Does anyone know a good (and stable) voip gateway product with 4 ports
(2 fxo and 2 fxs), with the following requirements:
* being able to connect analog phones to the FXS ports, and communicate
over SIP with an REGISTRAR/PROXY server (SER in our case).
* being able to connect the FXO port
Hi Greg!
Sorry, but I can't help you. I've never tried asterisk with the linejack.
regards,
klaus
greg wrote:
I found a message from you to the asterisk users mailing list from 2001. I was
wondering if you got (or still have) an asterisk system working with the
linejack? If so, would you be
Stephen J. Wilcox wrote:
Okay having taken in some suggestions and googled this topic to death I'm still
stuck - anyone got any ideas?
To recap, the faxes are coming in via a digium E1 card but failing to train
properly or if they manage it sending a garbled and very truncated fax.
A number of
Hi,
I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card.
Incoming calls and outgoing calls between my cisco and my SIP phone works
fine on G.729. Recording messages in the asterisk voice-mailbox also works
fine from both my SIP phone as well as PSTN - Cisco - Asterisk. I
Anyone else got a Sipura 3000 in the UK? Apart from CID not working it also
seems to not notice any of the line state changes on the PSTN when the
remote party terminates the call. It only recognises the offhook signal
which gets sent much later.
Chris
Hi,
Well, the Problem is not the ZAP Channel but the SIP Channel, because it
occurs no matter what channel I use the phone outside. Maybe this graph is
more descriptive:
1. ZAP or SIP == 2. Asterisk == 3. SIP (thru internet, provider sipgate)
== 4. PSTN
The connections on 1. hang up correctly,
On Fri, 30 Jul 2004 [EMAIL PROTECTED] wrote:
###zapata.conf
context=default
context=alex
pridialplan=unknown
echocancel=yes
echocancel=yes
echocancelwhenbridged=yes
immediate=yes
Why do you define echocancel and context twice?
a) when i try to make an inbound call to msn I get
Glare cannot be prevented on two way trunks (it is physically impossible
because the two ends are separated in distance and therefore separated in
time and any independent decision to use it at one end is never seen
instantly at the other end).
Ground start does not decrease glare at all (it
Chris Stenton [EMAIL PROTECTED] wrote:
Anyone else got a Sipura 3000 in the UK? Apart from CID not working it
also seems to not notice any of the line state changes on the PSTN when
the remote party terminates the call. It only recognises the offhook
signal which gets sent much later.
The
I have been seeing reccomendations for using asterisk as a soft-pbx with
the reccomendation being to use regular analog phones via FXS rather
than SIP.
Is this still a big issue? Or is this a left-over from previous bad
experiences? I have been doing demos with SIP phones, and some IAXYs
Andrew Kohlsmith wrote:
On Friday 30 July 2004 19:51, Mike Benoit wrote:
Tuning these [PCI latencies] should allow you to give your TDM cards
long burst lengths, and make your IDE devices very premptable...
I would have figured you want very short burst lengths to prevent any one
device
Hello,
Eric Bart wrote:
Flash don't work for sip
This affirmation is too broad, it might not work with X-lite, but flash
will work with may sip devices, including cheap ones (grandstreams,
sipuras, etc).
From: Jozeph Brasil [EMAIL PROTECTED]
I have one X100P installed with two SIP extensions
Andrew Kohlsmith wrote:
On Friday 30 July 2004 19:51, Mike Benoit wrote:
Tuning these [PCI latencies] should allow you to give your TDM cards
long burst lengths, and make your IDE devices very premptable...
I would have figured you want very short burst lengths to prevent any one
device
Hi,
I've been reading some manuals and have added a bunch of SIP Accounts for
outbound calls into my Asterisk Setup.
The local extensions are working perfectly.
The problem I am facing at the moment is that, if I try and make outbound calls
using a SIP Account, it rings thrice and then there is
I have the same issue with IAX2. I get messages anywhere from 5 min to 45
min of silence.
