Hello Brian,
It might be helpful for us all if the author could let us know more
about the environment in which this application was built. .
I'm getting all kinds of errors when I try to run it, and I suspect that
either my Postgres or PHP installations are incompatible with yours.
I am not
Brian Capouch wrote:
It might be helpful for us all if the author could let us know more
about the environment in which this application was built. .
I'm getting all kinds of errors when I try to run it, and I suspect that
either my Postgres or PHP installations are incompatible with yours.
On Wednesday 24 March 2004 06:49, Brian Capouch wrote:
It might be helpful for us all if the author could let us know more
about the environment in which this application was built. .
I'm getting all kinds of errors when I try to run it, and I suspect that
either my Postgres or PHP
Dear,
Are you using Redhat9.0, coz I discussed about this yesterday with mike
in Digium, and he said that this is normal with Redhat (due to its SMP
handling problems), further he suggested to shift either to redhat 8 or
debian. You can try debian or gentoo.
I will try gentoo tomorrow and will
hey
ive got call forwarding working using the example below. The first part is
basicaly my dialplan for all users... and does the call forward checking.
first it checks if its going to forward to an external number ( use IAX)
then it checks if it has to forward localy ( use SIP) then it rings
Peer Oliver schmidt wrote:
Brian Capouch wrote:
It might be helpful for us all if the author could let us know more
about the environment in which this application was built. .
I'm getting all kinds of errors when I try to run it, and I suspect
that either my Postgres or PHP installations are
Asterisk works fine across cipe tunnels, quite happily got IAX links
running to my home from work over a cipe link.
You probably won't get ssh port forwarding running because IAX uses udp
and I think ssh only forwards tcp by default.
Date: Tue, 23 Mar 2004 19:53:46 -0600 (CST)
From: [EMAIL
On Tue, Mar 23, 2004 at 07:53:46PM -0600, [EMAIL PROTECTED] wrote:
Hello,
I am interested in knowing if someone has done any work on
IPSec
I've used IPSec on transcontiental links with IAX no problems.
for Asterisk boxes. If so, it will be nice if we can all share our
experiences here.
Wipeout wrote:
Another thing I had to do was changing the defines.php file to reflect
my environment. After that, things went smooth.
On my server the links dont even work in the menu on the left.. Not sure
what is going on with the code and dont have the time to look right
now.. I will just
nat ha scritto:
earlier [EMAIL PROTECTED] wrote...
Is it possible to LookupCIDName from a unixODBC/MSSQL database?
If yes, how?
If Asterisk supports unixodbc or iodbc then you can use FreeTDS
http://www.freetds.org to connect to the MSSQL box.
We use Perl and FreeTDS to produce our
Peer Oliver schmidt wrote:
Wipeout wrote:
Another thing I had to do was changing the defines.php file to
reflect my environment. After that, things went smooth.
On my server the links dont even work in the menu on the left.. Not
sure what is going on with the code and dont have the time to
Does somebody have a script to export Master.csv data to a new
asteriskcdrdb mysql database? Please help
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hi,
I did the following:
1) Obtained asterisk-0.7.2.tar.gz from digium's website.
2) Extracted to /usr/src/asterisk/
3) Attempted the make command.
I obtained the following errors after quite some time in the build process:
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
For some reason, in an otherwise working * installation, I
cant hear any of the sound files. It doesnt seem to matter what
phone make or codec I use. Any help would be appreciated.
JC
Before going to gentoo or another version of linux, try the following:
1) turn hyperthreading off in the BIOS. It's probably called something
like virtual cpus.
2) Use a vanilla kernel from kernel.org e.g. 2.4.25.
We used to have times when the system would just hang. Using a
non-redhat kernel
A minor gripe with our current system (* + CP7960s) is calls answered by
one handset showing as missed on others. RFC3326 seems to answer this
problem but I see no support for it in the cisco phones or * (obviously
less of a problem with the latter being OSS). Are cisco likely to add this?
Do
Maybe I need to tweak the rxgain as I have problems with volume on the x100p
anyway.
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 24, 2004 12:48 AM
Subject: Re: [Asterisk-Users] Softfax problems
Hi Jon,
It seems spandsp
I posted this a week or two ago but no replies, so trying again...
Summary: Two phones in different locations, each behind NAT, can both
talk to an Asterisk server on the net, for the demo or for voicemail,
but can't maintain a call to each other via that asterisk.
