Hi all,
I bought a modem with ambient md
3200 for test asterisk.
When i load the wcfxo module ...
ok it recognize my modem as:
Zapata Telephony Interface Registered on major
196PCI: Found IRQ 11 for device 00:0d.0PCI: Sharing IRQ 11 with
01:00.0wcfxo: DAA mode is 'FCC'Found a Wildcard
You could call Digium, lol.
- Original Message -
From:
Gleidson Antonio Henriques
To: Asterisk
Sent: Friday, October 29, 2004 2:16
PM
Subject: [Asterisk-Users] Ambient MD
3200+incoming problem
Hi all,
I bought a modem with ambient
md 3200 for test
Hello list,
I'd like you to know that version 0.3.5 of XC-AST is out - now it is all
translated into English and has a 20 page user manual, so I guess it's a
bit more user friendly. See http://demo.xcept.it/xc-ast
Plans for the future include a real time queue monitoring feature; I was
If you're not using chan_agent you have to do this your own somehow I'm
going to attempt to use dialplan logic along with 'tagged' CID but it seems
sometimes * doesn't pass the CalleridName after I modify itthis might
be a problem in my dialplan here.
The other option is to use rrmemory
Check your zaptel.conf and zapata.conf.
It worked for everyone and it should work for you too.
-Kannaiyan
- Original Message -
From: Steve Totaro
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, October 29, 2004 8:20 PM
Subject: Re: [Asterisk-Users] Ambient MD
--On Friday, October 29, 2004 11:08 -0700 Trevor Peirce
[EMAIL PROTECTED] wrote:
Michael Loftis wrote:
Little tireds now so you may have already done all this but make sure
you have latest libpri, zaptel, and asterisk, in that order.
It seems that disabling MMX support in zaptel fixed *all* my
All,
Has anyone got some example config files for the Polycom IP 500 SIP
phones?Specifically the sip.cfg, ipmid.cfg phonename.cfg and any
others that are needed to get the phones registered with *. I have a
few of the Cisco 7960 phones working but the documentation and examples
for the
On the client side, I'm not sure
what the risk is to say a SIP phone that has 5060 and some rtp ports
forwarded to it. Maybe someone can come in and list the threats to
both ends of a double NAT setup? I'm sure hundreds of us would be very
interested in this!
Here is a simple example. A user with
Hi everyne,
I need some help configuring SIP Friends on MySQL. I have already
compile and install Asterisk with the MYSQL_FRIENDS = 1 and created
the MySQL databases and tables, but I do not know how to call the
tables from my sip.conf.
Please, anybody could help me telling me what file I have
I am trying to get a Cisco PRI gateway to send calls to * and it doesn't
appear to be working. It is a 2610 running 12.3 IP+. I've got the
config in there, I can see calls come into the Cisco using debugs, but I
never see it try to connect to *. When I do debugs, I see the called #
as the
Hello everybody!
I've got a problem here. I writing an AGI in Perl and when I used the
stream_file method It did not work. Then I realized that the next line has
no waited for the streamed file end, the program has just gone on.
What should I do to make the routine wait for the stramed file
- Original Message -
From: Stewart Nelson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, October 29, 2004 3:51 PM
Subject: Re: [Asterisk-Users] Suggestion re: SIP/NAT/*
On the client side, I'm not sure
what the risk is to say a SIP phone that has 5060 and some rtp ports
I think you are missing a dial-peer voice xxx pots entry. E.g.:
dial-peer voice 200 pots
description Match all inbound POTS calls
incoming called-number T
direct-inward-dial
I don't think the PRI will pick up the call unless the called number
matches a number in one of the pots dial-peers.
I'm having some trouble with DISA() in a call plan that worked before 1.0. If
anyone has experience with it, I would appreciate some advice.
Thanks.
--
-M
There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
Hello,
Is anyone using the VoiceEclipse vePipe service with Asterisk?
I subscribed least week and have outbound dialing working great but am having
trouble with inbound calls. I see the call come in but get the following
message:
Oct 29 12:57:38 NOTICE[1087273664]:
ZXP, Niels Peen wrote:
I connected a Snom190 (fw3.56, sip) to * a few days ago and was
surprised to see that not only the transfer button but also the
programmable buttons on the side (with the leds) worked right out of the
box (allowing me to switch between multiple calls).