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 30, 2004 9:59 PM
Subject: [Asterisk-Users] VoiceMail Not releasing
About twice a week we have a caller that
Hmmm,
Flash work for IAX?
-Mensagem original-
De: Eric Bart [mailto:[EMAIL PROTECTED]
Enviada em: sábado, 31 de julho de 2004 02:26
Para: [EMAIL PROTECTED]
Assunto: Re: [Asterisk-Users] Softphone - Freeware?!
Flash don't work for sip
- Original Message -
From: Jozeph Brasil
Is it possible to get a few 7960's and asterisk to allow all
of the 7960 phones to use one extentsion and can only be used
by one person at a time, have it indicate on the other 7960's
when one of the others has the line engaged. Basicly so like
I can setup a rule when an incoming call comes from
Sean McKay wrote (on Jul 31):
The goal here is
to allow me to pick up a call on any of the 7960's anywhere
in my house and be able to move from room to room as needed
by placing the call on hold and picking it up on one of the
other phones in the
On Saturday 31 July 2004 00:20, Steven Critchfield wrote:
Actually, it isn't VoIP data yet, VoIP is Voice over Internet Protocol.
The 1000hz interupt is still just digitizing the audio off the PSTN
link. When it comes time to read/write VoIP data, it is likely 20ms of
audio, plus headers and
Thanks for the correction
I didn't know that SIP would do. As I understood
the R key will send the flash signal.
However does it really act as a transfer ?
For the zap transfer, as said in :
http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20zap%20transfer
when the transferer hangs
On Friday 30 July 2004 20:26, Aidan Van Dyk wrote:
Yes, but by lowering the available time that another device can tie up the
PCI bus, you increase the chances that your TDM cards get their interrupts
handled very quickly. And since you know that your TDM card interupts are
very short (but
I'm guessing then CCM could handle this, and I could simply
link asterisk to the CCM as a gateway and acchieve the reult
I'm looking for
On Sat, 31 Jul 2004, Chris Luke wrote:
Sean McKay wrote (on Jul 31):
The goal here is
to allow me to pick up
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Saturday 31 July 2004 09:41 am, Chris Luke wrote:
Sean McKay wrote (on Jul 31):
The goal here is
to allow me to pick up a call on any of the 7960's anywhere
in my house and be able to move from room
I'm having problems compiling cdr_addon_mysql with MySQL 3.23.58
I get the following errors:
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o
cdr_addon_mysql.o cdr_addon_mysql.c
cdr_addon_mysql.c:50: warning: parameter names (without types) in
function declaration
On Sun, 1 Aug 2004 [EMAIL PROTECTED] wrote:
exten = _34.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
The r at the end of this line tells asterisk to generate a ringing sound
for you to hear. In other words, the ringing you're hearing isn't coming
from the far end SIP device. Taking the r out will
Hi Eric,
Eric Bart wrote:
Thanks for the correction
I didn't know that SIP would do. As I understood
the R key will send the flash signal.
However does it really act as a transfer ?
For the zap transfer, as said in :
http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20zap%20transfer
Hi-
I've used MySQL, both version 3 and 4 with no trouble. I copied the
following from my notes:
* MySQL version 4 - use the files in /usr/src. Move to new directory
/usr/src/mysql. Install by using rpm -U for each, in this order:
* shared-compat
* client
* devel
Hi
I plan to setup an asterisk box to function as a SIP gateway forwarding
lots of calls to/from a backend of several other asterisk boxes, each
with a TE410 card for PSTN connectivity. It will only gateway the
calls into the PSTN gateways. No transcoding is planned - only plain
ALAW. How
Hello everybody,
my situation is the following: I have an ISDN telephone connected to a
HFC ISDN card on an asterisk server. The asterisk server is behind a
NAT, but all the ports (i.e. 5060 and the range specified in rtp.conf)
are forwarded to the asterisk machine. I am using the German SIP
Fantastic - Many thanks!
