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launch
new services until we can get Asterisk installed and running in our
location.
Please let me know if you can provide this service or if you know of a
company that can do it for us.
thanks!
Sam Michelson
-Original Message-
From: Lubomir Christov [EMAIL PROTECTED]
To: [EMAIL
Any suggestions on how to go about
this?
so person calls, recording: "press2 to call
cell phone", user presses 2, call forwards to my cell phone.
Thank you
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Hi,
If I setup an IP PAX gateway to handle VoIP calls to a traditional phone
line, I am wondering how each VoIP call to the PSTN connection get
charged by a local Telecom.
Thanks
Sam
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Jean-Michel Hiver wrote:
Sam a écrit :
Hi,
If I setup an IP PAX gateway to handle VoIP calls to a traditional
phone line, I am wondering how each VoIP call to the PSTN connection
get charged by a local Telecom.
I am not really sure to understand the question. But assuming you
to discuss charging with your Telco.
PaulH
_
Thanks for the answers. I really appreciate that. It may be better for
me to talk to local Telco for further price negotiation.
Thanks
Sam
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IP PAX
(or PSTN) gateway to end phone. Assuming that there are 1000 VoIP
calls thru Telco's PSTN to end phones, how will these calls get
calculated? is the charge will be per-call basis?
Sam, I am still unsure to understand your question :-/
How much your telco is going to charge you
will just need to pay each VoIP call
to phone line at 20 - 30 cents / call.
Thanks
Sam
PaulH
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through a VOIP termination service is also good for testing.
When going thru a VoIP termination service, do I also need to have a IP
PAX gateway?
Sam
There are quite a few here in Australia.
PaulH
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If you can, try giving the highest priority to the UDP protocol or the
provider IP address.
Sam
everyones schedule to line up, I don't want to go through the
trouble if I will just be disappointed.
Thanks,
Sam
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Actually I think they will all be calling in using regular pstn phones
and cell phones.
Sam
Al Baker wrote:
The 2 big questions are:
-Are all participants using QoS end to end ?
-Are all of them using the SAME CODEC. As the amount of Transcoding goes up,
the work on the * box goes up
is coming from asterisk and not the ata. And all the dtmf modes
are rfc. Any one have any tips to further trouble shoot this?
Sam
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confirm yet if it has fixed my dtmf talk off problems, but I have
not had any problems navigating through company ivr's (of course I
didn't before either.)
Sam
Christopher Gray wrote:
Hello:
I need to be able to reliably send out touchtone to any calling party who
comes
into my pbx
Christopher, did you receive the email that I sent to your yesterday?
It was delivered Jan 23 20:47:31 -0600. Maybe it went to your junk box..
I will try again.
Sam
Christopher Gray wrote:
Hello:
Yes, DTMF can be a problem on the phones themselves as Sam observed, and
inband
can help
Muiz Motani wrote:
I am experiencing choppy sound when I bridge from a SIP peer to an IAX
peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am
experiencing choppy sound from the SIP peer to the IAX peer but not
vice-versa. I know that this is not a bandwidth issue because I
/magicjacks-eula-says.html
Sam
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Hi,
Is there any specific made embedded hardware designed VoIP software or
Asterisk? I want to build a router that have VoIP enabled, so that I can
use it connect to a VoIP ISP.
Thanks
Sam
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Hi,
I want to build a PBX base on Asterisk using an embedded device.
Can anyone please recommend an embedded device I can use for doing so?
I will install linux or freebsd in the device.
Thanks
A
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Hi,
I want to build a PBX base on Asterisk using an embedded device.
Can anyone please recommend an embedded device I can use for doing so?
I will install linux or freebsd in the device.
Thanks
A
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mustardman29 wrote:
Not a lot to go on sam. What do you want to do? If you just want to play
or have very minimal requirements then get a soekris NET4801 board, CF and
install Astlinux.
http://www.soekris.com/net4801.htm
-Original Message-
From: sam [mailto:[EMAIL PROTECTED
Jim Houser wrote:
http://gumstix.com/waysmalls.html
Thanks for your link. how to build asterisk into this hardware?