Snom 220 fw
Hello list,
I was looking for a way to implement non-blind call transfers with *, i.e.
the usual behaviour of most PBXs when pressing the flash button:
- A and B are talking
- A pushes flash
- A is free to compose a new number
- B hears music on hold
- A's call is answered by C
- A hangs up
- B
That was it. I knew it wasn't in there, but I was just trying to call
into the PRI to * and not from * out, so I didn't think it would matter.
Another goofy Cisco trick I guess.
Bruce Komito wrote:
I think you are missing a dial-peer voice xxx pots entry. E.g.:
dial-peer voice 200 pots
This should be a very nice option!, I have seen many requests now for the
supervised transfers. Are there some users that have a solution
(work-around) for this?
Does anyone know if and when this option comes available? Or can we write
a plug-in to inplement this future?
lenz wrote:
Hello
On October 29, 2004 04:17 pm, lenz wrote:
Hello list,
I was looking for a way to implement non-blind call transfers with *, i.e.
the usual behaviour of most PBXs when pressing the flash button:
- A and B are talking
- A pushes flash
- A is free to compose a new number
- B hears music on
Can you be more descriptive on what's happening? I use NuFone and
haven't had any issues, but I don't make that many calls.
On Fri, 29 Oct 2004 14:39:57 -0400, Paul Rodan [EMAIL PROTECTED] wrote:
We've been using NuFone for about 2 months, pushing an average of 10,000
minutes a month of
This is how I solved this for someone last week.
You have to add insecure=very and the only context you can land a call on is
the one listed in the general section. They send the callerid of the
calling party as the fromuser thus your system trys to auth the calling
party number and it fails.
I have * in several locations, one in a foreign country. To optimize on
the costs I am forwarding calls local for a remote server to the remote
server instead of dialing out locally. I want to keep it completely
transparent to the user however. Can the dial command rewrite numbers?
I forward
RE: SIP and NAT
Of course it is not nonsense running Asterisk and/or SIP behind a NAT.
In fact it can and does work well if set up properly by the various methods
available, many are technically sound and reliable and potentially secure.
I am running one Asterisk server behind a commercial,
I have some Sayson / Aastra 480e ADSI phones. They work great except for one
annoying feature. It's not really the phone, it's the ADSI programming
feature in Asterisk. I can't figure out how to recreate the factor default
flash screen key on the phone. When I create a ADSI script, it loads to
the
1-way audio problems. At least I think it's one way. We hear the remote
party breaking up. So it would be NuFone's ability to transmit, upload, or
our download bandwidth. We're not having bandwidth issues, we have 4 DS3's
at only about 70% of total capacity.
Last week I upgraded from a CVS Head
Can anyone tell me if the below is wrong for the phone configuration,
it keeps failed registration. (I had this working but lost all my tftp
config files so I know its a work configuration)
614p is my username password is my password and 10.20.30.3 is the asterisk box
Thanks in advance.
reg
Dialled locally Number that must be dialled remote
0034XX9X (-0034 and +9)
exten = _0034XX,1,Dial(CHAN/[EMAIL PROTECTED]/9${EXTEN:4})
ZXX 078ZXX (+078)
exten = _ZXX,1,Dial(CHAN/[EMAIL
Hi,
Remove the 614p@ from
reg.1.address=[EMAIL PROTECTED]
John Bittner
Simlab.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of
Matthew Marlowe
Sent: Friday, October 29, 2004 5:14 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
i have been using asterisk for a while now. and it has always worked
fine. till recently when i noticed this serious abnormally.
1. when i login remotely, it does not get up to the CLI prompt as seen
below.