For the purposes of the archive, this is what I did..
Edited /usr/src/asterisk/Makefile
Just after:-
ifeq (${OSARCH},Darwin)LIBS+=-lresolvendififeq (${OSARCH},FreeBSD)LIBS+=-lcryptoendifLIBS+=-lssl
I added:-
ifeq (${OSARCH},OpenBSD)LIBS=-lcrypto -lpthreadendif
And it
Yes, NAT is a problem.
Due to the changes on the Broadvoice side, my router will not work anymore.
I will change the router for linksys which works for other people.
This was also recommended by James Jones from broadvoice tech support.
One more time, thanks for your support James.
Bart,
I think you have to use parking.
- Original Message -
From: Sean McKay [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, July 31, 2004 9:36 AM
Subject: [Asterisk-Users] one extention, multiple phones
Is it possible to get a few 7960's and asterisk to allow all
of the 7960
Thanks
I don't understand why sipura can do consultative transfer
and why grandstream can't. They're both SIP, aren't they ?
If you want consultative transfers with asterisk, you can sort of have
it by using parking: you can dial '#' to transfer, then send the call to
the parked calls
Hi,
I've had a look at it and the timeout error is what happens straight after the
phone disconnects:
Aug 1 04:07:13 WARNING[106511]: pbx.c:922 pbx_substitute_variables_temp: The
use of 'EXTEN-foo' has been deprecated in favor of 'EXTEN:foo'
Aug 1 04:07:20 WARNING[5126]: chan_sip.c:673
In article [EMAIL PROTECTED],
Mike Benoit [EMAIL PROTECTED] wrote:
I also discovered my SPA-2000's silence suppression was causing a good
chunk of choppiness (much more so then any SS should), so I disabled
that too.
Asterisk requires that SIP devices have silence suppression disabled.
It uses
In article [EMAIL PROTECTED],
Malcolm Bader [EMAIL PROTECTED] wrote:
I'm having problems compiling cdr_addon_mysql with MySQL 3.23.58
I get the following errors:
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o
cdr_addon_mysql.o cdr_addon_mysql.c
cdr_addon_mysql.c:50:
[EMAIL PROTECTED] (Rich Adamson) writes:
Like *, it also has an internal dialplan, however understanding the
various interactions requires some experimentation, as each of the
interfaces seem to be considered a gateway, and part of the dialplan
directs calls to gw0, gw1, gw2 (etc) which
Hi !
I am connecting to Adit 600 thru a T100P card
I have configured 1-16 FXS channels and 17-24 FXO.
Everything looks fine on Asterisk side I get a tone
on all FXS channels, but when I try to dialout thru
one of the FXO channels 17-24 it doesn't connect to
the POTS line and echoes back my voice.
The box was truly targeted for the residential user where existing
phones interface on one side, the pstn line on the other side, and
the default call is sent to the voip interface. Disconnected (or
failed) ethernet results in a relay flipping, tying the fxs directly
to the fxo. Same
First of all I use this in humor:
Frank you ignorant slut!
I have to disagree on your analysis. I worked in telephone COs (DMS250,
Stromberg/Carlson) and with PBXes for over a decade. Glare can and is
controlled by ground start signaling. It does so because the ground is
tested for (or
Hello,
is there a way I can obtain the IP endpoint address when the telephone is
called from app_queue?
I even tried creating a pseudo number, so that instead of having my queue
call straight (say) OH323/1234 I call a number on asterisk where I log the
call id and then do the dialling. Of
Wolfgang S. Rupprecht [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] (Rich Adamson) writes:
Like *, it also has an internal dialplan, however understanding the
various interactions requires some experimentation, as each of the
interfaces seem to be considered a gateway, and part of the dialplan
Does anybody has the expirience configuring Asterisk with Cisco ATA 186
MGCP firmware ?
