I was trying to install and make the calleridnamelookup.agi work with my
installation of asterisk.
I am not familiar with perl or agi scripts. I would like to know if anyone
has gotten the calleridnamelookup.agi to work with a similar installation:
I am running Fedora Core 3
2.6.9-1.681_FC3
try SIPPS from Ahead (Nero)
klaus
Adi Linden wrote:
I am looking for a German language softphone. Is there such a thing?
Adi
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On Sat, 2005-01-01 at 13:49 +1100, Howard Lowndes wrote:
When I try to start up zaptel, whilst running ztcfg, I get the following
error:
Jan 1 10:48:18 bu ztcfg: ZT_CHANCONFIG failed on channel 2: No such device
or address (6)
If it's anything like Mandrake and 2.6 kernels this is what
Damon Estep wrote:
Any PC platform is only as stable
as the sum of what you run on it, put a single analog interface in a red
hat ES on $10,000 worth of hardware and you will have to reboot every 3
days.
I'm not seeing these problem with X101P, nor does any of my (not so
many) clients. And all
Adi Linden wrote:
I am looking for a German language softphone. Is there such a thing?
DIAX has german language support.
rgds
pos
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Adi Linden wrote:
On Thu, 30 Dec 2004, Lyle Giese wrote:
Is your X100P set for loop start or Kewl Start? I am betting loop start,
try changing to ks instead.
Lyle
This is what I have in /etc/asterisk/zapata.conf so it should be Kewl
Start.
It might be that your local telco does not supply
Greg - Cirelle Enterprises wrote:
Are you running a stable (v 1.0 - 1.0.3) or cvs
Asterisk CVS-v1-0-10/03/04
I've upgraded two months ago to get a feature I wanted (SMS support). It
should be round about Asterisk 1.0.2
Gilad
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We try a X-Lite client from remote to connect to my *
I can call X-Lite and X-Lite can call me. However, X-Lite can hear my
voice, while I cannot hear him.
add these to sip.conf
disallow=all
allow=alaw
*CLI shows
*CLI (date) NOTICE[4637] rtp.c:317 process_rfc3389: RFC3389 support
incomplete.
In IAX protocol, both rtp and signaling are handled on the same port,
so the Asterisk is always in the path of rtp traffic.
Am I right?
If yes, is there anyway to set Asterisk just as signal proxy ?
IAX doesn't use RTP
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On Fri, 2004-12-31 at 21:14 -0500, Jerry Geis wrote:
All,
I have FC3 fedora core 3 and just installed and compiled 2.6.10.
after rebooting I attempted to recompile zaptel-1.0.3. I did a make clean
then make. I got the following errors.
Any suggestions?
---
is there any problem with wiki
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On Thu, Dec 30, 2004 at 11:16:01PM -0800, Alspach Family wrote:
I don't want to sound like a TV evangelist from the 80's and 90's but if
you have it to give, please do. We have operators standing by to accept
your donation. All you have to do is PayPal it to [EMAIL PROTECTED]
mailto:[EMAIL
is there any problem with wiki
probably too much champagne last night.
I only get a connection closed.
roy
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is there any problem with wiki
__
It seems to be down
thorben
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Must be the whole site is down
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adnan Ahmed
Sent: Friday, December 31, 2004 6:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] is wiki drunk
is there any
Any PC platform is only as stable
as the sum of what you run on it, put a single analog interface in a red
hat ES on $10,000 worth of hardware and you will have to reboot every 3
days.
I'm not seeing these problem with X101P, nor does any of my (not so
many) clients. And all that's
Working fine right now. (10:25 CST)
From: Jeff Glassman [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] is wiki drunk
Date: Sat, 1 Jan 2005 11:03:37 -0500
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Must be the
Happy new year!
The last 3 months my asterisk has run perferct.
But after I have set 15 new SNOM 190 phones on it dies every hour.
Nothing to se in CLI ore in the log.
It dies with exit status 139
Is there anyone who has an idea of what is wrong - ore any tip on how
to test.
I'm not seeing these problem with X101P, nor does any of my (not so
many) clients. And all that's boils down to is that: a. I'm lucky and b.
I've have helped my luck by using only one card per machine, choosing a
good MB and making sure the card don't share IRQ with anything.
But the
I'm not sure but I think it's not about DMTF, but about the silence supression
or VAD
(voice activity detection) that * doesn't support. Try unabling it in the
client.
We try a X-Lite client from remote to connect to my *
I can call X-Lite and X-Lite can call me. However, X-Lite can hear my
In article [EMAIL PROTECTED],
Eric Bishop [EMAIL PROTECTED] wrote:
Hi all,
Just got a brand new server and a Digium TE410P. I get the sequential
(knight rider) lights before loading the zaptel driver. As soon as I
load the driver all loghts go off. It appears the card is not
generating
I'm not seeing these problem with X101P, nor does any of my (not so
many) clients. And all that's boils down to is that: a. I'm lucky and b.
