I am planning to sell prepaid calling cards to my service.
The system is already working but I wanna print a good quality
prepaid calling cards for it.
Anyone would recommend me a good and cheap pre-paid card printing company
anywhere in the world?
Thanks in advance,
Isamar
_
Your website is refusing connections at the moment. Or more properly I
should say I get "Connection refused" when I try to access the Cepstral
link you posted earlier today to the Asterisk-users list.
FYI.
Thx.
B.
___
Asterisk-Users mailing list
[EM
You have not described your hardware configuration. Zapata.conf is in
general used for Digium cards. If you have not such card you can just say
noload => chan_zap.so
in your /etc/asterisk/modules.conf
You can obtain more information about asterisk configuration from next sources:
a) http://www.
On Wed, 4 Feb 2004, John Todd wrote:
> At 11:50 PM + 2/4/04, Dan Tucny wrote:
> >;
> >; Talking clock (123)
> >;
> >exten => 123,1,SayUnixTime(|GB|HM 'vm-and' S 'digits/seconds')
> >exten => 123,2,Wait(1)
> >exten => 123,3,Goto(1)
> >
> >the seconds sound can be picked up from John Todd's site
yOn Thu, 5 Feb 2004, Deepakumar JV wrote:
> Thanks to everyone.
>
> I got the talking clock working the way i wanted.
>
> thanks again
> Deepak
How about a followup post showing exactly what your extensions.conf
entries look like, and what you had to go to get it twekaed to your
satisfaction?
Hi
everyone
I have TE410P with
one E1 link connected to telecom PSTN, and another E1 to my internal legacy PBX.
On this PBX I have one extension where my RAS server for both ISDN and analogue
calls is located.
Can anyone tell me
what has to be done to transfer voice call from one E1 to
Hi Tomica,
i had the same problem and here is the solution from Maik Schmitt:
exten => _X.,1,GotoIf,"$[${CALLTYPE} = DIGITAL]?50:100"
exten => _X.,50,Dial(Zap/g3d/${EXTEN})
exten => _X.,100,Dial(Zap/g3/${EXTEN})
But maybe the dataendpoint would never be reached, and so can try out this:
go to b
mattf wrote:
I have all of my Polycom's set to friend so I know that's not your problem.
One day you too will get bitten by the type=friend's EVIL and you will
see the light.
Trust me,
Jeremy McNamara
___
Asterisk-Users mailing list
[EMAIL PROTE
Could you tell us a little bit how exactly it works? The wiki pages don't
say much about type=friend, user, and peer. I tried using type=user but
can't seem to register.
And what implications are there for using type=friend?
David
- Original Message -
From: "Jeremy McNamara" <[EMAIL P
On Thu, 5 Feb 2004 [EMAIL PROTECTED] wrote:
> Hi
>
> I wonder if anyone has a fix or any advice for the IAX2
> jitter buffer.
>
> My internet connection here in South Africa has an
> international ping time of 550ms +- 50 ms. According to the
> scientific approach I would like to add a 100ms j
Hello
I am trying to use a VOIP provider (PC to
PSTN).
Is it possible to use asterisk as a client
and make calls via a H323 provider?
Can anyone guide me how the oh323.conf
should be and extension.conf should be.
I have a IP, userid and password given by
them.
I am using www.mywebc
> How about a followup post showing exactly what your extensions.conf
> entries look like, and what you had to go to get it twekaed to your
> satisfaction?
Here is the working extension.conf i came up with
[time]
exten => 5559,1,Answer()
exten => 5559,2,Playback(time)
exten => 5559,3,SayUnixTime
David Liu wrote:
Could you tell us a little bit how exactly it works? The wiki pages don't
say much about type=friend, user, and peer. I tried using type=user but
can't seem to register.
A type=friend is simply both a type=user and type=peer using the same
set of config directives. While a t
On the subject of South Africa
What are the laws regarding using the Internet to carry telephone traffic?