[EMAIL PROTECTED] root]# asterisk -vr
Asterisk CVS-03/20/04-11:04:01,
Could be a case of routing from you to them and the various links
inbetween. Hard to really pinpoint given the numerous factors that
could cause such issues
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
nevermind. I was looking too far back. The answer is *0
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Yeah that's the most frustrating part with VOIP.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
Suffill
Sent: Friday, October 29, 2004 5:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Is NuFone
On Monday 18 October 2004 17:43, Your Own ISP .com wrote:
I can't believe how excited I am about a friggin piece of telecom hardware
but this is getting to be adictive. What a geek ;)
I am with you here, it's been a long time since I stayed up days on end
without sleep just to mess with
Thanks for your answer.
We don't have to use Nortel's BCM, it is one of the option we're considering
(not sure if it is still in the game now). I will ask this way, what
commerical fullvoip PBX you will recommend? Unfortunatelly I can't use
asterisk for this central point, but I can (and will)
Hi. We're looking for a reliable platform to run 12 or more T1s in a
single system. Very little or no transcoding. Mostly IVR and some
conferencing.
We have been been running 12 spans using Dual Athlon systems on an
older Tyan motherboard and 1500 MP CPUs. This works for about
9 or 10 spans
Jim Gottlieb wrote:
Hi. We're looking for a reliable platform to run 12 or more T1s in a
single system. Very little or no transcoding. Mostly IVR and some
conferencing.
We have been been running 12 spans using Dual Athlon systems on an
older Tyan motherboard and 1500 MP CPUs. This works for
Did you get anywhere with this issue, I seem to have the
same problem Rev H board FXO in port 4.
Regards
Ian
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Felix
PizarroSent: 24 September 2004 19:30To: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: Re:
I just read what I typed... I meant to say put the 614p in
the reg.1.address field with out the ip.
reg.1.address=614p
Sometimes I am dyslexic.
John B
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of
John Bittner
Sent: Friday, October 29, 2004
Michael Loftis wrote:
It seems that disabling MMX support in zaptel fixed *all* my problems,
from hold music, to iLBC times, to random crashes. That's odd though, as
I'm using a Celeron 1.7 GHz chip which supports MMX. Perhaps the small
celeron cache is to blame?
Could be, celeron's are also
--SNIP ALL--
IAX is no adequate replacement option for SIP either.
--SNIP ALL--
What?! How on earth could you come to that conclusion?!
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
On 03:47 PM 10/29/2004, Jim Gottlieb wrote:
We wanted to try the new AMD MP 2800 chips on the newer Tyan S2466
motherboard, but the systems hang or panic (with DMA errors) after
starting the zaptel drivers. We tried putting the older slower CPUs on
the new motherboard and had the same trouble.
I
Hi,
I'm looking at trying out an IP10S with Asterisk. I'll be recieving a
single unit next week to try out and see what she can do.
It seems to be comparable to a Snom190, but I don't seem to find much
detail online about it with Asterisk.
Is anyone out there using these phones? Any quirks,
We don't have to use Nortel's BCM, it is one of the option we're considering
(not sure if it is still in the game now). I will ask this way, what
commerical fullvoip PBX you will recommend? Unfortunatelly I can't use
asterisk for this central point, but I can (and will) use asterisk on
satellites
Greetings,
I've recently encountered some strange behavior placing outbound calls
using IAX via a VOIP provider. Intermittently, calls placed will ring
the called phone a couple times, then the ringing stops. However,
even after a few seconds of silence i can pick up the phone and the
Dear Steve andother * e1r2 developers and users;
Please allow me for relatively a long post.
Thanks to Steve's work, I downloaded libunicall, spandsp and libmfcr2 and it seems those libraries are successfully compiled.
Now I faced trouble compiling Asterisk. I have installed and operated many
On 2004-10-29 at 17:02, Chris A. Icide ([EMAIL PROTECTED]) wrote:
I currently have a development system I use when developing configurations
for my clients. It's a Tyan 2466 motherboard with the latest bios
revision, running with two AMD 3000 MP processors.
I forgot to mention that the
Michael George wrote:
I'm having some trouble with DISA() in a call plan that worked before 1.0. If
anyone has experience with it, I would appreciate some advice.
Perhaps you could post relavent sections of your dialplan...?
___
Asterisk-Users
Probably since there are so many SIP devices out there now and only a couple
IAX. In the future it is an awsome replacement.
