[EMAIL PROTECTED] wrote:
Hello everyone,
Does anyone out there have actual experience with running * on a
mini-itx board from VIA?
yap... work just fine... we have several locations installed
They look good, but I have some reserves
because of VIA's problems with PCI latency in
[EMAIL PROTECTED] wrote:
Strange;
It works for me. The x100p (Digium 100 buck model) I have is slotted
for
3.3v and works fine.
I'm running gentoo with udev and the 2.6.11 kernel with soekris
patches (udev is cool, coldplug automagically loads the drivers).
The 4801 is flashed with
[EMAIL PROTECTED] wrote:
Everything's for sale . if the price is right . STARTING BID?
:-)
u start... :)
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Davin O'Neill wrote:
I have Asterisk running on a Linux 2.4.x box with ztdummy. Once I
did a modprobe on ztdummy I was able to enter into a conference room
using my softphone clients. I'm using SJphone and Firefly. I have
noticed a significant delay (1 to 3 seconds) while talking within the
[EMAIL PROTECTED] wrote:
Guys'n'Gals
vote for bug 3836 - Chanspy is back. Better than ever. Let's get this
one into CVS.
Julian
And where do you see it as back ?
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When I dial *78 I get a 404 error on the phone ( Call failed). Nothing
shows in the Asterisk console.
You need to check dial plan in the 480i...
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Clive Carter wrote:
Ben wrote
Is there a UK Asterisk users group? Would be interested in
contacting others in the UK who use asterisk for either home or
business applications.
I am in too...
Regards,
Senad Jordanovic
Bicom Systems
www.bicomsystems.com
.
I would not use mpg123... It is just taking too much resources and
sometimes
it crashes the server etc.
We use this with no problems:
http://bugs.digium.com/bug_view_page.php?bug_id=0002379
Regards,
Senad Jordanovic
Bicom Systems
www.bicomsystems.com
David wrote:
Hello,
Can anybody help me with this issue?
-- Called 999302
-- Got SIP response 488 Not Acceptable Here back from
202.125.154.12 == No one is available to answer at this time
Check codecs on both sides
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additional info would be most helpful?
Check the load on your server(s).
Regards,
Senad Jordanovic
Bicom Systems
www.bicomsystems.com
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[EMAIL PROTECTED] wrote:
Kevin P. Fleming wrote:
I have a patch in my local system that allows the canreinvite setting
(which I renamed) to actually be based on IP address masking, so that
Asterisk can make a more intelligent decision, but even that has
problems, because we don't actually
I know AGI is a method for languages like PHP to interact with
asterisk, but can it do this kind of stuff?
Should be quite straight forward to implement your requirement.
An AGI script using your favourite programming language is a way to go.
We have implement all bits you mentioned so
Kanuri, Seshu (Company IT) wrote:
Never used pbxware, but the context the sip phones dial out using
specified in sip.conf needs to include the dialplan context of
the phones in extensions.conf.
As mentioned above check the context for phones... :)
PBXware does not use the Conf files, but
[EMAIL PROTECTED] wrote:
You need to call PBXware it should not have anything to do with the
phone. If a phone registers there gui should put it in the correct
context. If it does not then get your money back due to there non
standard setup
Ariel.. for your information Seshu was told BEFORE
[EMAIL PROTECTED] wrote:
Also, I'm sure you've probably checked on this one,
but are the phones registered with asterisk?
You can make outbound calls on them without them
actually being registered. I'm assuming you can
still get in and see the CLI. What does sip show peers
look like? What
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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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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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
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
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 .
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
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,
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
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
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
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
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.
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
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
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
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?
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.
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
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
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
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
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
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
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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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
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?
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
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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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
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
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?
Title: Message
Hi,
I know
of a person who would probably write the docs for FREE.
He is
British with a superb understanding of English language.
Anyone
in digium interested, please let me know!
Senad
Jordanovic
-Original Message-From: Damian Flynn
[mailto:[EMAIL PROTECTED
, 28 Jul 2003, Senad Jordanovic wrote:
MessageHi,
I know of a person who would probably write the docs for FREE.
He is British with a superb understanding of English language.
Anyone in digium interested, please let me know!
Senad Jordanovic
-Original Message-
From: Damian Flynn
Hi,
Has anyone used http://www.pcphoneline.com/ products with asterisk?
Senad J
attachment: winmail.dat
James,
Thanks for that!
Do you have a list of those US AIX providers?
Also, is there anyone in UK providing
termination/origination on AIX?
