform says athlon-redhat-linux
- running yum update kernel downloaded kernel i686 2.6.18-1.2200.fc5
- running yum update
Hi
Most of our customers have generic names like Hospital, so I need to
rewrite their caller ID name by looking up the number in a database on the
Asterisk server, and rewriting the name such as Reading Hospital so that
we know who's calling.
Any idea if this can be done with Asterisk, and
At 22:07 22/11/2006 -0700, Marco Mouta wrote:
You can do it using AstDB, just load the database with callerid names and
numbers and then include a lookup on database in all incoming calls, so
you can override whatever you wanted:)
Thanks everyone. Indeed, it seems like using the embedded
At 22:07 22/11/2006 -0700, Marco Mouta wrote:
You can do it using AstDB, just load the database with callerid names and
numbers and then include a lookup on database in all incoming calls, so
you can override whatever you wanted:)
Thanks everyone. Indeed, it seems like using the embedded
At 12:00 25/11/2006 -0700, Tom Lynn [EMAIL PROTECTED] wrote:
By inverting the relationship, I found it easier to focus on the source of
the call and the treatments I want to apply. I can also wipe out entries
by family name and remove all attributes in one operation using database
deltree.
At 12:00 25/11/2006 -0700, Tom Lynn [EMAIL PROTECTED] wrote:
I like a challenge. I'll let you know if I come up with anything.
My eternal gratitude if you find something :-) And don't forget to update
the VoIP wiki so others can benefit too.
___
At 12:00 25/11/2006 -0700, Anselm Martin Hoffmeister
[EMAIL PROTECTED] wrote:
Do try
asterisk -rx database put cidname 12345676 \Me - cellular\
or
asterisk -rx 'database put cidname 3871263 Me - home'
These quotations seem to work.
Yup, I should have tried before posting. Thanks.
At 12:00 25/11/2006 -0700, Michelle Dupuis [EMAIL PROTECTED] wrote:
Try using smartCID (www.generationd.com). You'll get the benefit of
ranges of numbers mapping to single ID's (good for corporate blocks),
action field for blocking/accepting calls, etc).
Neat, although 411.com won't do as
Hello
I'm trying to add a VoIP trunk to Asterisk, but I'm getting the following
warning in the log file if I leave srvlookup=yes in sip.conf (OK if I
comment it out):
--
Nov 27 16:40:22 NOTICE[29660] chan_sip.c:-- Registration for
'[EMAIL PROTECTED]' timed out, trying again
Hi
I have the following setup to make outgoing calls:
X-Lite (build 34025) at home behind NAT - Internet - Asterisk at work
behind NAT - Internet - VoIP provider - GSM gateway - cellphone.
I just tried calling my own cellphone, but there is no sound either way.
Here's what I did on the
At 10:34 28/11/2006 -0700, Pavel Jezek [EMAIL PROTECTED] wrote:
I have done simple ael2 script, tak doing lookup in asterisk database
like: find full numer, if cidname isn't found, substract one digit from
right and try again, and so on
Thanks. If LookupCIDname doesn't come with its own
Hello
When I make calls from home to the PSTN by going through the Net -
Asterisk - the Net - VoIP provider - PSTN, I get no sound either way. I
assume it's because I must tell Asterisk to use fixed ranges of UDP ports
and map ports accordingly on the NAT firewall under which it is located on
At 17:01 01/12/2006 +0100, Noah Miller [EMAIL PROTECTED] wrote:
Just to double check - have you limited the RTP ports on the asterisk
server to 8000-8019 (in rtp.conf)?
Thanks. That what was missing. In rtp.conf, I fixed ports 1-10019 and
mapped those ports on the router, and it worked.
35 matches
Mail list logo