Thanks
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sam
Sent: Friday, March 31, 2006 8:01 AM
To: asterisk-users
Somewhat off topic but does anyone know if the price for the license
will go down in the future? It seems strange that I can use skype for
free on my computer but to put it on asterisk cost $66...
Sam
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Just a note, the asterisk mailing list server continually gets
blacklisted over at
http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
mail to spamtraps. Perhaps something needs to be looked into...
Regards,
Sam
On 08/21/2015 12:52 AM, Sam wrote:
Hello,
I have what I would think would be a common situation: I run asterisk at
home simply as a land line. I started a new job working remotely and
they gave me a SIP account with user name, domain, and proxy. I've never
had to deal with sip domains before
://tinyurl.com/ouy2ajr
Regards,
Sam
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asterisk
Hi all
Has anyone managed to send an outgoing call using
asterisk-h323 and successfully sent the H323_id ?
Sam
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i am using the sample config files and get a dial tone. i have also
gotten it to play greetings etc, but i need the phone to ring so that
i am not tieing up the one phone line, please help, i know this sounds
insanely stupid but i cant get it to work.
:16:39 WARNING[1089170112]: chan_unicall.c:634
unicall_error: UniCall: mfcr2 release_guard_expiredJan 10 16:16:39
WARNING[1089170112]: chan_unicall.c:2548 handle_uc_event: UC event Release
call -- UC channel 4 released
I am using channel Unicall
compiled without errors on Redhat 9
/Sam
-
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 10, 2005 4:35 PM
Subject: Re: [Asterisk-Users] Unicall errors
Hi Sam,
It looks like you have not set the default signalling bit pattern
Sent: Monday, January 10, 2005 5:38 PM
Subject: Re: [Asterisk-Users] Unicall errors
Hi Sam,
You are missing libsupertone, which you can find in the same place as
the other stuff.
Regards,
Steve
Sam Njenga wrote:
I tried to compile asterisk after updating unicall and got the following
: uc_channel_write
Jan 10 18:37:16 WARNING[1076216448]: loader.c:380 load_modules: Loading
module chan_unicall.so failed!
[EMAIL PROTECTED] asterisk]#
Sam Njenga
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk
What Linux distribution are you using ? I can help you if your using redhat
9 as am 90% done with R2.( Thanks to Steve Underwood)
You can start here
http://www.opencall.org/installing-mfcr2.html
/Sam
- Original Message -
From: Carlos Chavez [EMAIL PROTECTED]
To: Asterisk asterisk-users
chan_unicall.so failed!
Sam Njenga
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 11, 2005 3:17 AM
Subject: Re: [Asterisk-Users] Unicall errors
Hi Sam,
Did you
chan_unicall.so failed!
Sam Njenga
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 11, 2005 3:17 AM
Subject: Re: [Asterisk-Users] Unicall errors
Hi Sam
module chan_unicall.so failed!
Sam Njenga
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 11, 2005 3:17 AM
Subject: Re: [Asterisk-Users] Unicall
Hi Steve
Did that but still the same error :-(
PS. There is now unicall-0.0.2pre3. What are the changes in it ?
/Sam
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent
Hi
Am setting up * with R2/MfC support but am 90% done. I seem to be missing
something in my setup. Can you tell me what Linux distribution and the
packages you have used to complete your setup to a working level ?
/Sam
- Original Message -
From: Miguel Cavazos [EMAIL PROTECTED
://www.bashton.com/adm/adm-0.5.tar.gz
Please note that although this program uses some of my code (from
autovol) I did not write it, and all notes of thanks, complaints and
the like should be directed to Rick - rick [at] hamnett.org.
--
Sam Bashton - Bashton Ltd
www.bashton.com
If you buy the codec, it will do conversion...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of NetOne
Administrator
Sent: Friday, January 16, 2004 3:24 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] G.729 quiestion
Hi all!
If i purchase the G.729
the hangup and ending the connection?
