Title: Message
flamed? I hope
not.
I have already started reading up on mysql and c and Perl and xml and
java and r... So many things I need to get working
so little knowledge of coding and so little time. All I can offer anyone
right now is good will
Howard Tarlow wrote:
Anyone know of any GUI's that can be used to manage/setup asterisk?
What features are you looking to have in the GUI?
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To
In United Kingdom, we have time based dialling pricing from most of
Telco's
based on time the call is placed! It is called PEAK (08.00- 18.00
Mon-Fri), OFF PEAK(18.00-08.00 Mon-Fri) and WEEKEND (all other times!
Could someone from any of other countries let me know if time based
charging exists
Nik Martin wrote:
All those numbers kinda negate the whole purpose of 3 digit
nationally standardized numbers, huh?
Of course... But also no more dial 9 for outside lines with
Properly thought of and configured asterisk box! :)
___
Asterisk-Users
User Mode Linux is way better for that use, much more efficient.
Matteo.
I am using user mode Linux very successfully to run as many asterisks as
I need. Besides asterisk, UML is my other favourite open source
project with which I am involved developing complete turn key solutions
(including
Fabio Donaggio wrote:
Hi to all!!
Here is my problem:
[EMAIL PROTECTED] root]# cd /usr/src
[EMAIL PROTECTED] src]# export
CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot [EMAIL PROTECTED]
src]# cvs login -bash: cvs: command not found
[EMAIL PROTECTED] src]#
Anyone can help me??
Chris Stenton wrote:
Tony,
The patches work great, picks up the BT callerid everytime.
A really big thankyou!
A big THANK YOU from me too!!!
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I have 2 SIP phones (Cisco 7960 XTen) behind a NAT provided by a
Linksys firewall that supports UPnP. The Asterisk server has a
public IP. Here are the problems that I am having with this
configuration...
1. The 2 SIP phones can call MeetMe and have a conference but
cannot call
I'm having a similar problem with snom 200s would changing the port
work there also or is that just a 7960 issue? Do you or any other
know where I would change that on a snom 200 ??
thanks in advance
Barry
___
try adding
Canreinvite=no
Brian Cuthie wrote:
I'm using Coloco now, which so far is working well.
Where companies like VoicePulse buy services from a patchwork of CLECs
in order to cover their markets, Coloco is a CLEC. The upside is that
you cut out the middleman. But if you need a number in an area they
don't
Another problem with the SIPURA is the lack of a working STUN
solution. Even Grandstream works better with NAT today. /O
I second that!!!
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Leif Madsen wrote:
Afternoon all,
I'm trying to load Asterisk, however I am getting the following error:
[skipping res_musiconhold.so]
[chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240
ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined
symbol: ast_moh_stop May
Title: Message
put:
mode=immediate in your zapata.conf file
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alberto
SatoSent: 18 May 2004 16:14To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] X100P
answer in first Ring
How I can do
in first Ring
Is
this the same thing as:
immediate=yes
-Original Message-From: Senad Jordanovic
[mailto:[EMAIL PROTECTED]Sent: Tuesday, May 18, 2004 10:21
AMTo: [EMAIL PROTECTED]Subject: RE:
[Asterisk-Users] X100P answer in first Ring
put:
mode
I am sure that quite a lot of people would like to have Caller ID
working with their X100P and TDM400P cards outside of USA.
Judging from previous threads this is just a matter of implementing this
support in the software driver!
So, I was thinking, if we get together and put few $(USA DOLLARS)
Title: Message
here
you go :)
http://bugs.digium.com/bug_view_page.php?bug_id=214
Ta
SJ
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
KirklandSent: 05 April 2004 04:25To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] New Call
Asterisk - MD wrote:
X-Analitica - MD-MailScanner-OpenProtect-Information: Please contact
the ISP for more information X-Analitica -
MD-MailScanner-OpenProtect: Found to be clean X-MailScanner-MCPCheck:
is it already inside * 0.7.2?
Yap...
___
Hi,
I am trying to get priority + 101 to work with Dial application.