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of James Sharp
Sent: 04 August 2003 18:59
To: [EMAIL PROTECTED]
Hi,
X-Lite logs into * with no problems. I dial 1000 and * plays greeting, but
i can not hear it.
Tried many times with the same result.
After quite few tries * complains about:
-
WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt): Maximum
the allow=all in sip.conf to allow=alaw and see if
that works.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Senad Jordanovic
Sent: Friday, August 08, 2003 1:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] X-Lite - No sound
+ chan_sip issue
Change the allow=all in sip.conf to allow=alaw and see if that works.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Senad Jordanovic
Sent: Friday, August 08, 2003 1:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] X-Lite
I agree as well... phpbb is much better solution.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of James Taylor
Sent: 09 August 2003 15:08
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] list proposal
I'd still vote for phpbb.
Then we could have
have
you looked at digiums site? there are few simple sample
there.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Adelino
BaenaSent: 09 August 2003 21:47To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] H323 and
SIP
Dear
:[EMAIL PROTECTED] On Behalf Of
Senad Jordanovic
Sent: Friday, August 08, 2003 1:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] X-Lite - No sound + chan_sip issue
Hi,
X-Lite logs into * with no problems. I dial 1000 and *
plays greeting, but
i can not hear it.
Tried many
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
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! :)
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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
pesb wrote:
Hi there,
I have the following scenario:
I want user A to make a call to user B. If, any of these users
transfer the
call to user C, then asterisk should generate a CDR for the first leg
of the
call. And after the transfered call is finished, asterisk should
Holger Schurig wrote:
Would IAX2 support having signalling going through
all machines and voice data through the shortest path
No, Signalling+Voice is tightly coupled.
To my knowledge, IAX2 will take shortest route possible. I.e.
A call from UA A to UA C through server B will switch from
Jeremy McNamara wrote:
Chris Bond wrote:
I think one thing * is lacking at the moment is a web interface to
manage and add users and do anything you can do via a shell
interface. If it had that but on a simplified level (oblessly you
can have an advanced mode too).
The power of
Florian Overkamp wrote:
Hi,
-Original Message-
No, Signalling+Voice is tightly coupled.
To my knowledge, IAX2 will take shortest route possible. I.e.
A call from UA A to UA C through server B will switch from
original path
(ABC) to (AC) and this is default behaviour unless
Maybe, maybe not... Depending how one designs the GUI!
No, I think that GUIs though needed, do limit flexibility because the
information density is limited on the user-system direction (they are
better on the System-user end, however). However, this is NOT an
argument not to package
Sure... So, this issue is sort of a bug and it really needs to be
implemented then!
I'm afraid its not that simple. Unless I'm misunderstanding the
concepts of
IAX(2) design, it does not support such behaviour _by design_. Who
knows what would break if someone hacked our desires in
Chris Bond wrote:
Yes, you are right!!
However, GUI for newbie's will help some people to overcome the first
hurdles, and then plunge into more advanced stuff!
One thing quote a lot of companies do is outsource the initial
configuration, because they simply don't have the technical skills
Hi, all
This is certainly very good news!
http://www.neowin.net/comments.php?id=21119category=main
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Hi,
Anyone knows where can one get ptag.dat for sip.
I do not fancy waiting weeks to get it from Cisco!
Ta
SJ
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brian wrote:
That's the only way to make it work.
Forgive me and for clarification to others as well!
Devices behind nat, on same network, can call each other ONLY if
canreinvite is set to no? Is that what you are saying?
If so, why?
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James Jones wrote:
Do you just need a default config file for sip and which Cisco device?
Sure, I have default config file for Cisco ATA 186.
However, the task is to create config files for each device and place
them in TPTP root directory and apparently one needs cfgmfg and pdat
files in order
Randy Bush wrote:
sip debug shows that my * is trying to invite my spa and
being told 404
Reliably Transmitting:
OPTIONS sip:42.7.11.194 SIP/2.0
Via: SIP/2.0/UDP 128.9.0.39:5060;branch=z9hG4bK43efe1d7
From: asterisk sip:[EMAIL PROTECTED];tag=as39d40d19
To:
Dear Sirs,
I've got a weird problem with IAX2 transfers.
My setup consist of 3 Asterisk servers. One is located in Europe on a
public IP and a local PSTN connection through ISDN. Two are located in
South-east Asia - both on public, but dynamic IP. These two each have a
bunch of SIP phones
Hi,
Checking while back, it was possible for two or more UA to use same
login info to place
calls and use other services at the same time!
Does anyone know if there were any development done in order to prevent
this?
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