Sam
smime.p7s
Description: S/MIME cryptographic signature
Zaptel was the version from about 4 days ago when I sent this message, I
updated again yesterday night
Sam
Quoting Martin Pycko [EMAIL PROTECTED]:
How old is your zaptel code ?
Mark recently increased some timer for that.
Martin
On Wed, 18 Jun 2003, Sam Bingner wrote:
I have
--
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Thursday, June 19, 2003 11:57 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoiceMail recording dialtone
Well experiment yourself with the code.
in wcfxo.c
OK, I tried upping it to 2000.. See if that changes anything
I still don't understand why it would end up directly in voicemail if it
picked the line back up instead of calling extension s again if the
telco's hangup signal was interpreted as a ring?
Sam
-Original Message-
From: [EMAIL
Title: Message
Hi,
I am new to Asterisk
and was wondering if Asterisk has the ability to act as a protocol
converter.
I have an H.323
network and I want to know if Asterisk can convert the signaling to SIP so I can
send it to SIP Addresses?
Thanks for your
help!
regards,
Sam
Michelson
That would be correct, I posted it to list a few times, but my wiki
account is broken (bleh) so I never posted it there
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mario Velasco
Sent: Wednesday, June 16, 2004 9:19 AM
To: [EMAIL PROTECTED]
Subject
It's exiting before the output finishes printing, it's a known bug and a
timing issue There's a patch I put that's a hack to put in a timeout
for exit, you should be able to find it on bugs.digium.com
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
everything yet.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Saturday, June 19, 2004 10:17 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] asterisk -rx not working well
On Sat, 2004-06-19 at 07:05, Sam Bingner
Search the mailing lists, this has been answered a million times.
Edit the Makefile and remove the entries for both app_rxfax.o and
app_txfax.o and it will compile fine.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke
Sent: Saturday
this? Or have a suggestion
for further investigation?
Cheers.
--
--
Sam Tilders
[EMAIL PROTECTED]
(Move to Jupiter)
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for an echo canceller and
fairly inconsistent. Or perhaps that's normal for echo cancellers?
- Sam
On Thu, 24 Jun 2004, Sam Tilders wrote:
Hi,
Despite much searching, I can't find anything quite like the problem
we seem to be experiencing with our recently activated asterisk pbx.
Of four
to
postmaster's command line. Something like:
/usr/bin/pg_ctl -p /usr/bin/postmaster -o '-p 5432 -i' start
with whatever other options are already there.
Then after a restart, asterisk should be able to connect to
postgres.
--
--
Sam Tilders
[EMAIL PROTECTED]
(Move to Jupiter
and phones that are always members can help there.
If anyone knows about how to do proper timeouts when there are no
queue members to call I'd like to hear about it.
--
--
Sam Tilders
[EMAIL PROTECTED]
(Move to Jupiter)
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getting?
--
--
Sam Tilders
[EMAIL PROTECTED]
(Move to Jupiter)
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or pointers to online docs
would be appreciated.
Thanks,
Sam
P.S. Thanks to Jsmith for the fast, simple answer to my last question
re: version number in CVS not updating.
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it will die. This simplifies life for asterisk.
Mark, if you think this patch is stable feel free to apply it... You have
my waiver already.
Sam
moh_sox.diff
Description: Binary data
smime.p7s
Description: S/MIME cryptographic signature
I have a working MP3 decoder in a thread, using libresample and
libmp3lame, but I'm not really happy with it yet Not sure about the
legalities but if anybody wants to try this work in progress just let me
know
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
Hi all,
I'm having a problem with * being very finicky about the length of
DTMF key-presses during menus, voicemail, etc. Basically, short (100
ms) tones are ignored, anything between 100ms (or so) and about 300ms
is correctly detected, and anything 300ms is interpreted as multiple
presses of
sure you have
'canreinvite=yes' set in the appropriate section of your sip.conf.
--
Sam Bashton
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Hi all
I have MFCR2 successfully
installed but seems to get warnings a s seen below when I start asterisk. Am
running on Redhat 9.
