Hello... I'm having problems with H323/G729 setup. Below is the output
of h.323 debug when making a call. I use a SIP phone connected to an *
box in the same LAN. The * connects to a h323/g729 PSTN terminator
through internet. Calls rings and are answered in the other side, but I
get no sound at
Hello... I'm having problems with H323/G729 setup. Below is the output
of h.323 debug when making a call. I use a SIP phone connected to an *
box in the same LAN. The * connects to a h323/g729 PSTN terminator
through internet. Calls rings and are answered in the other side, but I
get no sound
Hi all.
I've installed a TDM card, with 1 FXO port. I've configured the zaptel
driver and everything seems to be ok: Asterisk answers the calls. Now,
the problem is that even when the caller hangs up (the caller is my
self from another PSTN line) Asterisk doesn't detect it, and it goes on
Hello.
This is my situation:
h323/g729 NAT (Asterisk as DMZ)
h323/g729
Asterisk ---
--- H323 Gateway --- PSTN
The problem is that everything works great, except
Hello.
I'm having a lot of trouble using asterisk behind NAT. This is the
situation:
h323
Asterisk ///--- H323 Switch
NAT Router
I see that my h323 traffic is going out to WAN with my internal IP, so
the H323
Hello again and thanks for the support. The problem seemed to be that I
needed to enable the g729 codec in the general section...
Thanks.
Aaron Johnson wrote:
Rich Adamson wrote:
Ok. Thanks a lot anyway. BTW, do you know how many g729 licenses I
need in this situation? Maybe 1 is not enough.
I tried type=friend and it is registering now... I'm happy with it this
time, but why can't I have the phone as user only (only to make calls)
and not as peer (to receive calls)??
Thanks,
RODOLFO
Rodolfo Grave wrote:
Hi again. I cant get my Budgetone registered in Asterisk, and I cant
find
Hi.
Has anyone accomplished to use the g729 codec? I have the license
installed, and I have tried with X-Pro and a Grandstream Budgetone
configured to use g729 only. This is what I get from Asterisk:
Dec 23 02:38:07 WARNING[21176]: chan_sip.c:2764 process_sdp: No
compatible codecs!
Dec 23
Yeap... :(
Kristian Kielhofner wrote:
Rodolfo Grave wrote:
Hi.
Has anyone accomplished to use the g729 codec? I have the license
installed, and I have tried with X-Pro and a Grandstream Budgetone
configured to use g729 only. This is what I get from Asterisk:
Dec 23 02:38:07 WARNING[21176
contact Grandstream support and they said they had
never have that kind of problem, and they asked me an Ethereal view of
network traffic.
Thanks a lot for your support.
RODOLFO
Kristian Kielhofner wrote:
Rodolfo Grave wrote:
Yeap... :(
Has (or are they willing) Digium support looked
:
Rodolfo Grave wrote:
I've seen other threads about the topic here (just that with a
Sipura)... this is the thread subject:
[Asterisk-Users] G729 and Sipura.
Digium's answer to this person was to blame the device, so I haven't
even try to contact Digium support. I'll do it now, and I'll let you
know
Hello.
I'm having this same problem with a Budgestream phone: I've correctly
installed G729 licensed codec (I've made all the checks you said in this
thread), but Asterisk keeps giving the fatal message:
*CLI Dec 21 18:02:49 WARNING[2375]: chan_sip.c:2764 process_sdp: No
compatible codecs!
Dec
Hello.
I'm having this problem with a Budgestream phone: I've correctly
installed G729 licensed codec in my asterisk box, but when I set my
budgestream to use only g729 codec, asterisk throws this message:
*CLI Dec 21 18:02:49 WARNING[2375]: chan_sip.c:2764 process_sdp: No
compatible codecs!
I'm crazy here trying to make X-Pro use ONLY g729, and you're struggling
to make it not to use it :)...
Can you please indicate what's your config for X-Pro and sip.conf?
This is mine:
X-Pro:
g729 is the only enabled codec.
sip.conf:
[12345]
type=user
username=12345
secret=12345
nat=no
Thanks.
The allow=g729a line was added in the desperation :) it doesnt work
without it neither. I tried the nat=yes but nothing...