What are the laws regarding connecting digium kit to Telkom equipment?
As I recall they are quite restrictive, have they been eased up a bit?
Regards
Chris
_
Anyone have comments on this? Really could use some suggestions or ideas
why this is happening. Thanks.
Rich
> Anyone recognize this as a sip logic bug?
>
> Example Case:
> C7960 -> * -> sip gateway -> pstn
> (sip gateway config'ed with canreinvite=no, but shouldn't ha
I would have thought that if that was the problem, we couldn't makle or
receive calls at all, or that we at least couldnt use all 3 Zap cards at the
same time, but we can.
The problem only happens every so often, but recently it's getting more and
more frequent... management are starting to get pi
Steve hi
Yup, adsl, seems to be getting slower by the day.
Maybe we can configure * to change the iax to port 21 udp ?
Regards
Clive
On Thu, 5 Feb 2004 13:21:08 +0200 (SAST)
Stephen Davies <[EMAIL PROTECTED]> wrote:
>
>
> On Thu, 5 Feb 2004 [EMAIL PROTECTED] wrote:
>
> > Hi
> >
> > I won
Does anyone have any documentation on Asterisk + oh323, I am trying to allow a H323
peer to send me calls that I want to push out to SIP phones but am having trouble
passing the digits dialed from the oh323 peer and dialing those digits onto a SIP
client.
Any docs much appreciated or even bette
Basically voip is only legal if used between branch offices
of a company that are connected using leased lines.
Archaic.. yes, stupid... yes, but thats the law here..:(
Our telco is strangling the country so they can line their
pockets.
On Thu, 05 Feb 2004 11:57:57 +
Chris Lee <[EMAIL P
Hi all,
by chance I have found an old Dialogic D300SC-E1 card that I would
like to test with Asterisk.
It should have voice capabilities on board, also.
I have ABSOLUTELY no idea regarding the steps to make it work, I
installed the card in a server with a working installation of *, then
browsed
Chris Lee wrote:
On the subject of South Africa
What are the laws regarding using the Internet to carry telephone
traffic?
Its 100% against the law, Telcom have the monopoly there still that
requires ALL voice trafic to go via the Telcom network.. The Mobile
phone operators there are in consta
[EMAIL PROTECTED] wrote:
Basically voip is only legal if used between branch offices
of a company that are connected using leased lines.
Archaic.. yes, stupid... yes, but thats the law here..:(
That is provided the leased lines are operated by Telkom.. ;)
There is no getting away...
_
Is it posible to make Asterisk execute a command on extensions.conf during a
call ¿ (That's to transfer H323 call by telnetting the gatekeeper so
Asterisk doesn't seem to like transferring h.323 )
Thanks!
Marc
___
Asterisk-Users mailing list
[EM
Alessio Focardi wrote:
Hi all,
by chance I have found an old Dialogic D300SC-E1 card that I would
like to test with Asterisk.
It should have voice capabilities on board, also.
I have ABSOLUTELY no idea regarding the steps to make it work, I
installed the card in a server with a working installat
Steve,
Since I have a rather short memory and receive about 250 posting per day, I
don't have a clue what has/hasn't been suggested. Here's a couple:
1. in logger.conf turn on debug, watch /var/log/asterisk/debug for size, and
and hints relative to the dropped calls
2. look at /var/log/asterisk/
Rich,
Try it again after executing: "sip debug" and give us the actual SIP
messages. The devil
is usually in the details.
Rich Adamson wrote:
Anyone have comments on this? Really could use some suggestions or ideas
why this is happening. Thanks.
Rich
Anyone recogn
So do the 7960's have to be on the same subnet as the * box ?
This seems like a major detriment to using them in a typical wan
environment.
- Chris Clifton
- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, February 04, 2004 1:58 PM
Subj
On Thu, 5 Feb 2004, Chris Lee wrote:
> On the subject of South Africa
> What are the laws regarding using the Internet to carry telephone traffic?