I've have helped my luck by using only one card per machine, choosing a
good MB and making sure the card don't share IRQ with anything.
But
Hi,
After a hiatus of several months, I decided to try spandsp again
because it is such an excellent addition to asterisk. Using 1.0.3 and
spandsp 0.0.1k.
When I send a fax from a laptop software fax that works with every
machine I've ever had to fax to, I get something like this as staus
return
Hi all,
I'm experiencing some problems with i4l and i can't find a solution. I'm
using
Eicon Diva 1 BRI
Eicon Diva Server 4 Bri
A ISDN PBX where I connect the first ISDN card (exten 204 in the
ISDN PBX) and a ISDN phone (exten 210 in the ISDN PBX)
Suse 8.1
Hi,
I had the same problem when i tried i4l, and as far as I remember the
solution was to set
the outgoing msn to the msn of the isdn-line.
From my old modem.conf:
incomingmsn=*
outgoingmsn=123456,123457
device = /dev/ttyI0
device = /dev/ttyI1
best regards,
Nils
On Sat, 1
Rich Adamson wrote:
The only issue I have with that is there are several people with digium
T1 and TDM cards in their systems, and its always the TDM that goes out
to lunch; not the T1. No doubt there are less then desirable mobos
around (and probably lots of them), but that doesn't explain why
I've seen something with the X101P that lead me to think so: I have two
cards and two lines. I also own a small UPS that happend to have a jack
for a phone line, to act as a power cleaner and I've put the line that
goes to one of these cards there.
Surge arrestors used for POTS lines
On Sat, 2005-01-01 at 21:05, Dave Cotton wrote:
On Sat, 2005-01-01 at 13:49 +1100, Howard Lowndes wrote:
When I try to start up zaptel, whilst running ztcfg, I get the following
error:
Jan 1 10:48:18 bu ztcfg: ZT_CHANCONFIG failed on channel 2: No such device
or address (6)
If
I just setup an Asterisk system on a small Shuttle box; I am only using
SIP channels and have no FXO/FXS cards. The system works fine in that I
can call my inbound number (Broadvoice) and have the system answer and
I can make outgoing calls. The problem is that every time I want to
change
Hello.
I am going to be putting together my first * system using FXO/FXS
interfaces. All the systems I have set up thus far have been pure VoIP
setups.
The system I need to set up should have 3 FXO interfaces and 1 FXS
interface, as well as several SIP phones. I have noticed people
complaining
On Sat, 2005-01-01 at 14:06 -0500, James wrote:
I've seen something with the X101P that lead me to think so: I have two
cards and two lines. I also own a small UPS that happend to have a jack
for a phone line, to act as a power cleaner and I've put the line that
goes to one of these
Scott Gruby wrote:
sip show peers
and it is blank with the system being hung.
snip
Any ideas on what is causing this? Is there any additional information I
can provide for assistance?
You can start with actually telling us what version of Asterisk you are
using, and how you installed it (from a
I am going to be putting together my first * system using FXO/FXS
interfaces. All the systems I have set up thus far have been pure VoIP
setups.
The system I need to set up should have 3 FXO interfaces and 1 FXS
interface, as well as several SIP phones. I have noticed people
complaining
What exactly are people seeing when they have issues with their TDM
card?
Brian Greul
Texas Shirt Company
www.txshirts.com
713-802-0369 / 713-861-6261 (fax)
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 01, 2005 3:09 PM
To: Asterisk Users
Current experience... three spa-3000's are far more stable
then a TDM card, and you'll get three fxo's plus three fxs's
for less money.
I have experienced nothing but grief when trying to set up the PSTN part
of the SPA-3000. Everything from crackly audio to fast busies.
Has anybody tried the
I personally have seen:
1) power alarms on FXO ports. It appears that rebooting the server is the
fix for this problem. I have not seen it enough to give definative answers
to questions about unloading/reloading the kernal modules to clear this
condition. But my experience is that once a port
[EMAIL PROTECTED] wrote:
What exactly are people seeing when they have issues with their TDM
card?
I have four of them in service, in everyday use--one RD, one home, and
two small office. None has given us the least problem, ever.
One caveat that might be germane, given the complaints of
[EMAIL PROTECTED] wrote:
What exactly are people seeing when they have issues with their TDM
card?
Brian Greul
Texas Shirt Company
www.txshirts.com
713-802-0369 / 713-861-6261 (fax)
Since you asked, and since I'm well into this bottle of Merlot on New
Year's day:
1. Power alarms. WTF does
On Sat, 2005-01-01 at 16:14 -0700, Michael Welter wrote:
Since you asked, and since I'm well into this bottle of Merlot on New
Year's day:
1. Power alarms. WTF does that mean? Wish I had some support docs.
2. On bootup, Excessive leakage module x, ProSLIC failed Auto
Configuration.