Original post with details:
I
Tony Mountifield wrote:
I posted this a week or two ago but no replies, so trying again...
Summary: Two phones in different locations, each behind NAT, can both
talk to an Asterisk server on the net, for the demo or for voicemail,
but can't maintain a call to each other via that asterisk.
Tony,
What is the BW connectivity at the [*] box?
You may try to set the GS phones to GSM codec to reduce BW,
and see if that improves the situation.
WW
- Original Message Follows -
I posted this a week or two ago but no replies, so trying
again...
Summary: Two phones in different
Hi,
I am developing ASTERISK as my SIP Proxy server. So, questions arise when
you begin adding new users, being like this, I would like to know if
ASTERISK
working as SIP PROXY has a limit of REGISTERS and if the
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] wrote:
Tony,
What is the BW connectivity at the [*] box?
It's lots. Much more than the broadband connections our phones are behind.
File downloads to the * box from elsewhere on the internet typically go
at several hundred kbytes/sec.
You may
Title: Message
just
having a post on the list would work...a lot of times searching through list
archives brings a lot of good info...another good place is on the
wiki.
Mark
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Umar
In article [EMAIL PROTECTED],
Umar Sear [EMAIL PROTECTED] wrote:
I recently posted some messages requesting help on how to configure
asterisk as a standalone SIP voicemail service. I was quite disappointed
as I did not get any response at all. However I understand that there is
no obligation
Maybe this helps. I have 4 sipuras on the same network as Asterisk. I had to
make sure each line on the sipura uses a different sip port: 5060/5061 on
the first one, 5062/5063 on the second, and so on.
Best regards,
- Original Message -
From: Matt McIntyre
To: [EMAIL PROTECTED]
Sent:
Try setting 'reinvite=no' in the sip.conf file. This will force Asterisk to stay in
the loop...it otherwise tries to step out of the connection and let the phones talk
directly to each other,which is fine on a LAN but if both are behind NAT firewalls is
asking for complication.
Rgds
Tim
Sorry, didn't read your mail thoroughly - you've already tried canreinvite=no...
Next step is to get an Ethereal log from both ends and investigate what is going on
with the SIP and RTP packets.
Rgds
Tim
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
Very strange ISSUE.
For an errouneus rm -rf command while exprerimenting softmodem, I've removed
my source tree, so I've downloaded a new one and recompiled without softmodem
but with CAPI.
Now I've this problem:
If I phone via CAPI from a SIP phone All goes right.
If I phone via PSTN through a
In article [EMAIL PROTECTED],
Robinson Tim-W10277 [EMAIL PROTECTED] wrote:
Sorry, didn't read your mail thoroughly - you've already tried
canreinvite=no...
Next step is to get an Ethereal log from both ends and investigate what
is going on with the SIP and RTP packets.
Yes, that's the next
You need to turn register globals = on
in your php.ini file, HOWEVER, this can make things VERY insecure if you
have other php scripts that are not thought out right.
On Wed, 24 Mar 2004, Peer Oliver schmidt wrote:
Wipeout wrote:
Another thing I had to do was changing the defines.php
[EMAIL PROTECTED] wrote:
Another topic of interest is securing the box itself. Does a firewall
(hardware outside of the box or a linux based firewall) suffice the need?
Depends what you are protecting against. If you want to assume some services are
exploitable, you could try to break some
Hello All-
I'm receiving *-Users in digest mode. Is there any way to change digest
mode so the messages are text-only. It seems that every message compiled
into digest mode has the text and mime parts (or worse HTML!) all rolled
into one. Multiply that by the number of messages, and digest
On Wed, Mar 24, 2004 at 07:09:43AM -0700, Jason Becker wrote:
[EMAIL PROTECTED] wrote:
Another topic of interest is securing the box itself. Does a firewall
(hardware outside of the box or a linux based firewall) suffice the need?
Depends what you are protecting against. If you want
It kinda looks like they are reselling Vonage service..
Zac
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Swan
Sent: Tuesday, March 23, 2004 11:10 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Nuvio users?
Hi,
Anyone gotten Asterisk to
There are a lot of wholesale VOIP providers now, they are probably just
buying capacity from one of them and either re-branding, or are using a
custom gateway and/or commercial solution.
-Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zac
Hello Everyone,
We are about to launch our International IAX2 worldwide termination
service from any IAX2 softphone. We would like people to make FREE calls
to the USA or Canada so we can check the stability of our platform. We
are allowing everyone to call the USA for free RIGHT NOW! You can
Hello All,
I just finished an other version, all my apologies, cause I made it for
mysql then I ve done the change to support postgresql and forget to
re-test again... not really professional at all ;)
By the way, I made some amelioration in the sql queries and now it works
even with few
On Wed, 2004-03-24 at 08:02, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Robinson Tim-W10277 [EMAIL PROTECTED] wrote:
Sorry, didn't read your mail thoroughly - you've already tried
canreinvite=no...
Next step is to get an Ethereal log from both ends and investigate what
is
Hi all,
I have installed Asterisk and SIP calls are successfull inside our office.
Then I created some extensions for my colleagues in other city. As our
offices are connected trough a dedicated point-to-point line, by now
I'll just create the extensions for the remote people in the Asterisk
Title: RE: [Asterisk-Users] Nuvio users?
Go through the signup process at vonage and
then at nuvio, and tell me.
Then nuvio boxes look like the Motorola units.
I will know for sure tomorrow when I get my Vonage unit.
Vonage also offers the cisco phone.
We will find out soon enough,
Dear everyone,
I am new to the mailing list and the asterisk system. I am looking for
suggestions to start a VoIP test lab and I am seriously interested in
the asterisk solution. After some homework, this is the end to end
system that I come up in mind:
1. Two PCs running asterisk: Intel Celeron,
In article [EMAIL PROTECTED],
Eric Wieling [EMAIL PROTECTED] wrote:
On Wed, 2004-03-24 at 08:02, Tony Mountifield wrote:
In article
[EMAIL PROTECTED],
Robinson Tim-W10277 [EMAIL PROTECTED] wrote:
Sorry, didn't read your mail thoroughly - you've already tried
canreinvite=no...
Does register_globals need to be on to work with this? And if so, any
chance that will be turned off in the (hopefully near) future?
Thanks, Ryan
On Mar 24, 2004, at 9:09 AM, Areski wrote:
I just finished an other version, all my apologies, cause I made it for
mysql then I ve done the change
For those experts in the expressions that can be used in
extensions.conf, I pose the following question:
In my extensions.conf, I have the following...
exten = s,3,Zapateller,nocallerid
exten = s,4,PrivacyManager
exten = s,5,GotoIf($[${CALLERIDNUM} : 777666 ${CALLERIDNAME} :
Privacy
Thomas B. Clark [EMAIL PROTECTED] wrote:
I have tried working around by setting up my own DNS to be authoritative
for provider.com, and providing a SRV record with a proxy in it, but
Asterisk is ignoring it.
There is a sip.conf option to tell Asterisk to do SRV lookups.
Doug
--
Doug Meredith
Ed Rubright [EMAIL PROTECTED] wrote:
I'm thinking about getting the Plantronics DSP-400 headset for use with
Xlite softphone. I currently have a analog headset that does NOT have a
DSP on board, which gives me mediocre call quality and echo when talking to
the PSTN thru my X100P card. I have
Oliver Wilcock [EMAIL PROTECTED] wrote:
A post in 2002 refered to Mike Sandman as a source for inexpensive (cheap)
message waiting indicators. I called Mike but he doesn't know what
Asterisk is (!) and wants to know what type of phone system I have or what
protocol it uses so that he can send
ASTRICON
It's (semi) official. The Digium team will be joining us for tasty Mexican
food and drinks. We are meeting on Wednesday at the front entrance to the
convention center (actually in front of Exhibit Hall C D, by the fountain
plaza) at 6:00 PM.
The current plan is to head to the
Ok I give up!
What do I have to do to my extensions to implement transfer? I've seen
mention of tacking a t or a T on the end of the dial string... So
here's a macro that I use for extensions:
[macro-stdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could
Hello Andrew,
Thanks a lot for the detailed response. It's deffinately informative.
I was wondering if you could discuss the IAX -- Ipsec setup you have?
Do you have a box outside of the Asterisk that takes care of the
business
or you have a PCI card of some kind? If so, did you have to muck
I have seen a number of postings cross this list that mention the
possibility of standards-tracking IAX2 with the IETF (generating an RFC,
etc.). Has that gone anywhere? What would it take to make it happen?