> What are the laws regarding connecting digium kit to Telkom equipment?
> As I recall they are quite restrictive, have they been eased up a bit?
Th
Chris Clifton wrote:
So do the 7960's have to be on the same subnet as the * box ?
This seems like a major detriment to using them in a typical wan
environment.
- Chris Clifton
- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, February
I followed the wiki instructions:
http://www.voip-info.org/wiki-Asterisk+non-root
Now I have a working asterisk running as user asterisk.
I do however have some problems:
1: I dont have access via asterisk -r
2: The pid file is no longer being updated
3: I want to create a file in init.d so that
No they do not. I am managing an installation running 7960 SIP release
6.0 and the phones
are on about 4 different subnets. Half of these are on remote VPN
connections at people's homes.
Chris Clifton wrote:
So do the 7960's have to be on the same subnet as the * box ?
This seems like a major
And the * server is in your hq location ?
Thanks,
Chris Clifton
- Original Message -
From: "Clif Jones" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, February 05, 2004 9:02 AM
Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
> No they do not. I am
On Thu, 5 Feb 2004, Clif Jones wrote:
> No they do not. I am managing an installation running 7960 SIP release
> 6.0 and the phones
> are on about 4 different subnets. Half of these are on remote VPN
> connections at people's homes.
>
Currently, The Cisco 7960 SCCP can hear me but I cannot hear
No, they don't. I've got multiple 7960's functioning reliably across the
Internet registering (and handling calls) just fine with * for months. And,
the 7960's are behind cheap nat boxes as well. All running v6.0, but worked
just as well with the v4 code. (Other 7960's are on the wire with * too.)
History:
1. Added X100P to my system
2. Added Sipura
3. Added TDM400P (2 port)
Worked fine so far
4. Now I want to add an additional X100P
My question is...is the following configs files ok and is there any issue
with adding the X100P (channel 4) after my 2 analog FXS channels?
Thanks.
Steve
Right... It just happened there now, this came up:
Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call [EMAIL
PROTECTED] for seqno 3 (Response)
I'm not sure if that's related to it, but it's the only thing that came up
when the call got cut off.
Here's the generic sip.conf stu
The asterisk PBX is on a private subnet 192.168.20.0, the 7960's are on
192.168.{20,200,201,202,203}.0
subnets. The SIP gateways are on 192.168.{15,20,22,13}.0 subnets.
Chris Clifton wrote:
And the * server is in your hq location ?
Thanks,
Chris Clifton
- Original Message -
From: "Cli
Hello Jeremy,
>>Anyone can help me starting the card ?
JM> List it on http://www.ebay.com/ and take the proceeds and purchase a
JM> Digium E100P card.
It has been my first tought but guess what ? E100P is not CE
certified and I'm fearing legal problems
Also I dont think that someone
I purchased a type 66 punch down with a Amp50 connect at Graybar. I
also purchased a male/female Amp50 cord. You may also need some bridge
clips. Use the punch down block to cross/connect to your lines. Row
one on the block is channel one on the Adtran, row 24 is channel 24.
You might consi
Hi All!
I´m searching for a compact external fxo
device , a little box like sipura adaptor, with one or maybe
two fxo.
Searching google the only device that shows is the
x100p,
Anyone knows about a device like that?
miklos
No they do not but apparetly his phones either didn't like the switch they
were on or they have something wrong with them.
bkw
On Thu, 5 Feb 2004, Chris Clifton wrote:
> So do the 7960's have to be on the same subnet as the * box ?
>
> This seems like a major detriment to using them in a typical
Hi,
Does anybody know if it is possible to record a
conversation with asterisk ?
Regards
Rattana
res_monitor.so: Resource for
recording channels.