One rather common problem (which started the most recent thread on the
subject) is the card simply fails to process pstn-fxo calls. Most seem
to suggest it happens about once per week or two. When it fails,
reloading the drivers clears the problem (which requires taking
* down to do it). There
hi, i.m. newbie in asterisk. asterisk 1.0.3 is my current version.
i like to store usernames and passwords in a sql database.
i like to log failed authentification-passwords, to create a blacklist for
securityreasons.
i thingk a sql-database is a good way to log these actions.
i don.t find
[EMAIL PROTECTED] wrote:
On Sat, 2005-01-01 at 16:14 -0700, Michael Welter wrote:
Since you asked, and since I'm well into this bottle of Merlot on
New Year's day:
1. Power alarms. WTF does that mean? Wish I had some support docs.
2. On bootup, Excessive leakage module x, ProSLIC
Rich Adamson wrote:
Current experience... three spa-3000's are far more stable then a TDM
card, and you'll get three fxo's plus three fxs's for less money.
Except for the little problem I've fought for about a week without any
Joy - no combination of efforts from numerous sources (wiki, this
Steven Critchfield wrote:
Those first 3 all sound like you have a problem with power supply and
consistency. You don't mention what modules you have in the cards, but I
bet you have FXS ports and have too light of a power supply for the
job.
I'm not at the client sites, but my test system BIOS
Except for the little problem I've fought for about a week
without any Joy - no combination of efforts from numerous
sources (wiki, this forum members, my efforts) has succeeded
in a spa-3000/asterisk combination that actually works. If
you have specific spa-3000 and asterisk configs that
Hi
Have anybody successfully installed ISDN with HFC chips on [EMAIL PROTECTED]
ISO ?
Please tell me how you did it ?
Thank you !
HB
Norway
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Sorry about not including the additional info (I realized right after I
sent the message of my mistake):
Fedora Core 3 (from ISO images); no updates applied
Asterisk from CVS (as of this morning)
Zaptel from CVS
Libpri from CVS
No extra Patches
No Extra modules
Hardware is a Shuttle SS51G w/
I've been having audio breakup problems (on my end) in my Asterisk
tests. I'm not sure of the most likely source of this quality problem.
99% of my LD calls are calling into a tele-conference service called
freeconference.com for group meetings. Its a free phone conference
system that works quite
Nabeel Jafferali wrote:
Except for the little problem I've fought for about a week
without any Joy - no combination of efforts from numerous
sources (wiki, this forum members, my efforts) has succeeded
in a spa-3000/asterisk combination that actually works. If
you have specific spa-3000 and
Dec 31 12:03:55 NOTICE[7145]: chan_sip.c:7486 handle_request:
Failed to
authenticate user WIRELESS CALLER
sip:[EMAIL PROTECTED];tag=83eaec7dcb80f5feo1
Have you tried the A prefix trick, which uses Line 1 Call Forwarding
as opposed to PSTN Line Call Forwarding (with the added advantage that
I have experienced nothing but grief when trying to set up
the PSTN part of the SPA-3000. Everything from crackly audio to fast
busies.
BTW I take that back. I spent an hour on this after posting my last
email, and with a little tweaking, everything seems to be working well
now.
--
Nabeel
Nabeel Jafferali wrote:
Dec 31 12:03:55 NOTICE[7145]: chan_sip.c:7486 handle_request:
Failed to
authenticate user WIRELESS CALLER
sip:[EMAIL PROTECTED];tag=83eaec7dcb80f5feo1
Have you tried the A prefix trick, which uses Line 1 Call Forwarding
as opposed to PSTN Line Call Forwarding (with the
Rich,
I have been wondering if the spa 3000 would make a good PSTN interface
for an * box where POTS is the only available (or practical) service.
Have you implemented this? Are there any limitations or known issues?
The SPA2000 sure seems to work well as an ATA, even had good luck with
fax over
I hate to ask the obvious
But what's your power quality like? Is the system on a UPS?
UPS supplied power makes a huge difference in system stability. I
wouldn't run a server for anything (including testing) without it.
Second, what class of hardware? You do get what you pay for and
On Sat, 1 Jan 2005 15:52:50 -0500, Nabeel Jafferali wrote:
Hello.
I am going to be putting together my first * system using FXO/FXS
interfaces. All the systems I have set up thus far have been pure VoIP
setups.
The system I need to set up should have 3 FXO interfaces and 1 FXS
interface, as
Hi All,
Everytime I make outgoing call, the channel at TDM
card doing hungup after might be a second the
destination number get ringing.The call is from sip
phones to PSTN phone. The sip phones was completely
registered to asterisk. here is my conf :
sip.conf :
[1234]
type=friend
username=1234
Hello--
What I'd like to do:
Use IAX softphones running on computers, in Auto-answer mode, with sound
going to speakers, as a sort of public announcement system.
What isn't working:
Well, my first experiment was to set up the MeetMe system described on
the Wiki...
This works fine for voice
On Sat, 2005-01-01 at 17:25 -0700, Michael Welter wrote:
Steven Critchfield wrote:
Those first 3 all sound like you have a problem with power supply and
consistency. You don't mention what modules you have in the cards, but I
bet you have FXS ports and have too light of a power supply
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