I have Cisco software v3.1.1 atamgcp (Build 040629A)
Asterisk 1.0-RC1
On ATA i only put domain test.
mgcp.conf looks like this
[test]
host = 192.168.195.55
context = default
line = aaln/2
line =
Hi Roy-
I've done a lot of load testing with asterisk and TE410P's.
My guess, with no transcoding, is that you might be able to handle 8 E1's
max on the PSTN side absolute max (ie: 2 TE410P's). This assumes you have a
fast processor.If you're using T1's, scale these numbers up accordingly,
Roy Sigurd Karlsbakk wrote:
Hi
I plan to setup an asterisk box to function as a SIP gateway
forwarding lots of calls to/from a backend of several other asterisk
boxes, each with a TE410 card for PSTN connectivity. It will only
gateway the calls into the PSTN gateways. No transcoding is planned
I am running * CVS head on Gentoo/i586 and Gentoo/Sparc64 (US60
2x450/1GB RAM),
they are running great.
On sparc64 * does not compile out-of-the-box, some hackings in the
Makefiles are needed,
Ming-Wei
___
Asterisk-Users mailing list
[EMAIL
On Sun, 1 Aug 2004 [EMAIL PROTECTED] wrote:
Hi,
I've had a look at it and the timeout error is what happens straight after the
phone disconnects:
Aug 1 04:07:13 WARNING[106511]: pbx.c:922 pbx_substitute_variables_temp: The
use of 'EXTEN-foo' has been deprecated in favor of 'EXTEN:foo'
Aug
I've got it to work in the past. I've
upgraded to SIP, seems to work better.
Is there a reason you MUST have MGCP?
Duane
- Original Message -
From:
Dmitri Baranov
To: [EMAIL PROTECTED]
Sent: Saturday, July 31, 2004 12:38
PM
Subject: [Asterisk-Users] MGCP
Besides playing with Asterisk, i'm also using Linux for all kinds of
multimedia things, especially recording music, mixing, etc.
In order to use Linux as a digital audio workstation, there are a few
things that one must do: use low-latency kernels, use pre-emption, use
apps that run with real-time
Hello,
I read the previous postings in asterisk-users mailinglists and I didnt
found any postings related to my problematic topic.
Problem:
If I need to set different pridialplan for different channels. For example:
group1 has first 15 channels and all calls what are sendt via this group
are
Eric Bart wrote:
Thanks
I don't understand why sipura can do consultative transfer
and why grandstream can't. They're both SIP, aren't they ?
They use different sip stacks... and yes, they are both sip. Try to mix
in the same environment sipuras, grandstreams, snoms, uniden, saysons,
cisco,
I am trying to get PrepaidCID working and,
it shows it connecting to the database correctly. I call the
extension and it Asterisk does a core dump.
Can anyone help me?
Norm
You can easily ring different phones at the same time within the dial
command. For example,
SIP/4024${PRITRUNK1}/16505551212${PRITRUNK1}/1411212
A blind transfer will move the call to the next phone. Or you can park the
call.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third
I don't understand why sipura can do consultative transfer
and why grandstream can't. They're both SIP, aren't they ?
They use different sip stacks... and yes, they are both sip.
Maybe the sipura transfer is using a sip reinvite or some
other SIP command.
Does the consultative transfer
Brian Elton wrote:
I think I am the first to use the $135 Avaya 4602 SIP phone, but I
need some support from the community to fix one problem I have with
it.
The phone stops working after about 20-30mins if I have
mailbox=context in Asterisk; when I do have mailbox=contect in
asterisk the sip
I am ready to close that topic.
Finally, I replaced my router from Multitech for Linksys.
It solved all the problems related to NAT and incoming calls issues.
My router model is Linksys BEFSX41.
Thanks for your help, asterisk people.
Bart,
Yes, NAT is a problem.