-Original Message-From: Rattana BIV
[mailto:[EMAIL PROTECTED]Sent: 05 February 2004 16:20To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Record
conversation
Hi,
Does anybody know if it is possible to record a
con
bkw,
I realised that I was running asterisk with just asterisk no cli options
changed it to safe_asterisk any my problem went away, so it might just be that it
doesn't want to work in asterisk, just safe_asterisk
when I some free time I'll get a coredump since there are no real informative debug
Clif and all...
At the bottom of this post is the "sip show debug" for the problem.
The underlying problem (again): when C7960 hangs up on working conversation,
the C7960 sends a BYE to *, but * never sends a BYE to the sip gateway.
Any suggestions would be greatly appreciated.
Rich
> Try it ag
Alessio Focardi wrote:
Hello Jeremy,
Anyone can help me starting the card ?
JM> List it on http://www.ebay.com/ and take the proceeds and purchase a
JM> Digium E100P card.
It has been my first tought but guess what ? E100P is not CE
certified and I'm fearing legal problems
Steve Foy wrote:
> Right... It just happened there now, this came up:
>
> Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call
> [EMAIL PROTECTED] for seqno 3 (Response)
>
> I'm not sure if that's related to it, but it's the only thing that
> came up when the call got cut off.
I had a previous error where, due to a faulty switch port, one of my
7960's was rebooting or locking fairly often. That was due to a
physical, electrical error.
This problem is significantly different. A fully-loaded (all six
lines) 7960 will gradually stop registrations to one of my (distant
At 3:05 AM -0500 2/5/04, Greg Boehnlein wrote:
[snip]
> [snip]
The file "seconds.gsm" is also in asterisk-sounds, which along with
many other interesting and amusing clips can be pulled from the CVS
server just like asterisk, zaptel, etc.
Kudos to whomever requested "All your base are belong to
Jeremy,
There is one small flaw in your reasoning with the need to register. You
said :
"You only need to register to Asterisk if you have a dynamic IP address
or you need to blow thru a firewall/NAT device"
But this is not true if you want to maintain true presence information.
If you do not r
>> I'm still getting a sementation fault with mpg123.
>Ah, adventures in the pubic school system.
Funny you should mention this... there is a large Hasidic community in our
town. The boys refused to ride in busses driven by women. Eventually a
separate school district was created (from the one I w
On Thursday 05 February 2004 05:50, Jeremy McNamara wrote:
> A type=friend is simply both a type=user and type=peer using the same
> set of config directives. While a type=friend makes things almost
> trivial to get calls working in both directions, it will limit the
> flexibility of your config an
On Thursday 05 February 2004 08:03, Chris Lee wrote:
> I followed the wiki instructions:
> http://www.voip-info.org/wiki-Asterisk+non-root
>
> Now I have a working asterisk running as user asterisk.
> I do however have some problems:
> 1: I dont have access via asterisk -r
Permissions problem. Us
Ejay Hire wrote:
Hi.
Low Temp Hot glue is what I use on my robots.
Stay away from silicone (conductive) and rtv (peels traces
off cheap pcb's)
The only silicones that are electrically conductive are those that are
loaded with some conductive material (like silver). These are rather
esoteric a
Hi
A colleague of mine read somewhere that it was possible to compile Asterisk
under OS-X which he has just tried with little success. Has anybody here had
any success and if so what things should my colleague take into account?
Regards
Martin
___
Tilghman Lesher wrote:
Permissions problem. User asterisk needs to have permissions to
write the file /var/run/asterisk.ctl
2: The pid file is no longer being updated
Again, permissions problem.
I was under the impression that changing the line:
ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run/asterisk
I have some phones that purport to handle this properly but am having
quite a time figuring out just when to expect it to work and when not
to. Limited to Zap channels? Zap and SIP, but in different manners?
Sieving the list archives yielded more questions than answers.
Pointers or discussion
Rich,
It is very important (at least to me) to have the whole SIP call flow.
That is, I must see the initial
INVITE come from the originating phone all the way to the last message.
