No I use chan_capi and H323 but not zaptel device.
So can I use it ?
When I lauch ztdummy I have some errors.
Regards
Rattana
- Message d'origine -
De : Brancaleoni Matteo [EMAIL PROTECTED]
À : [EMAIL PROTECTED]
Envoyé : mardi 4 mars 2003 18:32
Objet : Re: [Asterisk-Users] a problem with
Hi John,
That is correct. Note those are VCXO crystals, and not ordinary ones.
They need to have the right pullable characteristics to get reliable
results. If you downlod the info from Dallas for the framer chip they
give some recommendations for off the shelf crystals which will work.
Regards,
Hello all,
This new version of asterisk-oh323 has some bug fixes
(in configuration file parsing), changes in configuration file
parameters (an attempt to make it more readable!) an some minor
improvements like in-band DTMF detection (based on code
posted by Klaus-Peter Junghanns on this list) and
The only problem I can think that you would have with the
ztdummy would be that to used a kernel source other
then the one your running when you build it...
So what errors did you get when you build ztdummy?
Rattana BIV wrote:
No I use chan_capi and H323 but not zaptel device.
So can I use it ?
Storm3 Communications ([EMAIL PROTECTED]) wrote*:
Right, I was reading about them a few months back.
I was really excited there for a second.. 'WOW! three provinces away, ..I'm
famous!
So, how are you finding Asterisk? I may be building a toll bypass circuit
with it in a few weeks. While I'm
Hi,
I wanted to know in which code source the file sample.call is proceeded when
we put it in /var/spool/asterisk/outgoing/
I try to make an application to asterisk who check when an user in H323
(netmeeting) is connect or not.
Regards
Rattana
___
it is errors I have when I lauch modprobe ztdummy
/lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol
proc_mkdir_Rsmp_220b03b4
/lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol
create_proc_entry_Rsmp_3a9bfbd2
/lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol
Hi Michael,
I want to register * using oh323 to my GW. But it need to send me username
and password as H.235.
Can you tell me whether there is a support for this thing ?
Hemant
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 05,
pbx/pbx_spool.c
On Wed, 5 Mar 2003, Rattana BIV wrote:
Hi,
I wanted to know in which code source the file sample.call is proceeded when
we put it in /var/spool/asterisk/outgoing/
I try to make an application to asterisk who check when an user in H323
(netmeeting) is connect or not.
On Wed, 2003-03-05 at 00:18, Storm3 Communications wrote:
Right, I was reading about them a few months back.
I was really excited there for a second.. 'WOW! three provinces away, ..I'm
famous!
So, how are you finding Asterisk? I may be building a toll bypass circuit
with it in a few weeks.
Hemant Kumar wrote:
Hi Michael,
I want to register * using oh323 to my GW. But it need to send me username
and password as H.235.
Can you tell me whether there is a support for this thing ?
No, there is no support for H.235 password but I
will fix it and send an newer version.
Michael.
Hemant
Just a note to the list, I tried to apply that patch to the current CVS
but it failed. This was expected of course because the code has changed
since the patch was released.
Assuming this patch is stable it should defiantly be Incorporated into
the main code.
John Lange
On Tue, 2003-03-04 at
Would it just be easier to use a database system that managed concurent
access better? Most all of them don't have trouble with this and
don't require work-arounds like hot copies and so on.
--- [EMAIL PROTECTED] wrote:
I am always wary of allowing ad-hoc queries that may lock tables on a
Certainly, Oracle does a bangup job there and it's almost as affordable as MySQL.
Chris Albertson wrote:
Would it just be easier to use a database system that managed concurent
access better? Most all of them don't have trouble with this and
don't require work-arounds like hot copies and so on.
On Wed, 5 Mar 2003 [EMAIL PROTECTED] wrote:
Certainly, Oracle does a bangup job there and it's almost as
affordable as MySQL.
If it means the difference between getting paid and not getting paid, it's
considerably less expensive than mysql.
dave
--
Dave Weis I believe there are
On Wed, 2003-03-05 at 08:33, Steven Critchfield wrote:
What I learned from home, you do not want X running at all on your phone
system. The VoIP quality on my home system took a severe drop when my
screensaver would activate. It would also clear as soon as the screen
saver would go away. This
On Wed, 2003-03-05 at 12:36, Karl Putland wrote:
On Wed, 2003-03-05 at 08:33, Steven Critchfield wrote:
What I learned from home, you do not want X running at all on your phone
system. The VoIP quality on my home system took a severe drop when my
screensaver would activate. It would also
Oracle does do a good job. but there are others that
cost less, However if you really do need Ocacle I'd argue
it's worth the price. There is also IBM's DB2
If you need free then:
PostgreSQL is well known and is not unlike Oracle. PostgreSQL traces
it's roots back to the first ever
I don't know (haven't tried myself) but Kostya V. Ivanov's 'R' patch to the
dial application (december 2002) might be of help for you. Check the
archives for Barge (Intrusion) Capabilities. It might be some manual work to
apply after all the allmost daily CVS changes but worth a try!
