On 06/12/05 16:16 trixter http://www.0xdecafbad.com said the following:
ports doesnt like it. How did you build it on 4.x?
and which 4.x specifically?
i've built and used the wcfxs and zaptel drivers on 4.11 from
/usr/ports/misc/zaptel. i've tried with the latest ports
(zaptel-freebsd-0.1
On 06/12/05 14:55 [EMAIL PROTECTED] said the following:
Basically, there is a fax line that I don't want to answer inbound, but I
want it available to do dial out from. Right now, we are using a busy wait
around the ringing line, but I was hoping for something that might be a
little more elegan
On 06/11/05 20:52 trixter http://www.0xdecafbad.com said the following:
linux distributions, freebsd 4.x it wont build, 5.x it will (havent
checked if it supports any of the FXO/FXS cards since that isnt a
it does build and run well on freebsd 4.x and 5.x, with drivers available
for the FXO/
On 05/05/05 18:43 Peter Svensson said the following:
My guess is that the Sangoma card does not switch to internal clocking
when the external clocking is lost, thus depriving Asterisk of the zaptel
if this is the case, the sangoma would need to fix this asap. it'd be hard
explaining to a custom
On 04/30/05 02:42 Matt Roth said the following:
Does anyone have an interest in forming a hardware architecture group?
absolutely !
It seems that Asterisk is so tightly linked to specialized hardware and
its corresponding architecture that developing the software alone is
insufficient for its ad
On 04/15/05 16:22 chawki hammoud said the following:
--- Andrew Kohlsmith <[EMAIL PROTECTED]>
wrote:
communications. ulaw is about 80kbps, and gsm about
28-30kbps.
I monitored the download and upload data rate during
my call using mandrake linux and it gave me 9.3 kb/s
using ulaw and 3.1 kb/s fo
On 04/04/05 22:51 Jesse D. Guardiani said the following:
:) I doubt it. The zaptel driver for FreeBSD isn't up-to-date with the
Linux version, so I doubt Zaptel support on FreeBSD will ever be quite
as reliable as Linux.
well, since you're sketchy on details, including exactly what/when the
hangs
On 04/01/05 00:00 Matthew Boehm said the following:
Steve Underwood wrote:
And your EU bias is clearly demonstrated by this. I've never seen a
BRI product outside he EU. :-)
Come to Houston, TX. We were running a BRI for quite some time before
upgrading to a T1.
ahem, ISDN BRIs are fairly commo
On 04/02/05 10:11 Mike Mueller said the following:
I don't think an Asterisk box can generate enough calls to cause sockets
related performance penalties. Five packets per phone call. What's the
max call rate an Asterisk box can support?
i think that would require an OS dependent answer.
but gen
On 03/23/05 04:15 Jesse Guardiani said the following:
This should be "has some issues". I do not consider
the FreeBSD zaptel support to be production quality
in any way. I experienced reproducible system hangs
(mostly after an asterisk restart), interrupt issues
(audio skips and SSH pauses during
On 21/02/2005 11:41 david said the following:
Hello,every one,
I have recorded the voice files with mandarin (China). Where should
I contrib the files ?
you could host it on a web server, and then modify the wiki page at
http://www.voip-info.org/wiki-Asterisk+sound+files+international to p
On 15/02/2005 21:05 Vledder, Hans said the following:
Hi Dan,
I've been investigating the same thing. Try to Google for Asterisk+Soekris,
Soekris is the company (http://www.soekris.com) that makes cute little 586
class fan-less single board computers that run both Linux and FreeBSD ...
i've got as
On 08/02/2005 19:23 [EMAIL PROTECTED] said the following:
and SIP clients. I was looking for chan_spy application but it seems to be
no longer available.
oddly, ChanSpy seems to be removed from mantis. any idea why this was done ?
--
Regards, /\_/\ "All dogs go to heaven
On 27/01/2005 03:38 Eric Wieling said the following:
Vahan Yerkanian wrote:
Was wondering if there are any news on the native MoH patch for
1.0.3/1.0.5.. or this still works on CVS HEAD only?
Since 1.0.x is for bug fixes only and not new features, I doubt that
patch will ever be in 1.0.x.
i'm u
On 19/01/2005 22:08 Jorge Mendoza said the following:
Thank You.
At least, we are not alone on this misadventure.