I can only speculate at
this point but it appears that the second leg (destination) may never
have ACK'd the cal
Hi
A colleague of mine read somewhere that it was possible to compile Asterisk
under OS-X which he has just tried with little success. Has anybody here had
any success and if so what things should my colleague take into account?
Regards
Martin
*CLI> show version
Asterisk CVS-10/24/03-01:48:29 bui
Title: Message
Hello all, I am
trying to figure out how to have * release a phone call. We are noticing
some call quality issues on people who have a "find-me" feature, and answer the
call through a cell phone. Here is the call path we are seeing, and all
VoIP connections are using SIP.
Title: Message
I don't really have a answer for you on you issue
but have a question about what "find-me" is. I see it on the feature list but am
unable to find any real information about it. Is this simply call forward or is
their more to it.
thanks
- Original Message -
From
Title: Message
The
way we have it setup is simply calling multiple numbers/channels. It is
either setup manually in the configs, or through a very ugly menu interface I
constructed.
B.
J.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Be
Title: Message
Well after thinking a little more about your
scenario I have had situations where I have hairpinned calls back to the device
they have come in on and in my experience without canreinvite=no the device will
usually complain that a loop was detected. In situations were I am
usin
Anyone have any
experience configuring VegaStream's with Asterisk.
I have run
into a few of questions. 1. It appear that after turning on
registrations I am seeing two request for registration per
linesip:[EMAIL PROTECTED]sip:[EMAIL PROTECTED]What is
purpose and how do I handle this?2. DT
On Thursday 05 February 2004 11:13, Chris Lee wrote:
> Tilghman Lesher wrote:
> > Permissions problem. User asterisk needs to have permissions to
> > write the file /var/run/asterisk.ctl
> >
> >>2: The pid file is no longer being updated
> >
> > Again, permissions problem.
>
> I was under the impr
Sorry Clif, as a professional working with protocol analysis at corporations
in more than 40 states, I should have known better. Never gave it a thought
the issue could have been earlier in the call/session setup. I'll dig into
that, and if still need help/suggestions will post the full debug trace
(Please forward this to Martin Pycko in Digium)
Martin, this is all about mail that I have sent to you regarding data
call setup.
Hi Thomas,
Thanks for your hint. I have tried it but it doesn't work. Here are few
lines from my config...
;
;RAS
;
exten => 290,1,GotoIf,"$[${CALLTYPE} = DIGITAL]?5
Hi,
I am very new to this forum, and to Asterisk
world.
I have two two X100P cards, and was trying to
setup something which looks like
phone line (PSTN) --> Asterisk X100P card -Asterisk (Linux Box 1)
<-->
Asterisk X100P card -Asterisk (Linux Box 2) -->
phone line (PSTN)
So I finally get an opportunity to try the data modes of the T400P but I
can't seem to get sethdlc to work or sethdlc-new to compile...
I've checked the docs, mailing list archives and the wiki but not found
anything useful to point me in the right direction.
sethdlc compiles but seems to be unusa
Hello,
I have made many changes/improvements/bug fixes to the Asterisk GUI client I
have written in Perl/TK and have released a third beta version on
sourceforge:
http://sourceforge.net/projects/astguiclient/
Here are the screen shots of the client application running on Linux and
Windows:
http
Hi,
I am trying to forward calls from one cisco gateway to another cisco gateway
using asterisk
cisco(5300)A 192.168.1.1
asterisk 192.168.1.2
cisco(5300)B 192.168.1.3
pstn --ciscoA-asterisk --ciscoB--pstn
I have the below in my extension.conf
[default]
exten => _1905XXX,1,D
You were close. Try this:
phone line (PSTN) --> Asterisk X100P card --> Asterisk (Linux Box 1)
<---NETWORK/INTERNET--->
Asterisk (Linux Box 2)--> Asterisk X100P card--> phone line (PSTN)
And use IAX2 to connect the two Asterisk boxes.