Michiel
--- Tilghman Lesher [EMAIL PROTECTED] wrote:
Actually, I think you'll find that it's easier to manage and plan
for database usage than it is to recover from unexpected database
load. It really doesn't matter what database you're using; even
the most heavy-duty mainframe will underperform if
I have recently begun experimenting with Asterisk, and have been
mightily impressed by its capabilities and flexibility. I have run
across one problem, however, that challenges my ability to use it as a
production system.
My Asterisk box has a public Internet IP, and works great with
On 2003-03-05 at 15:08, you wrote:
I would like to put these SIP phones into are behind NAT.
I was quite surprised when I tested Vonage's service that I could plug
in their ATA 186 behind my NAT/VPN box and it immediately worked, even
for incoming calls.
They must make an outbound connection
Finally someone has hit the same problems that we have. Everyone on this
newsgroup seems to have static IPs!
The problems you get can manifest in 2 ways:
1) you cannot get through to the phone at all
2) one-way audio - you can hear the other end but they can't hear you.
The problem is a
When situations like this arise all the time, why is there such a delay in
getting ipv6 rolled out when it solves all these problems ?
I realize no one acting alone can roll it out in any sort of meaningful
way, but the solution has already been around over 10 years, just not
getting used.
At
Wade Weppler wrote: I've just been getting into this as well, and I've
run into the same
problem.
It looks like the only solutions so far are to use a SIP client that
supports STUN (the SNOM 100 supports this), or that supports UPnP (with a
NAT router that supports UPnP). For Linux NAT routers,
I also did a change to asterisk to be able to record a channel irrespective
of the application it is currently in. You can start/stop recording by
calling an application from the dialplan or sending a message using manager.
I'll be submitting a patch to Mark shortly.
-Original Message-
According to Jon Pounder:
When situations like this arise all the time, why is there such a delay in
getting ipv6 rolled out when it solves all these problems ?
How does IPv6 solve address translation problems? If you mean to suggest
that more addresses would eliminate the need for NAT,
Greetings,
There seems to be a bit of buzz on the list about
ADSI phones and their configuration, but no clear progression of what really
needs to exist to have a basic config. Could someone please post what they
had to do to get an unlocked ADSI phone to work?
Thanks
Raymond McKay
http://lists.digium.com/pipermail/asterisk-users/2003-March/008088.html
why is there such a delay in getting ipv6 rolled out when it solves all these
problems ?
===
There are many reasons...
1. Leasing Address Space from the I* society (small s...aka the Big Lie Society) is
not desirable by all
Matthew Farley wrote:
I have recently begun experimenting with Asterisk, and have been
mightily impressed by its capabilities and flexibility. I have run
across one problem, however, that challenges my ability to use it as a
production system.
My Asterisk box has a public Internet IP, and
At 05:45 PM 3/5/2003 -0600, you wrote:
http://lists.digium.com/pipermail/asterisk-users/2003-March/008088.html
why is there such a delay in getting ipv6 rolled out when it solves all
these problems ?
===
There are many reasons...
1. Leasing Address Space from the I* society (small s...aka the Big
At 06:08 PM 3/5/2003 -0600, you wrote:
On Wed, Mar 05, 2003 at 05:45:13PM -0600, Jim Fleming wrote:
http://lists.digium.com/pipermail/asterisk-users/2003-March/008088.html
why is there such a delay in getting ipv6 rolled out when it solves
all these problems ?
I doubt that users will stop
Finally someone has hit the same problems that we have. Everyone on this
newsgroup seems to have static IPs!
The problems you get can manifest in 2 ways:
1) you cannot get through to the phone at all
2) one-way audio - you can hear the other end but they can't hear you.
The problem is a
- Original Message -
From: Jon Pounder [EMAIL PROTECTED]
...Asterisk is a natural extension of the NAT transition/evolution and
helps to negate any need for IPv6...
I think all the extra layered on crap for NAT and the hacks to make a few
static ips last longer in a world that
We just added support for the ;received= method of NAT translation.
This works only if your NAT does not translate the port number. Otherwise
your call will come up w/out RTP.
Mark
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Find me on IRC and I'll try to (yet again) work on this.