My dial plan is like this:
[dial-mobile-peak]
exten = s,1,AbsoluteTimeout(${ABSOLUTETIMEOUT})
exten = s,2,Dial(${TRUNKONE}${CALLEDNO:1})
exten = s,103,AbsoluteTimeout(${ABSOLUTETIMEOUT})
exten =
Angus Berry wrote:
A quick search on eBay turned up this 4 port FXO external box for
US$299:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3087347715category=5
1279
...anyone know if it's compatible with Asterisk?
Yes.. I can confirm I had it setup and it is working great.
A quick hunt around the net shows this Ericsson unit on E-bay as
H.323 only. The price is good, but I'd rather have SIPactually
I'd rather have IAX2!
Michael
Yeah.. It is pain to set it up... But it does work very well...
And.. IAX2 box... :) would be very very nice..
Mark?
Now could Asterisk
- Be configured to allow the phone (ideally IP phone) to display which
real number is being called (ideally a name for that number)
( sales line ringing, support line, etc.)
YES
- Cause only phones part of a group to call if the number related to
that group rings
Andy Powell wrote:
- Let the caller know its position in the queue (ie: you are number
# in the queue, please hold and an operator will hang on you)
This is not possible at the moment.. Anyone know better?
Actually it is possible have a look at the bug tracker - I would
give you the
[EMAIL PROTECTED] wrote:
Ron,
It is a multi-reported problem, yet no resolution.
I would suggest it is a bug. I have had intermittent
success with POTS provided by AllTel in Texas.
My opinion, you're SOL and there is very little you can do.
I keep hoping that someone at digium will pick up
simprix wrote:
How much would a two site location voip pbx cost with a t1 between
the location with a isdn pri coming into the main site. With about 65
users
How much would a system like that cost
2 X PC boxes $350+ each
1 X T1 Digium card $600
65 X User agent devices. This is tricky
Does anyone know if qualify=XXX should be used ONLY for user agents
behind NAT.
I tried to use it on a devices (ATA 186 and SPA 2000) on a public IP,
and * goes to segmentation fault every time it starts.
If it is meant to be used just behind NAT fine, but what and how does *
monitor user agent
I tried to use it on a devices (ATA 186 and SPA 2000) on a public IP,
and * goes to segmentation fault every time it starts.
Does it crash even if you remove Qualify= from sip.conf?
No it does not...
Only when:
Host=dynamic OR host=$PUBLIC IP AND qualify=YES
TO help you we need to get
And yes, there's a config in iax.conf so you can turn it off if you
for some reason want to bother B with staying in the middle of the
call.
Yap. Great stuff :)
Just so everyone knows the config is: notransfer=yes
It would be good to know what happens with cdr records and call control?
Matteo Rancilio wrote:
How can I make an extension that will do the follow:
- Operator A pick up an external call
- Operator A Blind Transfer the call to Internal X
- If Internal X is busy the call will get back to Operator A
Use call forwarding on busy to forward the call back to operator.
Matteo Rancilio wrote:
Senad Jordanovic ha scritto:
Matteo Rancilio wrote:
How can I make an extension that will do the follow:
- Operator A pick up an external call
- Operator A Blind Transfer the call to Internal X
- If Internal X is busy the call will get back to Operator
Hi,
Does anyone have any experience getting Dialogic PRI-ISA48 T96-6028
working with Asterisk? Is software licence required from Dialogic
Etc.
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[EMAIL PROTECTED] wrote:
We're using a 7940 from Europe, connecting to a US Asterisk server,
and it works great. We setup a local Asterisk server in Europe, had
the 7940 connect to it, and used IAX2/GSM to connect to the US. It
is choppy using all CODECS, and I am curious if there
Linus Surguy wrote:
Well why not? Everyone has to eat at the end of the day!
Is it worth considering setting up an asterisk-trading mailing list
specifically for this purpose?
But surely we'd all just end up trying to sell to each other that
way! At least being on the main mailling list
Essentially these are general issues I have with Sipura SPA 2000:
* If SPA 2000 is behind NAT, calls are not hanged up when
receiver is replaced. I think asterisk does not get hung-up
signal from SPA so called party user agent is ringing until
timeout expires or
Paul Cheng wrote:
See
http://bugs.digium.com/bug_view_page.php?bug_id=0001195
This is resovled now...
and also
http://bugs.digium.com/bug_view_page.php?bug_id=0001220
This is related to above and it is solved now...