Several of my clients have indicated that they love the firewall/NAT
neutrality of
http://www.cityonevoice.com is reselling vonage... From what Nuvio told
me, they're not.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zac Amsler
Sent: Wednesday, March 24, 2004 10:28 AM
To: Asterisk Users
I get unable to negotiate codec
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Stephen Karrington
Sent: Wednesday, March 24, 2004 10:09 AM
To: Asterisk Users
Subject: [Asterisk-Users] IAX2 International Termination
Hello Everyone,
We are
This was posted to the hylafax-devel list; I presume it
is also relevant to spandsp:
---BeginMessage---
On 2004.03.22 05:16 David Brownlee wrote:
Has anyone tried using Hylafax with libtiff 3.6.1?
On a NetBSD/i386 box libtiff 3.6.0 works flawlessly
but 3.6.1 gives corrupted tif files on
Matthew Marlowe [EMAIL PROTECTED] wrote:
I get unable to negotiate codec
I was able to get ILBC and GSM (the only two I tried) without any drama
at all. I sent in a quick report to the service address listed in the
original article.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
On Wednesday 24 March 2004 19:40, Matthew Marlowe wrote:
I get unable to negotiate codec
GSM and ILBC works.
Dima
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Stephen Karrington
Sent: Wednesday, March 24, 2004 10:09 AM
To: Asterisk
Hi to everyone:
Does someone know if the ATA 182 works OK with asterisk or should I get a
HandyTone 486 instade or an ATA 186 and a FXS to FXO converter
Thanks
Erick
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I forgot to mention that ILBC and GSM are the only two supported at the
moment.
Sincerely,
Stephen Karrington
Dreamtime.net Inc.
http://www.dreamtime.net
http://www.emailblaster.us
Corporate Office
101 California Street, 22nd Floor
San Francisco, CA 94111-5802
Dreamtime is your global choice
Stephen,
Worked good and sounded good from my end, used GSM as codec. Out of
curiosity, how long do you intend to have this up?
Thanks,
Jeremy
-Original Message-
From: Stephen Karrington [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 24, 2004 11:18 AM
To: [EMAIL PROTECTED]
Subject:
Steven Critchfield [EMAIL PROTECTED] wrote:
On Mon, 2004-03-22 at 16:28, Matt Riddell wrote:
I'd have to say that if given the choice, I'd much rather top posting than
having to scroll through ten pages of information I've already read to
arrive at the point.
Of course, bottom posting
Thanks for the feedback. I will leave it up as long as it takes for us
to get some good information and feedback from people.
Sincerely,
Stephen Karrington
Dreamtime.net Inc.
http://www.dreamtime.net
http://www.emailblaster.us
Corporate Office
101 California Street, 22nd Floor
San Francisco,
I made an Update, now don't need register_globals on anymore...
By the way, I fix some bugs, cause it was not possible to choose
criteria and then browse the result page by page... now it's work fine
:)
So, better to make an update of your version
Hello all!
Do you know if Wildcard E100P supports R2/MFC protocol?
Anybody did install with this protocol?
Thanks!
Fernando G. Testa
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.639 / Virus Database: 408 - Release Date:
try to do ps -auxm to list all the threads of the asterisk.
Then connect with gdb to the thread that takes 99% of CPU and find out
what it's doing.
Martin
On Mon, 22 Mar 2004, Bill Hamlin wrote:
Nope same problem. I just started it and did a couple of ps aux's and got
this output:
[EMAIL
I have configured a basic * box which allows external sip calls in. This works
correctly if someone calls say [EMAIL PROTECTED] and thus gets put through to
extension 23. However, I have not been able to figure out or find the documentation on
how to direct sip calls made to just foo.bar to
--On Wednesday, March 24, 2004 11:13 am -0600 Steven Sokol
[EMAIL PROTECTED] wrote:
I have seen a number of postings cross this list that mention the
possibility of standards-tracking IAX2 with the IETF (generating an RFC,
etc.). Has that gone anywhere? What would it take to make it happen?
Hi I'm trying to install but I think I have a problem!!!
Would I be correct in saying if I don't have the jp graph libs, the
links on the form would be followed but nothing would be displayed
Areski wrote:
I made an Update, now don't need register_globals on anymore...
By the way, I fix some
Noobie question...
I have the app installed ok, and set up CDR just now as well, Mysql is also
working as expected, however is there something that needs to be added to
extensions.conf to enable global CDR logging ..