Jorge Mendoza
Vahan Yerkanian wrote:
That device is complete waste of time and money. I've been contacting
their support for the past 3 months and all I could get were promises
and la
On 10/01/2005 04:39 Leif Madsen said the following:
The Asterisk Documentation Project is proud to present, "The History
of the Zapata Telephony Project as it relates to the Asterisk PBX".
Written by Jim Dixon, the founding father of the Zapata telephony project
(http://www.zapatatelephony.org) whi
On 11/01/2005 04:21 Miguel Ruiz Velasco Sobrino said the following:
In a setup I've made i have a problem in the two way origination of the call.
Asterisk 1 <==> Public internet <==> NAT <==> Asterisk 2
I'm pretty sure it's a NAT loosing state too fast, and i can do nothing to fix
the NAT.
Is ther
On 08/01/2005 15:01 Dean Thompson said the following:
Just not sure whether it is SJPhone playing up and not sending the right
signal for the #, or whether there is a more underlying problem here.
in which case you could try using a dialplan with Read() and SayDigits() to
see of the SJPhone is ac
On 08/01/2005 11:35 Dean Thompson said the following:
Thanks for pointing that out. I have made the appropriate
configurations, but the system stil refuses to accept the pin for the
conference.
The definition now looks like the following:
meetme.conf file:
conf => 1234 OR
conf => 1234,5678
+-->
On 07/01/2005 03:22 César Davi Ávila do Nascimento said the following:
Hi all,
I'm trying to install a TDM400P card, and I need some help.
Please, see below...
**
*after dmesg command:*
did you run ztcfg to configure the zaptel devices /before/ starting
asterisk ?
--
Regards,
On 04/01/2005 05:59 Shoval Tomer said the following:
Grandstream, and many other two Ethernet phones have only 10Mhz network
interfaces, making it not so appropriate for a business environment that
needs 100Mhz.
a. take note of it before you buy :-(
b. does anyone know of a 100Mhz phone in the pric
On 30/12/2004 23:18 Jason p said the following:
As a community is there anything we can do to help with communications
to the Tsunami area ? we all sit on top of a welth of knowledge on
communications can we use this to help these area's in any way? IE
free sip calls , maybe there are * users in
On 26/12/2004 03:35 Eric Wieling aka ManxPower said the following:
I doubt Digium will provide support for using *BSD with Digium hardware.
Your best bet might be the asterisk-bsd mailing list or try using
i feel that digium should consider supporting BSD formally, given that the
number of peop
On 25/12/2004 16:48 Ronald Wiplinger said the following:
I want to record new sound files in different languages, but I need the
text files of the English ones, which I would use as basic.
Since some languages already exists, I believe such a list should be
exist, but where?
see http://www.voip-
On 25/12/2004 16:27 Ronald Wiplinger said the following:
I am looking for a small device with four FXO and one WAN connection.
Simple, so that the cleaning woman can make a hardware reset if
there're a number of FXO/SIP gateways you can consider.
--
Regards, /\_/\ "All
we're evaluating the use of a Lucent APX8100 E3/SS7 to SIP gateway for use
in conjunction with asterisk, serving something like 4000+ lines. does
anyone have experience with the APX8100 and it's integration with SIP on
asterisk ? does the APX8100 handle SS7<->SIP signalling well enough to be
us
On 19/12/2004 20:38 Rich Adamson said the following:
I'm 95% sure iax is not dependent on the ztdummy type timers.
trunked iax requires a timer, either ztdummy or a digium card.
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED](0 0)http://w
On 19/12/2004 16:40 Chris Miller said the following:
seems to be due to a timing issue, one that can't be solved under
FreeBSD with the zaprtp (linux) stuff, and I haven't seen anything as
the ztdummy pseudo timer works well under freebsd 4.x and 5.x. i used it
for a bit before i got my digium c
On 17/12/2004 22:21 Jon Bebeau said the following:
Let me jump in. Seems that the ChanSpy "patch" worked just fine in
pre-1.0.x. Provided MOH plus a bunch of there useful stuff. Now it
seems it's gone in 1.0.3 and scant little info on why or when (or if) it
will be back.
i've applied the Chan
On 15/12/2004 09:14 Kavit Munshi said the following:
Hi,
We are looking for a regular supplier of Digium hardware in Australia.
any help will be appreciated.
take a look at Australian Technology Partners in Melbourne. we've purchased
numerous TDM and TE410P cards from them and are quite pleased w
On 13/12/2004 21:08 Ali Ziaee said the following:
I'm very interested if somebody using asterisk on *FreeBSD* and not
Linux without problem ?
we've run it on freebsd 4.10 in pure voip mode as well as with TDM40B and
TDM22B digium cards. will soon be testing the TE410P with availability of
the be
On 09/12/2004 05:54 Steven Critchfield said the following:
Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
the nokia 9500 (communicator) as sold in asia uses symbian and has built-in
802.11b. i can see a software SIP phone here being us
On 04/12/2004 00:00 Andrew Kohlsmith said the following:
On December 3, 2004 10:33 am, Garry Taylor wrote:
Ouch, part reset, quickly restoring reality (0)
Power alarm on module 1, resetting!