Asterisk Ready.*CLI Dec 20
08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2
far_unblocking_expiredDec 20 08:40:38
welcome.
http://www.bashton.com/autovol/
--
Sam Bashton - Bashton Ltd
www.bashton.com
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PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, December 20, 2004 4:33 PM
Subject: Re: [Asterisk-Users] MFC/R2 errors
Hi Sam,
You can ignore that. Its just debug. The next version will make it
configurable.
Steve
Sam
and the amazing thing is that when
I connect through ip directly I don't get any errors and I can hear the
prompts
/Sam
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, December 20
You need to install the mysql-devel rpm if you use redhat
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown
Sent: Sunday, March 28, 2004 2:13 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Error installing/compiling cdr_mysql addon
* listens for fax tones as soon as you Answer() the line. If you Answer
the line before ringing the local lines, it will actually detect fax tones
while in the Dial statement.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird
Sent: Sunday
is in order, so it
is not an Asterisk problem.
Please Help!
Thanks,
Sam Bacsa
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No, the voice mail runs without any
hardware.
Check out http://www.voip-info.org/ for information
about implementing voicemail into Asterisk.
- Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
PTCHENSent: Sunday, April 04, 2004 8:18 PMTo:
[EMAIL PROTECTED]Subject
and "choppiness" as if there were packetloss or lots of
jitter.
I'm using SIP for outgoing on Cisco
7960.
Any ideas on fixes or what may be causing this
problem?
Thanks,
Sam Bacsa
If you have a new enough version of the IAXy firmware on the IAXy, then it
will automagically be upgraded as soon as * sees it has an old firmware
(via the IAX protocol) --- if you don't have a new enough version, digium
has to do it by what I've heard
Sam
-Original Message-
From: [EMAIL
It's worked good for me... Only had a garbled page once when it was a 15
page fax, and that was a few versions ago so I'm not sure if it would do
the same now or not
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Enger
Sent: Sunday, April 18
So I went to go compile the Zaptel library from the HEAD CVS and I get
some really really odd errors which don't make any sense. I've attached
the console output ... any idea why this is going on and how to fix
this?
Thanks,
Sam Bacsa
SNIP
[EMAIL
You can get an upgrade contract with Cisco for like $8 or
something to download the SIP firmware for your phone.
So no, no waste of money -- unless you bought the wrong
phone.
- Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
TyremanSent: Thursday, April 22, 2004
Use ulaw
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedro Vela
Sent: Friday, April 23, 2004 7:52 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Fax problem
Hi,
We have a machine with an *'s with Digium TDM400P and connected wit other
machine
Title: Message
Even
if you could get that to work properly, which I dont know... the callprogress
detection is horrible; if you want to do that reliably you need a T1,ISDN or
IPinterface to the switch (something that actually provides proper call
progress)
Sam
-Original Message
Really you should link /usr/src/linux-2.6 to /lib/modules/`uname -r`/build
then you don't have to do anything special and it'll build... That
directory and all the files in it are installed by the kernel rpm, you
don't even need kernel-source for it... Although I haven't tried compiling
without
You should Answer() your calls... In the 5000 exten, you could move your
Answer to after the dial if you like... And the h exten hangs up if it
doesn't exist so that's redundant, but not bad
Sam
[internalexten]
exten = 5000,1,Answer()
exten = 5000,2,Dial(SIP/mike,60,tr)
exten = 5000,3
think.
You could always code it in ;)
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Graham Turner
Sent: Sunday, May 23, 2004 9:09 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] extension pattern matching
dear all, was hoping someone could give me
Change your symlink to not point to the linux source tree, but rather
point at /lib/modules/2.6.5-358/build, and just do a make linux26
Or apply this patch to your makefile...
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Taz Man
Sent: Sunday, May
Add the path to it to /etc/ld.so.conf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone
Sent: Friday, May 28, 2004 7:14 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] spandsp wont compile.
got it to load but now it errors when starting
Did you actually look at that patch? --- It fixes some bug in 3.6.1
related to faxing... If so, sorry for wasting all your bandwidth :b
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: Monday, May 31, 2004 1:53 AM
To: [EMAIL
You shouldn't put /usr/include in ld.so.conf, needing to do so means you
have something installed wrong... And I've never heard of anything getting
installed that wrong ;)
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone
Sent: Sunday, May
not
a developer so if you have a ready I'd be glad to have it .