I looked through the Wiki pages and couldn't find any examples..
thanks
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 24 March 2004 04:10 am, Asterisk DEV. Mailing List wrote:
Asterisk works fine across cipe tunnels, quite happily got IAX links
running to my home from work over a cipe link.
You probably won't get ssh port forwarding running because
Back in February I found * pinning the CPU (Slackware 9.1, Feb CVS). I
ran a strace and found that it was looping on this:
-begin-
write(1, \nUse STOP NOW to shutdown Asteri..., 35) = -1 EIO
(Input/output erro
r)
write(1, *CLI , 6) = -1 EIO (Input/output error)
read(0, , 1)
Hi all,
I do not have any experience with Asterisk but I suppose that in principle
it should be possible to use Asterisk as a pure Gateway between SIP and PSTN
for carrier application. But maybe this is not the right equipment e.g. from
reliability or administrative point of view.
From PSTN
Iain Stevenson wrote:
--On Wednesday, March 24, 2004 11:13 am -0600 Steven Sokol
[EMAIL PROTECTED] wrote:
I have seen a number of postings cross this list that mention the
possibility of standards-tracking IAX2 with the IETF (generating an RFC,
etc.). Has that gone anywhere? What would it
Hi there,
I am still trying to make the asterisk SIP proxy server work with my
Grandstream 100 IP phones.
I tried Stephen advice and it did not work. I stil got the 404 error message.
So, rigth now, I am trying the following configuration(sip.conf):
###
;
; SIP
Marko Rakar wrote:
this is now getting interesting;
when I do echo test from my mediatrix unit to asterisk it works
correctly
when I do echo test from my isdn4linux adapter it also works correctly
when I connect two mediatrix units through asterisk they work correctly
when I connect my
Hi,
I been playing with RxFax and it worked fine for me with last
CVS+spandsp-0.1f+tiff-3.5.7.
I received a FAX and I displayed it with eog and it seems that the
aspect ratio of the image is different, it seems that the received image
is stretched (on the Y-axis). The image
Can I just say that the Mexicali Bullet is a really Amazing drink.
cameron.
Steven Sokol wrote:
ASTRICON
It's (semi) official. The Digium team will be joining us for tasty Mexican
food and drinks. We are meeting on Wednesday at the front entrance to the
convention center (actually in front
While cruising Ebay looking the the holy grail in cheap FXO adapters
and stumbled upon Immixtel (www.immixtel.com), who seem to be reselling
some Korean VOIP products. They have 2,4 and 6 port FXOs, FXO/FXS
combos and FXS adapters. I called them and talked to an engineer named
Jorge who had just
I have been experiencing the same problem with my 4-port Digium FXS
card. I see Ouch, reset... messages on the console. One of the lines,
maybe 2, will either lose dial tone completely, or have tons of static
on the line. On the latest event, today, I noted that even when the
dialtone was gone,
Just for a heads-up, if you're trying to play some Music on Hold, and it
seems like it has strange undertones and echoy lag, I found a solution
(at least it fixed my problem).
I was encoding at 8 kHz. When I resampled to 16 kHz, the problem went
away. Sadly, it took me the better part of a day
Michael Graves wrote:
While cruising Ebay looking the the holy grail in cheap FXO adapters
and stumbled upon Immixtel (www.immixtel.com), who seem to be
reselling some Korean VOIP products. They have 2,4 and 6 port FXOs,
FXO/FXS combos and FXS adapters. I called them and talked to an
engineer
Olle E. Johansson wrote:
An informational RFC documenting the protocol would be a good start,
it would
make it more open but not an IETF product. Security specialists would
get something
to read and analyze. A VOIP protocol with RSA authentication,
implemented today.
Is there any IAX2
I am using the wildcard X100P with *. PSTN line comes in to the FXO port of this card. Everything works fine most of the time. However, occasionally Asterisk doesn't seem to be able to detect the user has hung up and therefore tie up the line for quite a long time. Does anyone know if there's
Greetings,
I have our system setup so that when I am not available my message gives
you the option to either press 1 for voicemail or 2 to forward to my
cell. The problem is the long pause after the choice has been made and
before the vm-intro starts playing. The only thing that comes up in the
I think this has been discussed a lot in the last 3 days - do some legwork
before posting!