I have looked though a lot of email on this issue, and no one seems to have
the answer.
How many people are
ahh, the famed steve critchfield has honoured me.
On 03/12/2004 13:21 Steven Critchfield said the following:
[EMAIL PROTECTED]:~$ ldd /bin/bash
libncurses.so.5 => /lib/libncurses.so.5 (0x40028000)
libdl.so.2 => /lib/libdl.so.2 (0x40067000)
libc.so.6 => /lib/libc.so.6 (0x4006
On 03/12/2004 04:01 Nick Bachmann said the following:
There's an excellent reason they're the first: those are both such
unbelieveably terrible ideas, especially the PHP init scripts.
I would reccomend IPCop, because their designers are a little more
would you elaborate why these are terrible
On 24/11/2004 23:21 Victor Alvarez said the following:
Hello,
I'm just wondering what is the situation today, 24 Nov 2004, regarding
asterisk timer for freebsd.
I would like to know if there is any way to run Meetme on Freebsd or if
there is anybody currently working on it.
zaptel and ztdummy d
BroadVoice is the most competitive companies I've seen on the net for
residential users, it's why I chose them for my own personal home service. I
mean, $19.95 a month, unlimited USA and 21 countries. $24.95 a month,
unlimited USA and 35 countries.
are they really /unlimited/ in the truest sense o
On 19/11/2004 22:44 Steve Underwood said the following:
as a gateway between what ? if it's SS7 on one side, what's on the
other ? SIGTRAN (SS7 over IP) on top of SCTP ?
Yep, that kind of gateway. He has his own SCTP, and doesn't use the
native Linux 2.6 one.
in which case, if * got itself a SIGT
On 19/11/2004 21:30 Steve Underwood said the following:
I can't imagine anyone successfully integrating openss7 into anything. I
believe it works OK on its own, and is in use as a gateway. It wasn't
as a gateway between what ? if it's SS7 on one side, what's on the other ?
SIGTRAN (SS7 over IP)
On 19/11/2004 21:13 alexandre::aldeia digital said the following:
I like to know why iLBC and GSM generate a 40-50kbps bandwidth
Is very high, if compared with your calculations for other codecs(G723.1
/ 17kbps and G729 / 24 Kbps).
the other codecs have better compression, but there's a higher
On 18/11/2004 17:25 kido noagbodji said the following:
I just purchased 10 G729 licenses for my asterisk box from Digium I was able
to register the key. But when i start asterisk it fails with the error
message:
[codec_g729a.so]Nov 18 09:27:01 WARNING[135073792]: loader.c:248
ast_load_resou
On 17/11/2004 16:13 Lex Lethol said the following:
Is there a list of loadzone/defaultzone country codes included in
zonedata.c? Can this list be updated to include my country? (I have
the tones)
edit zonedata.c, add in your tones and recompile/reinstall the zaptel
drivers. also, since libtonezone
On 18/11/2004 04:17 Steven Critchfield said the following:
On Wed, 2004-11-17 at 11:49 -0800, Tracy R Reed wrote:
And it seems to be something the developers are not interested in
supporting. Whenever someone asks about this feature they are normally
told that this is a feature of small-office "key
On 17/11/2004 04:33 Matt Riddell said the following:
Dinesh Nair wrote:
doesn't it pull it from the structures hardcoded into zonedata.c ?
iianm, indications.conf is only used for PlayTones().
Don't know, but I have non-standard tones defined which I analysed in
Wavelab, and the
On 16/11/2004 16:18 Matt Riddell said the following:
In zaptel.conf where you have the loadzone=xx and defaultzone=xx, the xx
says which indications.conf entry to use.
doesn't it pull it from the structures hardcoded into zonedata.c ? iianm,
indications.conf is only used for PlayTones().
--
Rega
On 16/11/2004 01:58 Eric Wieling said the following:
Dinesh Nair wrote:
i do not believe that digium sells the g729 codecs for freebsd.
however, i too am a freebsd user, and i guess what is needed is more
people telling digium that we need the g729 codec on freebsd.