Regards
Sam Njenga
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it to
Argentina R2 . Am not a developer so if you have a ready I'd be glad
to have it . Regards Sam
NjengaThere is no support in * for the
D300-E1.Regards,Steve
Actually the card is
Digium Wildcard E100P E1/PRA Card
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This will be very helpful Steve. Kindly provide us
with the location where we can download it
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(finally!)
in their 7.x SIP firmware releases. I actually upgraded all our 7940s
today for this very reason.
--
Sam Bashton
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...
Actually, we should be able to get a pretty good idea on that by using two
gigabit interfaces and VoIP?
Sam
Quoting Andy Hester [EMAIL PROTECTED]:
I talked to Imagestream this morning about the possibilites. Their lead
engineer said that there would be no way to do voice over their
DS-3 cards
been reversed so I can neither confirm nor deny this.
Sam
Quoting Dan [EMAIL PROTECTED]:
Hi,
I have a FAX machine connected to a TDM400 card FXS port.
When I receive a fax call through X100 and transfer it to that extension,
the FAX machine display REC, but nothing happen (no fax received
Also, if you ONLY run * on the system, you can lock it down so that the
security bugs are pretty much non-exploitable... Ipchains/etc. You don't
even HAVE to run ssh or any remote management if you want to to be just
like a regular PBX system
Sam
-Original Message-
From: [EMAIL
Why not get a proper GSM Gateway .
We have some for sell for £60 each ..
Contact me on sam AT cyber-telecom DOT net
Or visit cyber-telecom DOT net
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Rojas
Sent: Sunday, January 22, 2006 2:03
AM
I don't think there is any laws on GSM Gateway itself.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Augustyn
Sent: Friday, January 06, 2006 11:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
Why not get a asterisk or Normal VoiP Gateway and then connect those
together .
Sure that will still cost less than 300 USD
and then you can run sip or iax or h323 on it.
Email me on sam AT cyber-telecom DOT net for more info
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto
Why not try to purchase one of our GSM
Gateway at £60 and then you can route all the mobile calls through the GSM
Gateway?
http://cyber-telecom.net/store/product_info.php?products_id=29osCsid=4e787773c7c03212c43c51368d6ae387
Sam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Do anyone know how to setup asterisk to authenticate the user through IP
rather than username and password?
I know most carriers will do that but smaller end user providers will not
do.
Sam
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The long waited Ultimate GSM Gateway is finally out. This time we have managed
to source a new patch of brand NEW GSM Gateway at prices that is only 50% of
what the market rate. And with the SMS Function and many more...
For purchase please email gsm AT cyper-telecom.net. We accept paypal and
plug
Antenna connection: SMA antenna tie-in, N type port(optional).TNC
port(optional)
For more info please email gsm AT cyber-telecom.net for more info or visit
www.cyber-telecom.net to purchase right away.
Sam
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t;, "") in new stack ==
Spawn extension (default, 400, 3) exited non-zero on
'SIP/203.125.68.66-081ee3d8'
Asterisk*CLI
Any idea what is
this all about ?
Regards,Sam
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Asterisk-Users maili
k refuses to take anymore call at
extension 400 for VoicemailMain() . Please let me know if you don't understand
what i mean.
Please
help!
Regards,Sam
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Asterisk-Users mailing list
To UNSUBSCRIBE or
You can even set it to zero. Mine works well when in zero. The line pick up
immediately :
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton
Sent: Thursday, February 09, 2006 6:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
We have got some ATA for only $55 if you are interested?
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Sampson
Sent: Thursday, February 09, 2006 11:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk
I think this is a
question that has been discussed before.
But you see nowadays most carriers will provide thing like SIP using IP
authorization rather than username and password and I am now wondering whether
Asterisk can do something like that or not?
Sam
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