If case this info is relevant: I have the same problem with a Budgetone.
When I force it to use g729 it the same happens.
More ideas?
Thanks again,
RODOLFO
Kanuri,
Hi again. I cant get my Budgetone registered in Asterisk, and I cant
find what's wrong... uff. This is my config:
This fragment is from my sip.conf:
[12345]
type=user
user=12345
username=12345
secret=12345
authuser=12345
qualify=1000
nat=no
host=dynamic
dtmfmode=rfc2833
reinvite=no
, Rodolfo Grave [EMAIL PROTECTED] wrote:
Thanks.
The allow=g729a line was added in the desperation :) it doesnt work
without it neither. I tried the nat=yes but nothing...
If case this info is relevant: I have the same problem with a Budgetone.
When I force it to use g729 it the same happens.
More ideas
Hi and thanks. Im looking into the ServerWorks model. I would like some
advice about using AMD since I've heard it is problematic... is there
any already tested model?
Thanks again,
RODOLFO
Steven Critchfield wrote:
On Mon, 2004-12-20 at 18:08 +1300, Richard Scobie wrote:
Steven Critchfield
Hi.
I gave up with the IBM NetFinity, so I'm going to buy new hardware. I'm
going to install:
1-)One X100P (1 FXO module)
2-)One TDM03B (3 FXO modules)
I'll have the 4 FXO channels busy almost all the time, and I would like
quality to be as good as possible without going to the high-level
Hello.
I have installed asterisk in a IBM NetFinity (single Pentium-II, SCSI
controller, SuSE9.0, one X100P card).
The thing is that when I run modprobe zaptel everything seems to be ok
(I've left the PC running after it long time and nothing happens). Then,
after I execute modprobe wcfxo (it
rtc
10: 30 XT-PIC aic7xxx, usb-ohci
11: 9365 XT-PIC ips, PCnet/FAST+ 79C972
15: 30 XT-PIC aic7xxx
NMI: 0
LOC: 0
ERR: 0
MIS: 0
Michael Vogel wrote:
Rodolfo Grave schrieb:
The thing is that when I run modprobe
Management version 2
The X100P card and the SCSI storage controller both have IRQ 15. Is this
what you thought about? What can I do to solve it?
Rodolfo Grave wrote:
These are my interrupts... I dont know enough to say if there is
something wrong there.
ghostserver:~ # cat /proc/interrupts
reboots.
Any other ideas? This card worked great on another PC, so a hardware
missfunctioning is not a probable choice.
RODOLFO
Eric Wieling aka ManxPower wrote:
Rodolfo Grave wrote:
The X100P card and the SCSI storage controller both have IRQ 15. Is
this what you thought about? What can I do
Hi. I've had the same problem when installing asterisk. Did you try a
make dep in the kernel source dir?
Att,
RODOLFO
Don Hughes wrote:
On 13 Dec 2004 at 16:38, Rick Green wrote:
Asterisk'd ones are different from yours. Since the sources were
retrieved successfully, I don't suspect a problem
Sorry... Im running SuSE 9.0
kido noagbodji wrote:
what os are you running?
K.
- Original Message -
From: Rodolfo Grave [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion [EMAIL PROTECTED]
Sent: Monday, December 13, 2004 1:27 AM
Subject: Re
asterisk looks for this file? Or if the
cause for this is another?
Im using the H323 channel included in the Asterisk tree.
Thanks,
RODOLFO
Corvin wrote:
Rafael J. Risco G.V. wrote:
On Sat, 11 Dec 2004 16:49:12 +, Corvin [EMAIL PROTECTED] wrote:
Dnia sobota, 11 grudnia 2004 15:32, Rodolfo
Hi.
I need to set up H323 on an Asterisk box. I've succesfuly compiled the
asterisk oh323 (including of course all the dependencies: PWlib and
OpenH323), and then compiled asterisk. However, asterisk doesn't report
a registered H323 channel (when it starts, it reports IAX2, ZAP and SIP
Hi all.
I've been reading through Wi-Ki and at the extensions.conf file
description (http://www.voip-info.org/wiki-Asterisk+config+extensions.conf)
The author says this:
One day, someone is going to write a proper scripting language for
Asterisk that can understand a simpler, easier (and more
Hi.