They do. It's consi
On 15/11/2004 22:46 Martin List-Petersen said the following:
On Mon, 2004-11-15 at 07:46, Talbot Neil wrote:
Hi,
I was wondering if there is any PRI hardware that is Austel certified
and works
well with Asterisk under FreeBSD???
If anyone has any information please let me know as I seem to be
ha
On 16/11/2004 00:08 kido noagbodji said the following:
i have an easy way to install the codec under FreeBSD? It was tough enough
to install asterisk even with the FreeBSD ports.
i do not believe that digium sells the g729 codecs for freebsd. however, i
too am a freebsd user, and i guess what is n
On 14/11/2004 16:40 Olle E. Johansson said the following:
Dinesh Nair wrote:
would this patch help those who're not using broadvoice, i.e. does it
fix an issue with the way asterisk does not handle SIP registrations
correctly ?
No. The actual registration is still handled the same way.
On 11/11/2004 06:08 Steven Sokol said the following:
The patch is necessary because (I think I have this correct -- forgive
me if I scramble any of the details) the Asterisk SIP channel was not
caching the MD5 result of the original authentication dialog, and was
instead forcing the BroadVoice s
On 12/11/2004 16:08 Matteo Brancaleoni said the following:
I too demand sysmaster either pay Digium for a non-gpl license or
publicly admit the fact that they have repackaged Asterisk and
contribute enhancements to Asterisk back to the GPL.
*if they have made any enhancements* :)
actually, the t
On 07/11/2004 21:06 Richard Airlie said the following:
I had been running asterisk-0.9 with zaptel 0.7 with no problems (both
built from FreeBSD ports). Yesterday I cvsup'd my ports tree and build
asterisk 1.0.1_1 and zaptel 0.8_1, which seemed to work except that
any attempt to play music on hold
On 07/11/2004 12:35 Jeff Maki said the following:
2. Result is '200 result=0 endpos=0'
-- Playing 'digits/20' (language 'en')
-- Playing 'digits/3' (language 'en')
-- Playing 'digits/million' (language 'en')
-- Playing 'digits/4' (language 'en')
-- Playing 'digits/hundred' (language
On 17/10/2004 03:27 Vahan Yerkanian said the following:
On the site note, I was able to get a reply from Welltech, that accepts
this and 3 other bugs with 35xx and 38xx SIP versions, and saying that
all those require a new firmware, and that it'll take a lot of time for
oh, the holy grail: a res
On 06/10/2004 00:29 Tim Connolly said the following:
Is it fair to say the IAXy could be used to provide POTS lines for a
small, say 4 line PBX? The * would sit elsewhere providing the services for
the remote office (or home) pbx? Obviously, 4 units would be needed...
in this vein, does dig
On 06/10/2004 17:29 Vahan Yerkanian said the following:
1. setup a 35xxA FXS with all ports authenticating properly with *?
or
search this list's archives for a discussion on the wellgate 3504A SIP
proxies. i've fashioned a patch to chan_sip.c which allows the 3504A to
register all ports, but it
On 27/09/2004 00:50 Jay Milk said the following:
Eliminating the need to specify (and keep track of) priorities would
make changes to extensions.conf much easier to implement.
or perhaps allow non-consecutive priorities.
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL
On 26/09/2004 15:41 Vahan Yerkanian said the following:
I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought
several WellGate 3502A FXSes to play with till welltech guys fix the
3504a's registration bug.
welltech still hasn't responded to my complaint to them regarding their SIP
On 24/09/2004 00:27 Roger Schreiter said the following:
and where do I get a Zaptel-version matching
asterisk 1.0?
I only know CVS as source for the zaptel drivers.
this may have been asked before, but how does * release engineering work ?
1.0 has just been released, and would it be correct to say
On 23/09/2004 07:03 [EMAIL PROTECTED] said the following:
hi, do any of you guys using the port from freebsd have other problems?
the whole thing doesnt work for me, as in, after the first phone calls,
all calls dont have outgoing audio, also if i have a register line in
what version of freebs
On 23/09/2004 08:10 Nicolás Gudiño said the following:
This feature is already implemented and working in the next to be
released version of the flash panel (but now it will only dial numbers
predefined in the panel itself). You can get it from
http://www.asternic.org
when's the next release of FOP
On 22/09/2004 22:25 Michael Bielicki said the following:
depends where you are in india. If you are in Delhi, forget it. Get
whatever ISP as long as you can be sure his upstream is REACH and not
stupid telecom india and that your packets never get routed via
Singtel.