Due to the changes on the
On Sat, 2004-07-31 at 12:27, Florin Andrei wrote:
- if Asterisk doesn't already do that (correct me if i'm wrong), does it
make sense to make it run with real-time privileges, just like JACK? (i
have no idea how JACK accomplishes that, to me it's just a command-line
option that makes it a lot
Wolfgang S. Rupprecht [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] (Rich Adamson) writes:
Like *, it also has an internal dialplan, however understanding the
various interactions requires some experimentation, as each of the
interfaces seem to be considered a gateway, and part of the
Steve Totaro wrote:
[I think you'll find that inline-posting makes treads easier to read]
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 30, 2004 9:59 PM
Subject: [Asterisk-Users] VoiceMail Not releasing
About twice a week we have a caller that
My ATA with V3.0 firmware works fine.
Check that test can be resolved by your DNS or is in /etc/hosts. You
might just want to put the IP address directly.
Dmitri Baranov wrote:
Does anybody has the expirience configuring Asterisk with Cisco ATA 186
MGCP firmware ?
I have Cisco software v3.1.1
Sorry to just jump in on the list like this, but I'm in a hurry. I've
become lost in the mumbo-jumbo of the Wiki. Is there anyone on this
list who would set up a Asterisk system for me?
I have a Fedora Core 2 fresh install on an HP Pavilion 6553, along with
a single POTS line. I'll buy the
On Sat, 31 Jul 2004, Paul Mahler wrote:
You can easily ring different phones at the same time within the dial
command. For example,
SIP/4024${PRITRUNK1}/16505551212${PRITRUNK1}/1411212
A blind transfer will move the call to the next phone. Or you can park the
call.
That's not what I
Rich Adamson [EMAIL PROTECTED] wrote:
Kevin Walsh wrote:
You can apparently use the SPA-3000 dialplan to specify that the
call should go via its FXO port, without going via Asterisk. This
could be useful for emergency services. I don't have a SPA-3000 yet,
so I can't say what happens if
Florin Andrei [EMAIL PROTECTED] wrote:
On Sat, 2004-07-31 at 12:27, Florin Andrei wrote:
- if Asterisk doesn't already do that (correct me if i'm wrong), does it
make sense to make it run with real-time privileges, just like JACK? (i
have no idea how JACK accomplishes that, to me it's just
For those that are interested, here is my report back to Sayson on the
480i
-- Forwarded message --
Date: Sat, 31 Jul 2004 22:03:31 -0400 (EDT)
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: 480i User Feedback With Asterisk
Seshu,
I am using a 480i, and I am impressed
On Sat, Jul 31, 2004 at 04:36:51PM -0400, Norman Tomlnis wrote:
I am trying to get PrepaidCID working and, it show's it connecting to the
database correctly. I call the extension and it Asterisk does a core dump.
Can anyone help me?
If you'd like to read over
[EMAIL PROTECTED] wrote:
For those that are interested, here is my report back to Sayson on the
480i
Thanks for the report, some of us are very interested!
Look forward to hearing back from you. Nice work on this phone. If it hits
the market with a complete firmware, at $200 or less, they will
Does anyone know if the 480i supports 802.1Q?
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 31, 2004 10:17 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 480i User Feedback With Asterisk (fwd)
[EMAIL PROTECTED] wrote:
For those that
On Sat, 2004-07-31 at 20:11, Sean McKay wrote:
Say I want someone to join in on the conversation, I'd rather much have
them be able to just lift the receiver and begin to talk rather than have
to do conferencing.
This is done on PSTN (normal home phone), and I've seen it done on PBX's
such
On the Merlin Legend I believe the function you're talking about is on the
MLX-20 receptionis's console. When the system is in hybrid-PBX mode, you can
simply press a line button that's in-use and you can listen in on the
conversation and even talk, it basically puts you in a conference with the
I found that it was also necessary for me to add
-lm to the LIBS line for it to work on my nice fresh OpenBSD 3.5 installation.
--
Greg Broiles, JD, EA
[EMAIL PROTECTED] (Lists only. Not for confidential communications.)
Law Office of Gregory A. Broiles
San Jose, CA
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