Mark
On Wed, 5 Mar 2003, Wade Weppler wrote:
There is now (thanks, Mark!) an addition in sip.conf called nat=1
that can flag a sip user/peer/friend as being behind a NAT address
translator. The good news is that the REGISTER and
I believe I found my problem - an incorrect IP address for a network
time server in the ATA-186 configuration. I entered a valid network
time server address in the NTPIP field, and the error is gone.
Art
On Sunday, March 2, 2003, at 11:33 AM, Art O'Dea wrote:
I have zaptel and zapata compiled
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 05, 2003 8:33 AM
Subject: Re: [Asterisk-Users] OT: PRI costs in US
On Wed, 2003-03-05 at 00:18, Storm3 Communications wrote:
Right, I was reading about them a few months back.
http://lists.digium.com/pipermail/asterisk-users/2003-March/008088.html
Jon Pounder [EMAIL PROTECTED]
Wed, 05 Mar 2003 17:28:20 -0500
why is there such a delay in getting ipv6 rolled out when it solves all these
problems ?
I doubt that users will stop using NAT until ISPs stop charging
Maybe I should quit complaining already! The average business customer
pays somewhere in the neighborhood of $30 base per plain (non-Centrex,
etc.) analog line here. ISDN is $60 if you don't sign up for 3 years, or
about the same price.
Consider yourself fortunate. For those of us stuck in
On Wed, 2003-03-05 at 16:34, John Todd wrote:
I know a solution exists here. My ATA-186 works behind my NAT when I
have it configured for iconnecthere.com, and they don't have magic
UDP elves, so it must be able to work for other SIP servers if the
right trickery can be implemented. I
Jon Pounder ([EMAIL PROTECTED]) wrote:
At 06:08 PM 3/5/2003 -0600, you wrote:
On Wed, Mar 05, 2003 at 05:45:13PM -0600, Jim Fleming wrote:
http://lists.digium.com/pipermail/asterisk-users/2003-March/008088.html
why is there such a delay in getting ipv6 rolled out when it solves
all these
At 08:30 PM 3/5/2003 -0700, you wrote:
Jon Pounder ([EMAIL PROTECTED]) wrote:
At 06:08 PM 3/5/2003 -0600, you wrote:
On Wed, Mar 05, 2003 at 05:45:13PM -0600, Jim Fleming wrote:
http://lists.digium.com/pipermail/asterisk-users/2003-March/008088.html
why is there such a delay in getting ipv6
Hello,
I have my sip stuff seemingly working fine as well as my zaptel stuff
working great... But I have a problem with sip registration timers (I'm
guessing here).
In my extensions.conf file I have this...
exten = 2244,1,Dial,Zap/2|25
exten = 2244,2,Dial,Sip/brian|25
exten =
Hi,Steve,
My ISA E1 card does not work yet. I would like to know if there are other
difference bettween then two kind of ISA card?
john
Steve Underwood writes:
Hi John,
That is correct. Note those are VCXO crystals, and not ordinary ones.
They need to have the right
qualify=1000 in sip.conf in the phone config entry
regards
Martin
On Wed, 5 Mar 2003, Mark Spencer wrote:
But if I close my sip phone and a call goes through it will still wait
the 25 seconds before it goes to voice mail even though my Sip phone is
not even on. If I restart Asterisk and
Symptoms: when calling my iconnect phone number (13033913323 in my
bogus example below) from my cell phone, I can see that the call
makes it to my asterisk server, and my phones even ring once as *
passes the call through during the 180 Ringing period. However, it
seems that iconnecthere.com
On Mon, 03 Mar 2003 around 17:16:56 -0500, Jon Pounder wrote:
3) I am under the impression that with isdn4linux you lose a lot of the d
channel information like called number, bulk callerid, etc which are
things the CAPI project seems to be working towards
Is this actually the case or am I
On Mon, 24 Feb 2003 around 08:09:04 -0600, Mark Spencer wrote:
Caller*ID on a PRI has various levels of trust associated with it. It
could be some providers don't trust the callerid.
Here in Holland you can only use the numbers assigned to the line
by the telco (BRI or PRI). If you use any
John,
A heads-up: iconnect has apparently put up a filter against my IP address,
for whichever reason (apparently they don't like people using asterisk?).
I've sent them an email and am pursuing this also through sales side (I'm
about to make a resale deal with them), so hopefully tomorrow
John Todd wrote:
Symptoms: when calling my iconnect phone number (13033913323 in my bogus
example below) from my cell phone, I can see that the call makes it to
my asterisk server, and my phones even ring once as * passes the call
through during the 180 Ringing period. However, it seems that
51 matches
Mail list logo