Ta
SJ
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Asterisk-Users
Miguel Cavazos wrote:
no it wont happend with zap cards or other sipphones such as
grandstream and wisip.
I am referring to noise DSL service produces on the line. It is a very
tiny but it it is there... So.. May be somehow it transfers into your
IP network...
Miguel Cavazos wrote:
if it was related to the dsl line i would notice my other phones such
as grandstream and the ones on zap cards with the same problem im
only having this issue with sipura.
Sure...
When did this start.
I am using sipura devices with no such problem. (I have other
Title: Message
Sure... I will call SIpura..
Thanks
for the info!
Ta
SJ
Bill Reid wrote:
I have had a similar problem upgrading to .24 . Sipura support
suggested
using tftp which worked successfully.
On the tftp server you use the URL
http://aaa.bbb.ccc.ddd/upgrade?/path_name/spa.bin
where aaa.bbb.ccc.ddd is the IP address of the Sipura.
Do not know
Anton wrote:
Better get a hardware expert, We are currently adapting asterisk to a
cpci platform to get around the serious hardware limitations that
digium always stops at.
Anton
Sphyrna Inc
Could you explain what are those serious hardware limitations?
Andres wrote:
Michael Shuler wrote:
When I use reinvites everything works perfectly (so phoneA--phoneB
directly works fine). When I shut off reinvites
(phoneA--asterisk--phoneB) I get the following with PhoneA
initiating the call:
Mar 12 14:43:23 DEBUG[1209277232]: rtp.c:943
Hi,
Anyone knows what needs to be changed in sipura dial plan:
(*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.)
In order to dial *.
Ta
SJ
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exten = 1,AbsoluteTimeout ($SECONDS)
Ta
SJ
Hi,
I saw somewhere that it was possible to set a limit for how long time
a call could be, for an extension in extension.conf. But I can't find
it anymore.
Can someone please help.
Calls to '411' an operator may max. be 5 min.
I have
Hans-Henrik Andresen wrote:
Hi,
I have 3 friends trying to connect to my Asterisk using x-lite, all
of them are using 3 dif. adsl-provider.
For each of them I got this in sip.conf:
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=g729
allow=g723.1
[seholm]
type=friend
Yes, it is. (If I remember correctly :)
It is T that you need to include in that context.
[$CONTEXT]
exten = 1,AbsoluteTimeout($SECONDS)
exten = 2,Dial($SOMETHING)
exten = T,Playback($YOURMESSAGE)
Save $YOURMASSAGE in /var/lib/asterisk/sounds
If above does not work, please let me know.
Ta
SJ
/HHA
Senad Jordanovic [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Yes, it is. (If I remember correctly :)
It is T that you need to include in that context.
[$CONTEXT]
exten = 1,AbsoluteTimeout($SECONDS)
exten = 2,Dial($SOMETHING)
exten = T,Playback($YOURMESSAGE
Anyone knows what this means?
Mar 6 21:15:40 DEBUG[311316]: rtp.c:943 ast_rtp_raw_write: Difference
is 720, ms is 110
Mar 6 21:15:43 DEBUG[311316]: rtp.c:943 ast_rtp_raw_write: Difference
is 13832, ms is 1749
Mar 6 21:15:44 DEBUG[311316]: rtp.c:943 ast_rtp_raw_write: Difference
is 648, ms is
Hi,
While terminating calls to Cisco 5300 the called party hears converstion
all OK.
However, calling party hears periodic short bursts of interferance
and/or lost packets noise.
I can see on CLI this:
Mar 5 14:35:53 DEBUG[458773]: rtp.c:943 ast_rtp_raw_write: Difference
is 3760, ms is 490
Mar
I had (and still have) similar problem. Once SPA 2000 registers with
* it all works well for few minutes. After that all incoming calls
are not answered by SPA 2000. Is that what you mean?
If so, I have temporaraly got SPA 2000 to re-register every 3
minutes. This seems to work at the
Steve Kennedy wrote:
On Mon, Mar 01, 2004 at 08:10:11PM -, Senad Jordanovic wrote:
Steve Kennedy wrote:
On Mon, Mar 01, 2004 at 06:23:08PM -, Senad Jordanovic wrote:
You could port your numbers to a licenced telco... Install SDSL (or
even ADSL if you have a lot of faith in your
Matt Riddell wrote:
We do 4 per adsl with gsm every day.