I'm using asterisk as a PSTN - SIP gateway, so that you can call to any
of the 4 PSTN lines connected to the asterisk box from and dial your
number, and asterisk will dial out through one of the 4 sip accounts I
have on a SIP - PSTN provider. I think of something like this in the
Hi. I'm getting new lines for using with Asterisk. In my Telco they said
I could choose between Analogic lines and RDSI lines... I've already
bought a TDM400P with FXO modules. Can you give some hints on the
differences between RDSI and normal Analogic lines? Would I have
problems for using a
is an
POTS/Analog NOT ISDN device
--On Tuesday, September 21, 2004 11:28 -0300 Marconi Rivello
[EMAIL PROTECTED] wrote:
On Tue, 21 Sep 2004 15:52:57 +0200, Rodolfo Grave
[EMAIL PROTECTED]
wrote:
Hi. I'm getting new lines for using with Asterisk. In my Telco they
said
I could choose between Analogic
Hi.
Can you give me some hints on how I can create a rotational board?
I dont even know how to spell it in english. What I want is to have more
than one line reserved, but with a single phone number, so that people
can call to the same number and get a ringing signal if any of the lines
is
repost this with a different title.
I hope you can understand what I mean in my bad english.
Thanks for your reply.
RODOLFO
Marconi Rivello wrote:
On Mon, 20 Sep 2004 23:22:29 +0200, Rodolfo Grave [EMAIL PROTECTED] wrote:
Hi.
Can you give me some hints on how I can create a rotational board
Hi and thanks a lot... information like that is what I was looking for.
Many thanks,
RODOLFO
Jorge Mendoza wrote:
Rodolfo,
What you are looking for is named hunting group. This is done at
Telco side, nothing to do at * side.
Jorge
Rodolfo Grave wrote:
Thanks, but no, that's not what I meant
this helps
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rodolfo Grave
Sent: Monday, September 20, 2004 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] How can I make a rotative board?
Hi.
Can you give me some hints
Hi.
I cant make SIP calls from asterisk.
When I start asterisk, I get the following message: What does it means??
Asterisk is not behind NAT or Firewall.
--
[chan_sip.so] = (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
Sep 16
this? Can't I
just say asterisk to use the eth0 IP??
Thanks a lot.
RODOLFO
Brian Wilkins wrote:
If it's what Andrew is talking about, then add the hostname to /etc/hosts.
On Thursday 16 September 2004 05:27 pm, Andrew Thompson wrote:
Rodolfo Grave wrote:
Hi.
I cant make SIP calls from
You can use WinAmp or xmms... it has a Plugin for playing GSM files.(not
included in the standard installation but you can find it in google)
RODOLFO
Sys.Concept wrote:
How to play GSM files?
I want to go through some of them but I'm not sure which player to use.
---
avast! Antivirus:
Hi... I think I've done that by typing dial 12345, the digits...
For example:
dial sip/[EMAIL PROTECTED]
and when the answer is attended, you type again:
dial 123456778... the digits you want to send... you can repeat that as
many times as you like... at least I think it has worked with me. I'm
Hi.
I have a x100p card installed on my asterisk box... my zapata.conf file
includes the following lines:
[channels]
context=default
switchtype=national
signalling=fxo_ls
rxwink=300
echocancel=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
Basically, the zapata.conf file generated
,
Shouldn't it be siganlling=fxs_ls for the x100p ?
Where is your channel = 1
What is in your zaptel.conf ?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rodolfo
Grave
Sent: 15 September 2004 22:52
To: Asterisk Users Mailing List - Non-Commercial
Wao!!! I did it and worked (honestly, It seemed to mystic to me)...
Thanks a lot, I would never had try that by my self.
Benjamin on Asterisk Mailing Lists wrote:
Any hints on why asterisk doens't get the call?
This happens from time to time on analog boards, especially on the
X100P.
Wieling wrote:
The Phone port wired to the Line port so you can still use a phone
plugged into the card when the server is down or powered off. Let me
repeat this: You cannot plug a phone into the X100P and expect it to
work with Asterisk.