Try a bit of research on loligo's site and you'll get
Hello,
does anybody know how stable is OpenCall's fax sending/receiving
software? Is it still in development? I see only an old version on the
ftp.
Does anybody have any experienci with fax sending and the PC
performance needed for this?
What kind of PC hardware would I need when I would like t
Could someone point me to docs on how to set up a simple * test box
I just d/l and installed those rpms mentioned a couple of days ago onto a
fedora box
I hope to get simple config, and two softphones working with each other (one
windows and one linux)
Hopefully, such an article would include so
On Thu, 2004-02-05 at 14:03, Chris Lee wrote:
> I followed the wiki instructions:
> http://www.voip-info.org/wiki-Asterisk+non-root
Glad someone's finding it useful :)
> Now I have a working asterisk running as user asterisk.
> I do however have some problems:
> 1: I dont have access via asteris
You can have a look at wiki on iax trunking plus notes on setting up
x100p card.
David Kwok
Message: 5
From: "Maninder Bhatia" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Date: Thu, 5 Feb 2004 16:12:07 -0500
Subject: [Asterisk-Users] X100P - Asterisk - Asterisk - X100P setup help
Reply-To: [EMAIL
Hi,
We're just about to bring up Asterisk in a small business setting
with a broadband carrier. In this case, we have no reason to have
any POTS lines to make incoming and outgoing calls using our
SIP phones (Cisco 7960, 7940 and Grandstream 102.) We're
probably selecting Voiceglo simply because w
> 1. Can someone confirm whether Voiceglo needs to use SIP or
> can it handle IAX? This link seems to indicate it uses SIP:
> http://www.mail-archive.com/[EMAIL PROTECTED]/msg20561.html
> although other messages on the mailing list indicate that
> Voiceglo is using Asterisk in its internal archite
I can't find Allison saying "Do Not Disturb" Anybody got this If
not, is there a place to submit generic requests for sounds ???
-Lance
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Has anyone had good or bad experiance with http://www.oneunified.net. I
need a DID for incomming calls only. Nufone does not have service in my area(614-XXX)
:(
Anyone have worked with these people. Good comments bad comments.
Should we create a area in the WIKI for all of the VOIP providers
I have a few questions, and I'm hoping all of you nice people have an answer and can
share the info.
I don't need exact numbers, just asking for general info like yes, no, not as good as
expected, etc.
Has/is anyone selling Asterisk commercially? And is it successful or a flop?
Has anyone sold
Recently we established connection between two Asterisk systems using IAX.
Since then, we have observed that these boxes randomly stop
working. Checking the processes shows the Asterisk process is not running.
There is no error message on the remote consoles that we use to monitor
these boxes.
On Thu, 5 Feb 2004, Michael Swan wrote:
> We're just about to bring up Asterisk in a small business setting
> with a broadband carrier. In this case, we have no reason to have
> any POTS lines to make incoming and outgoing calls using our
> SIP phones (Cisco 7960, 7940 and Grandstream 102.) We're
>
Oneunified.net seems a little bit on the high side for pricing, no?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Thursday, February 05, 2004 5:43 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] http://www.oneunified.net
Has
> Should we create a area in the WIKI for all of the VOIP providers so we
> can leave comments about them someplace, and not take up mailling list
> time?
Many of the providers already have a page on the Wiki. (You can create one if not)
Please feel free to add comments to these pages about you
If you're serious about Apple and Linux and Asterisk, Kai Staats might be
worth talking to. He runs an Apple shop and has done some work in this area.
Below is my e-mail from some time back, along with his reply...
Charlie Hatchette
PS: We did NOT end up using Apple hardware, though TerraSoft prob
You forgot to mention they have incredibly poor web page Javascript. You
cannot use Mozilla to subscribe.
cameron.
On Thu, 5 Feb 2004, Greg Hill wrote:
> On Thu, 5 Feb 2004, Michael Swan wrote:
> > We're just about to bring up Asterisk in a small business setting
> > with a broadband carrier. I
I cannot recommend using Voiceglo for a business. Unless ringing and DTMF
start working in a sensible way. Call quality has been reasonable.