Who is your ADSL provider?
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Mark Messmore, Technical Support, University Telcom Inc. wrote:
I was just wondering if anyone has had this situation...or one
similar to it.
I've got a Sipura SPA 2000. After hooking it up and configuring it
with my * box, it has worked well. From both lines we are able to
dial out at
Steve Kennedy wrote:
On Mon, Mar 01, 2004 at 06:23:08PM -, Senad Jordanovic wrote:
You could port your numbers to a licenced telco... Install SDSL (or
even ADSL if you have a lot of faith in your current provider) and
get all lines working throught SDSL.. :) You prabably should keep
one
Well...
I am using 0.72 version, but problem persist.
Please let me know if you have same experince once you update CVS.
Ta
SJ
Yes, I have a SPA2000 as well, and noticed this on CVS from 2-3
months ago. I have pulled the newest CVS a week or so ago, but not
tested this scenario since then.
When placing a call from Sipura SPA 2000 to other extensions, for some
reason
dialled extension keeps ringing even though SPA 2000 hangs up the call.
Asterisk does not end that call until it is not answered by dialled
extension.
Anyone has experienced similar problem?
Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 03 February 2004 10:31, Senad Jordanovic wrote:
As I've been unable to get app_transfer to work, could someone
explain how it is supposed to work? Currently I have two Asterisk
boxes. A call comes in via zaptel
Title: Message
If you
are running 0.72 version... then in meetme.conf you need to
have:
conf
= ROOMNO,PASSWRD ie. 100,123
Ta
SJ
Scott Russ wrote:
Does anyone know if/how well Asterisk will run under User Mode Linux?
Will the
ztdummy or zaprtc modules work with it?
Thanks,
Scott
I have few boxes running it with no major problems.
Ztdummy will not work becaause uml does not have real usb support.
Zaprtc could
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 have two cards in one of the servers. If I bind SIP port to public IP,
it all works fine. If I do not bind to specific IP (ie. Bind = 0.0.0.0),
I get segmentation fault while starting *.
Can SIP (and other protocols), bind to more then one IP address?
If yes, what is syntax?
SJ
Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 02 February 2004 16:53, Senad Jordanovic wrote:
As I've been unable to get app_transfer to work, could someone
explain how it is supposed to work? Currently I have two Asterisk
boxes. A call comes in via zaptel
Does anyone know, is there a way to get current status of device
From * using some variable or similar in relation to qualify=XXX
statement.
I am referring to qualify= which qualifies and monitors if device
is reachable.
I need this in order to include it in my dial plan so that incoming
Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
As I've been unable to get app_transfer to work, could someone
explain how it
is supposed to work?
Currently I have two Asterisk boxes. A call comes in via zaptel to
ast1. ast1
dials ast2 using iax2 and gets
Steve Foy wrote:
Still no luck, calls are still dropping off about the same amount as
before.
Any more ideas!?
On Fri, Jan 30, 2004 at 05:14:27PM +, Steve Foy wrote:
Thanks, I'll try that and see how it goes.
Cheers,
Steve
On Fri, Jan 30, 2004 at 11:46:05AM -0500, Bill Hamel
Philipp von Klitzing wrote:
Hi!
I would add:
reinvite=no in addition to canreinvite=no.
It may do the trick.
There is no such parameter as reinvite=. Use canreinvite= only.
Ta
SJ
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Philipp von Klitzing wrote:
Hi!
It's also showing up on the wiki:
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
Where? ;-
Philipp
Interesting...!
Mysteriously... reinvite has EDITED it self in above URL to
canreinvite in space in few hours... :)
Ta
SJ
Hi,
Once the call enters into asterisk and then that call gets transferred,
cdr does not record CLID and SRC field data for the transferred call.
Is this a bug or I am missing something?
Ta
SJ
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Hi,
The X100P ZAP channel seems to provide ringing signal to the calling
party for up to 3 rings and only then the extension gets dialed.