Iain
--On Wednesday, March 24, 2004 3:53 pm -0800 Ron McMillin
[EMAIL PROTECTED] wrote:
I am using the wildcard X100P with *. PSTN line comes in to the FXO port
of this card. Everything works fine most
Ariel Batista wrote:
Michael Graves wrote:
While cruising Ebay looking the the holy grail in cheap FXO adapters
and stumbled upon Immixtel (www.immixtel.com), who seem to be
reselling some Korean VOIP products. They have 2,4 and 6 port FXOs,
FXO/FXS combos and FXS adapters. I called them and
quote who=Adam Hart
I also like to see two
people behind the same nat being able to communicate directly (without
requiring pin-wheeling). Ie The client attaches their private ip to the
register packet, which is used when client A B's public ips match.
192.168.1.0/24 -- NAT-BOX -- Internet
On Wed, 24 Mar 2004, Olle E. Johansson wrote:
An informational RFC documenting the protocol would be a good start, it would
make it more open but not an IETF product. Security specialists would get something
to read and analyze. A VOIP protocol with RSA authentication, implemented today.
On Wednesday 24 March 2004 17:54, Dustin Knuttgen wrote:
Greetings,
I have our system setup so that when I am not available my message
gives you the option to either press 1 for voicemail or 2 to
forward to my cell. The problem is the long pause after the choice
has been made and before the
On Wednesday 24 March 2004 16:40, Steve Murphy wrote:
I have been experiencing the same problem with my 4-port Digium FXS
card. I see Ouch, reset... messages on the console. One of the
lines, maybe 2, will either lose dial tone completely, or have tons
of static on the line. On the latest
On Wed, 24 Mar 2004 17:57:06 -0500, Ariel Batista wrote:
Michael Graves wrote:
While cruising Ebay looking the the holy grail in cheap FXO adapters
and stumbled upon Immixtel (www.immixtel.com), who seem to be
reselling some Korean VOIP products. They have 2,4 and 6 port FXOs,
FXO/FXS combos
Robert Hajime Lanning wrote:
quote who=Adam Hart
I also like to see two
people behind the same nat being able to communicate directly (without
requiring pin-wheeling). Ie The client attaches their private ip to the
register packet, which is used when client A B's public ips match.
On Wed, Mar 24, 2004 at 08:54:44AM -0800, Asterisk wrote:
Hello Andrew,
Thanks a lot for the detailed response. It's deffinately informative.
I was wondering if you could discuss the IAX -- Ipsec setup you have?
Do you have a box outside of the Asterisk that takes care of the
business
or
Tested from Bulgaria.
The quality is great, even that the ping from here is 170ms.
Some troubles with dtmf sending.
Stephen Karrington wrote:
Hello Everyone,
We are about to launch our International IAX2 worldwide termination
service from any IAX2 softphone. We would like people to make FREE
Thanks for the feedback. What kind of phone are you using?
Sincerely,
Stephen Karrington
Dreamtime.net Inc.
http://www.dreamtime.net
http://www.emailblaster.us
Corporate Office
101 California Street, 22nd Floor
San Francisco, CA 94111-5802
Voice - 877-203-9308
Fax - 310-943-2606
Dreamtime is
I also experienced the DTMF problem. I wasn't able to use the Qwest
Voicemail system.
I'm on the west code of the US.
I was using my Siemens cordless phone attached to my Digium TDM400P card.
Thanks,
Ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On Thu, 25 Mar 2004, Anton Tinchev wrote:
Some troubles with dtmf sending.
I tested here (I'm preparing a report to send to support at diamondcard
dot us) and I found that they only support dtmfmode=info. Before I was
using dtmfmode=rfc2833. Using a Cisco 7960G phone. I don't know if this
quote who=Adam Hart
from my post: which is used when client A B's public ips match.
meaning in this situation both clients would have different public IPs
and it wouldn't be used.
Do'h!! My bad.
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-MCP
___
Asterisk-Users
No guarantee then when public IPs match that clients are both on same NAT LAN.
Client A 192.168.0.1 - NAT Router A - NAT Router X with Public IP
123.123.123.123 ---
Internet
Client B 192.168.0.1 - NAT Router B -|
Jim
James H. Thompson
[EMAIL PROTECTED]
-
James H. Thompson wrote:
No guarantee then when public IPs match that clients are both on same NAT LAN.
Client A 192.168.0.1 - NAT Router A - NAT Router X with Public IP 123.123.123.123 ---
Internet
Client B 192.168.0.1 - NAT Router B -|
Jim
James H. Thompson
[EMAIL
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 on this
and look at the
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