If you ignore that I would recommend:
g729 licenses for Asterisk
Broadband? Are you going to QoS to the broadband connection? How broad is
your broadband?
So Star-Six-Settings won't reboot the phone in this state?
cameron.
On Thu, 5 Feb 2004, John Todd wrote:
> I had a previous error where, due to a faulty switch port, one of my
> 7960's was rebooting or locking fairly often. That was due to a
> physical, electrical error.
>
> This problem is
I know this is fairly old thread, but I have a question regarding this.
The following line:
exten => s,2,GotoIf($[${autoattendant} = "1"]?auto|1)
Is basically saying goto context priority 1. So the last line also has
a goto to statement. When is this being trigered. So could you use the
same l
cvs checkout asterisk-sounds
bkw
On Thu, 5 Feb 2004, Lance Arbuckle wrote:
>
>
> I can't find Allison saying "Do Not Disturb" Anybody got this If
> not, is there a place to submit generic requests for sounds ???
>
> -Lance
> ___
> Asterisk-Users
Are you using iax or iax2?
If you are using iax1 please noload it and use only iax2.
bkw
On Thu, 5 Feb 2004, MLS Drop for SysAdmin wrote:
> Recently we established connection between two Asterisk systems using IAX.
> Since then, we have observed that these boxes randomly stop
> working. Checki
- Original Message -
From: "Michael Swan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, February 05, 2004 5:23 PM
Subject: [Asterisk-Users] Voiceglo questions
>
> 1. Can someone confirm whether Voiceglo needs to use SIP or
> can it handle IAX? This link seems to indicate it
Hi All!
Okay, let me understand this. No offense intended. I've struggled
for about a month with an Asterisk system, seeking to establish at
least the minimal functionality of the PBX we wanted to retire
(Nortel SL1). My objective was to try and use Asterisk as a
replacement/backup telephone s
IAX is what they use with glophone. http://webphone.voiceglo.com. It is a
seperate server from the myphone.voiceglo.com SIP gateway. The IAX gateway
is msps01-nyc.voiceglo.com on port 5036.
cameron.
On Thu, 5 Feb 2004, Jim Flagg wrote:
> - Original Message -
> From: "Michael Swan" <[E
Es posible que no tengas bien configurada tu interface zapata.???
etc/asterisk/zapata.conf
jorge
On Thu, 2004-02-05 at 00:29, FRANCISCO PEREZ-LANDAETA wrote:
> can anyone help me on this ?
> i am having problems configuring the asterisk.
>
> i have included an attachment because for some reas
I've seen its possible to use the System applications, but what about
passing arguments to the command ?
Thanks for your help!
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Marc Fargas
Enviado el: jueves, 05 de febrero de 2004 13:37
Para: [EMAIL PROTECTED
On Wed, 4 Feb 2004, Greg Boehnlein wrote:
> On Wed, 4 Feb 2004, Chris Tooley wrote:
>
> > Well, I don't really know all that much about SuSE either. I just
> > installed it about 19 hours ago for the first time.
>
> Well, depending on the version of RPM that they installed, you'll either
> nee
Correct. Wedged hard.
JT
So Star-Six-Settings won't reboot the phone in this state?
cameron.
On Thu, 5 Feb 2004, John Todd wrote:
I had a previous error where, due to a faulty switch port, one of my
7960's was rebooting or locking fairly often. That was due to a
physical, electrical erro
Eek.
cameron.
On Thu, 5 Feb 2004, John Todd wrote:
> Correct. Wedged hard.
>
> JT
>
>
> >So Star-Six-Settings won't reboot the phone in this state?
> >
> >cameron.
> >
> >On Thu, 5 Feb 2004, John Todd wrote:
> >
> >> I had a previous error where, due to a faulty switch port, one of my
> >>
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