Here is context in question:
[home]
exten = s,2,Dial(SIP/4003,25,tr)
Anyone know what I am doing wrong or is this how it is?
Ta
SJ
HI,
Once the call enters into asterisk and then that call gets transferred,
cdr does not record CLID and SRC field data for the transferred call.
Is this a bug or I am missing something?
Ta
SJ
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Linus Surguy wrote:
IRC channel chatter says that there are some new developments with a
cool presence trick that Mark has come up with for bluetooth devices.
I know a bit about it, but I think the general population here would
like to see some details if they're available.
I don't know if
Ariel Batista wrote:
clipcomm people?
Well, I was/am looking for a device with PSTN FXO backup.
www.dlink.com does one like that, but is way too expensive.
I found information on a D-link DVG-1120S (Sip unit.) It has 2 FXS
ports and on FXO port. This would make a nice small office
Kannaiyan Natesan wrote:
Can anyone give an idea how much does it cost if we want to buy the
Licensed asterisk source code? I hope asterisk has two type of
licenses,
1. GPL
2. I can buy and develop software on my own.
Am I right ?
Kannaiyan
Get in touch with www.digium.com .
zoa wrote:
sure,
Its not impossible to have g729 and scsi only systems, although
several
people with scsi systems have had issues with the g729 installation,
i did not.
That doesnt mean that g729 is rock stable, every now and then the
license
disappears or stops working for some
WipeOut wrote:
Greg Boehnlein wrote:
On Thu, 22 Jan 2004, WipeOut wrote:
This is great to see.. but why RH7.3 (or RH8 for that matter) since
it has already been EOL'ed by RH??
Couple of reasons..
1. It is a stable, known quantity that uses solid components and
closely
WipeOut wrote:
Senad Jordanovic wrote:
WipeOut wrote:
I understand or agree with all of your points..
My biggest problem is that RH has basically dropped me in the poo
by killing off their free version and stopping support for all the
free versions as well.. I have been looking
Kannaiyan Natesan wrote:
Can anyone recommend me a fxo device with SIP or IAX functionality.
I have tried with ,
http://www.clipcomm.co.kr/
They were worster than any device. Device itself costed me $270/-
including shipping but not working.
Kannaiyan
Kannaiyan Natesan wrote:
SJ,
I'm also dealing with Andrew, they were good at telling you
stories but nothing professional with the product.
I registered with fwd and started dialling 14551 my fauvorite
where i get clear voice. It gave me with completely noisy sound,
I tried
David Gomillion wrote:
According to digium's site, Note: Please do not attempt to use the
G.729 code in a SCSI-only system. We are currently working with
VoiceAge to correct this issue. (found at
http://www.digium.com/index.php?menu=asterisk_g729).
Does anyone know what these issues are?
zoa wrote:
This is absolutely not true.
I have 3 (raid) scsi asterisk machines in production.
Joachim.
At 11:32 21/01/2004 -0500, you wrote:
In my view at least one IDE drive must be installed in order for *
g729 license to work.
To simplyfy, here is the matrix (This is how I think
Hi,
I am trying to use the RoutCall application.
Do you guys have any more info on RouteCall info.
In particular what all those fields in the database should be used for?
Ta
SJ
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Hi,
I am trying to use the RoutCall applicaation.
Do you guys have any more info on RouteCall info.
In particular what all those fields in the database should be used for?
Ta
SJ
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Walt Reed wrote:
I'm getting some warnings:
NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support
incomplete. Turn off on client if possible
Asterisk Version: CVS-01/06/04-13:50:26
Cisco ATA 186 version: v3.0.0 atasip (Build 031210A)
Is this something I should be
Brian West wrote:
I sent 10.00 :)
On Sun, 11 Jan 2004, calvis wrote:
I just sent $20.00 to [EMAIL PROTECTED]
I am new to the list so I don't really know what I am donating to,
but
the whole Asterisk program sounds pretty cool and I hope to work
myself to setting up an experimental
WipeOut wrote:
Here is the scenario...
SIP UA's can use either GSM or G.711 ( in that order of preference in
the sip.conf )..
Asterisk Server1 is linked to Asterisk Server2 via IAX2 and also
supports GSM and G.711 ( also in that order of preference)..