On Sun, 2004-09-12 at 13:09, Rodolfo Grave wrote:
But I
/~eric/asterisk
On Sun, 2004-09-12 at 15:42, Rodolfo Grave wrote:
OK. Thanks a lot. I'll change my first set up then...:)
I want to be able to make a PSTN call to the line connected to asterisk,
and that asterisk answer that call and ask for a sip number to dial
is this also simple? can you give
Hi.
I have a x100p card installed and also asterisk, but I just dont get
asterisk to register with my sip provider (FWD)... when I start asterisk
using the following command I get the following messages (first, a lot
of messages show up immediatly after starting up: I'read this is normal,
then
Hi.
I have succesfuly installed asterisk and after I added a x100p card, but
the system doesn't seem to know the card is there. This is what I've done:
compiled and installed zaptel, libpri and asterisk in that order using
make clean ; make install commands. also, make samples for asterisk.
updated to recognise the card. If the
hardware does not see it the OS wont either.
I just built Asterisk on an old single 450 PII Compaq Prolient 800
with an x100p and 3 ata186's. It works just fine.
Hope this helps.
--john
- Original Message - From: Rodolfo Grave
[EMAIL PROTECTED
The mainboard is not recognizing the x100p card. It is not showing the
card on the PCI devices listing after POST
Any ideas, please?
RODOLFO
Rodolfo Grave wrote:
Hi and thanks.
Is there any especial issue about x100p? I'm building the system in an
old 400 Celeron but it detects
Hi.
I have installed a x100p (THE x100p for those who have seen my former
post). Now I just want to connect a normal phone (not an IP phone) to
the card and use it as a sip extension (I have a FWD account)... more
clearly:
I want to be able to pick up the phone and call any FWD user using my
voltage
FXS = expects to PROVIDE dialtone and ring voltage.
On Sun, 2004-09-12 at 12:59, Rodolfo Grave wrote:
Hi.
I have installed a x100p (THE x100p for those who have seen my former
post). Now I just want to connect a normal phone (not an IP phone) to
the card and use it as a sip extension (I
Hi,
I've seen that each country has its own PSTN qualities. I would like to
know the minimal characteristics needed in PSTN to use Asterisk and also
if some body knows which are Spain's PSTN's.
Thanks.
RODOLFO
---
avast! Antivirus: Outbound message clean.
Virus Database (VPS): 0436-4,
Hi again. My last post was incomplete, so I' reposting it.
I've seen that each country has its own PSTN qualities. I would like to
know the minimal characteristics needed in PSTN to use Asterisk and also
if some body knows which are Spain's PSTN's.
I'm interested in buying a TDM card (probably
Hi all.
I've being reading posts from the list since yesterday and I feel this
question was answered a lot time ago, but the list archives are a mess
(yet). I hope some one is willing to help me out.
I want to set up this:
caller - PSTN (SOMETHING1) -- VoIP - (SOMETHING2)
connection?
Is it possible to set a route for the IP packages? This is to optimize
the packets transmission over internet.
I'm outside US, so, why should I use tdm instead of x100p?
Thanks again,
RODOLFO
Rich Adamson wrote:
From: Rodolfo Grave [EMAIL PROTECTED]
I've being reading posts from the list
Thanks again Rich. I hope you dont mind to answer a few more
questions
My response is inline...
Hi! and thanks a million for your answer. You hace cleared many of the
doubts I had, including the differences between the cards. At the same
time, new questions has arised:
Is
Well, I wont get tired of saying thank you for the quick answers.
Rich Adamson wrote:
Is there a possible configuration in case I dont have a broadband
connection in the called-end, for example, a modem connection?
No, there is no modem support built
I was checking on the harware page of Digium and I found that there are
many TDM cards.
TDM10B - 1-port FXS bundle
Order Online
TDM40B - 4-port FXS bundle
Order
Online
TDM01B - 1-port FXO bundle
Order
Thanks.
Tony Mountifield wrote:
In article [EMAIL PROTECTED], Eric Wieling [EMAIL PROTECTED] wrote:
Rodolfo Grave wrote:
Can you explain further what a FXS and FXO port represents in a call
process in general?
FXO port - Expects to RECEIVE dialtone and ring voltage
FXS port
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