is singtel that bad ? we may be
On 21/09/2004 20:20 Jan Baggen said the following:
Compiled Asterisk from FreeBSD port (0.9.0_2)
When I start asterisk it uses 100% cpu. Searches on Google
say to comment the noload => chan_oss.so in modules.conf
But this is already commented. Make.conf contains some
optimizations.
add 'noload =
On 21/09/2004 16:13 Danny Zak said the following:
Hello Leo,
3802 is doing the same (it is the fxo one)
danny,
i've sent you the patch for 1.0-RC1 in a private email. could you apply
that, rebuild asterisk and the test it with the 3802 to see if the problem
goes away ?
--
Regards,
On 21/09/2004 15:07 Leo Ann Boon said the following:
I've just gotten the box to register all 4-ports with an external SIP
provider. The provider is running an old release of Broadsoft backend.
Seems like Broadsoft supports this strange way of authentication.
the 3504As do work with welltech's SI
On 18/09/2004 05:50 Chris Shaw said the following:
or audio quality. If it's cost savings, you could push 25 calls through a T1
using GSM encoding, but it would not sound quite the same as a regular line.
If you use G.711 (mu/A-Law) then you would get toll quality audio but only
be able to push abo
On 20/09/2004 18:15 Leo Ann Boon said the following:
a) The gateway will register all 4 ports if you're not using password.
b) If using password, the gateway will only register the 1st port
correctly.
yes, and this is why, as i posted a week back:
i think i've nailed it down to the fact that the 3
On 19/09/2004 16:12 Senad Jordanovic said the following:
Well.. since we are talking about this topic I owe you guys notes of my
experience
with SC1100 CPU used by various boards (www.soekris.com , www.pcengines.ch
etc.).
i'd be eagerly awaiting these results. i've tested a 16MB image of
asterisk/
On 15/09/2004 22:41 Benjamin on Asterisk Mailing Lists said the following:
On Wed, 15 Sep 2004 06:36:48 -0600, Raul Elizondo (wizardteam)
<[EMAIL PROTECTED]> wrote:
Acording to http://www.voip-info.org/wiki-Asterisk+iax+rsa+auth, and my
understanding of peer and user, [FWD-service] in the sample yo
On 10/09/2004 07:07 Leo Ann Boon said the following:
The 3502 does register with 2 accounts.
not the 3504A in Proxy Mode. i think i've nailed it down to the fact
that the 3504A (firmware 107a) uses the same SIP Call-ID but changes the
tag= parameter in the From header when it responds to the 407
On 09/09/2004 21:16 Evert Meulie said the following:
But... when I boot it, I get:
Registered to '192.168.11.6', who sees us as 10.138.3.2:4569
Why doesn't server A see server B as 192.168.2.44??
there's most likely a box between A and B which is doing NAT, and it's
most likely 10.138.3.2. however
On 09/09/2004 14:12 Olle E. Johansson said the following:
The "Unathorized" really suggests that the password is wrong for 1235.
Check that you have the same secret on both asterisk and the welltech.
that's the first thing i looked for, and i've more than triple checked
this. the passwords are ide
On 09/09/2004 05:17 Danny Zak said the following:
Hello Dinesh,
i'm pretty sure that asterisk doesn't support multi logins for each line!
erps, even though the logins are using different usernames/passwords ?
how would asterisk work with an ATA with multiple FXS interfaces then ?
--
Regards,
On 08/09/2004 20:29 Dinesh Nair said the following:
am trying to configure a WellGate 3504A FXS SIP ATA
(http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set
up two SIP clients in sip.conf as follows:
i forgot to mention that i'm running asterisk 0.90.0 on freebsd 4.1
hey * folk,
am trying to configure a WellGate 3504A FXS SIP ATA
(http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set
up two SIP clients in sip.conf as follows:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP chan
hey * folk,
need to tap the collective wisdom of this list for any details or
pointers to vendors who manufacture/sell SIP or IAX phones with builtin
magnetic stripe readers. these phones will be used in combination with *
in a prepaid application. it would be advantageous if the mag stripe
dat
hey asterisk folk,
i'm new to asterisk, but not to freebsd as i've been using it and
developing on it for quite a few years. we're planning on deploying
asterisk as part of a voip provider's service platform and naturally i'd
rather be using freebsd instead of a linux based distro. as such, i am
ve
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