- Original Message -
From: Matt Riddell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Friday, October
On Fri, 2004-10-29 at 21:53 -0400, Steve Totaro wrote:
Probably since there are so many SIP devices out there now and only a couple
IAX. In the future it is an awsome replacement.
So you would rather drive a '70s pinto instead of a Bugatti because
there are more 70's fire bomb pintos?
-
On Fri, 29 Oct 2004 16:59:56 -0400, Victor Cartes
[EMAIL PROTECTED] wrote:
Hello everybody!
I've got a problem here. I writing an AGI in Perl and when I used the
stream_file method It did not work. Then I realized that the next line has
no waited for the streamed file end, the program has
On Thu, 21 Oct 2004, Kevin P. Fleming wrote:
No, Asterisk cannot control an MGCP gateway at this time. If the AS5400 is in
MGCP mode, it will be expecting a softswitch to control it, and it will
Can you expand ... I'm hoping to use Asterisk to do SIP to MGCP mediation
... what does it not work?
On Fri, 29 Oct 2004 20:13:02 +0200, Wilson Pickett
[EMAIL PROTECTED] wrote:
I've used them for calls terminating in the US with good results. I
happened to put through a call to Romania today and it seemed the
person was hearing me very much lagged behind. The actual asterisk IAX
figure given
Hello everyone,
I want Asterisk works as a SIP client ofmy SIP
Proxy(running SER),
soI adda register definition in
sip.conf:
register=user:secret:[EMAIL PROTECTED]:port/extension
Now calls from SER can be handled well,But I can't call to SER
,
so I add a section in sip.conf
[EMAIL PROTECTED] wrote:
I've used them for calls terminating in the US with good results. I
happened to put through a call to Romania today and it seemed the
person was hearing me very much lagged behind. The actual asterisk IAX
figure given was like 80 ms which is usually pretty decent
--On Friday, October 29, 2004 17:02 -0700 Chris A. Icide
[EMAIL PROTECTED] wrote:
You may want to migrate away from the redhat kernel, and build your own
kernel whether you build a 2.4 or a 2.6 is up to you, but from what I
understand the 2.6 kernel is more efficient in SMP form.
I concur with
Hello,
I'm having a few problems with getting zapatta.conf to work properly:
threeway calling is enabled by my telco and works when not connected to
asterisk. The problem is that threeway calling doesn't work nor does call
forwarding. With threeway calling I am unable to send a hook/flash to
I think his point is that for a commercial rollout (say, a VSP), IAX is not
practical for all clients right now. It's not strange to have a personal
preference that is technically better but not commercially viable. That's
not an insult, just how things are sometimes. Maybe if there were some ~$70
Francois Menard (Mailing List Account) wrote:
Can you expand ... I'm hoping to use Asterisk to do SIP to MGCP
mediation ... what does it not work?
I don't know the particulars, because I've never used (or even looked at
MGCP). All I know is that whenever the issue comes up, people here say
that
On 06:05 PM 10/29/2004, Jim Gottlieb wrote:
I have had X100P, TDM4XX, and TE4 cards in it with no issue.
Have you had multiple cards in it at the same time?
Only in the X100P format, and only 2 of them
I never even tried the 2.4 kernels in the
system, I built the 2.6 kernel before installing
The only thing wrong with RedHat as far as asterisk is concerned is that
they do something goofy with their kernels and all you need do is recompile
a kernel from source. IMHO, you should always compile a kernel for your
specific hardware.
Does this mean that RHEL wouldn't really be a benefit
If you are going to be running a system at the central location as a
core switch, you will want to ensure you have, at minimum, an
enterprise-class PBX (Asterisk is an example of an enterprise-class
PBX). A carrier-class system would be better (but you're going to pay
for it!).
I have yet to hear
Has anyone had any experience with these folks?
http://www.citel.com/index/index.asp
That could be a compelling way to displace a legacy system with an
Asterisk.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hello,
Is anyone using the VoiceEclipse vePipe service
with Asterisk?
I subscribed least week and have outbound
dialing working great but am having trouble with inbound calls. I see the call
come in but get the following message:
Oct 29 12:57:38 NOTICE[1087273664]:
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