1. If a call comes in from the UA
Lion Templin wrote:
Don't know if this has been addressed, but why isn't there a
file_include style directive for extensions.conf?
I find that my extensions.conf grows a lot, and it would be a lot
nicer to have a tree of files rather than one big file to try and
navigate. Also, I've got a
Title: Message
Yes, in most places in the USA local
calls are totally free, no per mincharge.
This is not true
in the US for business lines. Residentiallines have a "free" local
calling area. However, business lines from an incumbent local exchange
carrier like SBC nearlyalways charge rates
Hi all,
A feature I think should be included in 1.0 version is playing
a message to calling and called party while the call is being
transferred.
Something like this:
Calling party (whose call is being transferred)
Please wait, your call is being transferred
Called party (who is transferring
TC wrote:
Yes, it does work behind NAT quite well. (at least it does for me
with Draytek Vigor 2600 router)
Can you clarify one point, do you do any redirection on that router I
know most ppl redirect the SIP TCP control port and a say 10 UDP rtp
ports on the NAT device then on the SIPura web
Hi,
Do the callers in USA dialling from USA Telco lines always have to
prefix the CITY/AREA code with 1 in order
To successfully make a call to other USA destinations?
I have not been to USA (yet) :)
Ta
SJ
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Title: Message
I have been trying to read everything I can find on
Sipura 2000 and Asterisk. I am trying to make the Sipura-2000 act as two
analog lines off my asterisk. I have followed (what I believe) the example
on http://www.voxilla.com/Article39.phtml and I still can't get my Sipura to
Hi, All
Is there a provision for AbsoluteTimeout application to notify
called and calling party of the reason why the call suddenly ended?
This way, the parties will be much better informed, hence they
will/should not think that
their VOIP/telco provider(s) are providing bad service.
Ta
SJ
Olle E. Johansson wrote:
Senad Jordanovic wrote:
Hi, All
Is there a provision for AbsoluteTimeout application to notify
called and calling party of the reason why the call suddenly ended?
This way, the parties will be much better informed, hence they
will/should not think that their VOIP
Philipp von Klitzing wrote:
Hi Matteo,
are you 100% sure this works - will the call drop to the t extension
after AbsoluteTimeout has been reached? I think not...
Just curious,
Philipp
exten = T,1,Playback(yourtimehascomeahahahaha)
Is there a provision for AbsoluteTimeout
TC wrote:
Messagei have few sipura units working with no major problems.
send me your sip and ext files and i will compare it.
I am just curious does the Sipura device work smoothly in NAT
environment, if the SIPura device does the registration from behind
the NAT device hold the connection
WipeOut wrote:
Hi all,
Let me be the first to wish everyone, especially the Digium team, an
awesome year in 2004..
Later..
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Same,
WipeOut wrote:
Senad Jordanovic wrote:
WipeOut wrote:
Hi all,
Let me be the first to wish everyone, especially the Digium team,
an awesome year in 2004..
Later..
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Karl Putland wrote:
On Sat, 2003-12-20 at 03:22, Senad Jordanovic wrote:
Hi,
I am trying to figure out if * can register as a client on a remote
MGCP service. Just like SIP and other protocols Do. Anyone tried
this?
No I don't believe it can. The MGCP implementation in Asterisk
Hi,
I am trying to figure out if * can register as a client on a remote MGCP
service. Just like SIP and other protocols
Do. Anyone tried this?
Ta
SJ
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Mike M. Tkachuk wrote:
Hello,
I'm using satellite link (1024/256) Eutelsat.
With Gnugk and Asterisk. The average roundtrip
to my Gateway (DualTalk) is about 650 ms.
I think that's fine for non business telephony,
just for calling to friends.
Hi, thanks for that.
Could you give me a
Mike M. Tkachuk wrote:
Hi,
I have not incoming phone number to test, but I think I can call
you. If I have termination to your country I'll call you (please give
me your stationary phone, not mobile).
Ok, thanks for that.
(USA)1-212-400-7921
Ta
SJ
David Gomillion wrote:
Senad Jordanovic wrote:
Hi all,
Anyone has experience using * through
128 kbs (or bigger) satelite link?
One word of caution: you may have latency problems. Even at the
speed of light, the information has a LONG way to travel
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