Good day!
Have a weird problem with HT-286 and Conference room. I use Asterisk
CVS-HEAD-06/04/04.
Here it is:
When HT-286 get into the conference room first and nobody in that room
everything seems ok (with any codec HT286 allowed), but when HT-286 get
into conference room when somebody already
On Fri, 2004-07-02 at 09:07 +0300, Vladyslav wrote:
Good day!
Have a weird problem with HT-286 and Conference room. I use Asterisk
CVS-HEAD-06/04/04.
It's not weird at all.
Here it is:
When HT-286 get into the conference room first and nobody in that room
everything seems ok (with any
On Fri, 2004-07-02 at 09:22, Dave Cotton wrote:
2. When HT use G729 codec = it gets busy signal and I could see such
output on asterisk console (
Jul 1 07:26:14 WARNING[737298]: chan_sip.c:1611 sip_write: Asked to
transmit frame type 4, while native formats is 256 (read/write = 4/256)
For example:
http://sdcc.sourceforge.net/
SDCC is a Freeware, retargettable, optimizing ANSI - C compiler that
targets the Intel 8051, Maxim 80DS390 and the Zilog Z80 based MCUs.
Work is in progress on supporting the Motorola 68HC08 as well as
Microchip PIC16 and PIC18 series. The entire
I have a Sipura 2000 working fine, but whenever
I dial any extension there is a delay of 5-10 seconds before
it starts ringing. However, if I dial the extension and hit
the pound key after the number, it goes through right away.
Is there any way around this?
On Fri, 02 Jul 2004 17:33:30 +1000
Master Abi [EMAIL PROTECTED] wrote:
New firmware version at http://www.hellofone.com/downloads.html. Might
fix the no register issue and others.
I tried this a few days back, totally hosed my phone, has to back to Grandstream,
don't touch it!
-Max.
--
On Fri, 02 Jul 2004 17:33:30 +1000
Master Abi [EMAIL PROTECTED] wrote:
New firmware version at http://www.hellofone.com/downloads.html. Might
fix the no register issue and others.
I tried this a few days back, totally hosed my phone, has to back to
Grandstream, don't touch it!
-Max.
I just
We can give you 800 and local access numbers in US and Canada VoIP (SIP and
H.323). All you need is a good internet connection, or you can collocate
with us if you like.
Aram Ter-Martirosyan
Senior Account Manager
Hi-Tech Gateway, Inc.
http://www.hi-teck.com
1225 Grand Central Ave.
Shaun Dawson [EMAIL PROTECTED]
However, if I try to dial out, I connect, but get
absolutely nothing. The cell phone doesn't ring, and
no audio.
If I try to dial in, I get a phone system recording
that says due to phone system trouble, this call
cannot be completed.
This is the type of
Leif Madsen [EMAIL PROTECTED] wrote:
Also, from what I have been told (and I've tested this by building
zaptel, but not any of the other sources) is that you no longer need
the sourcecode with the 2.6 kernel. You can create a symlink to:
/lib/modules/`uname -r`/build/
Instead of the
Hi all,
I would like to ask if Asterisk will allow to be monitor via web browser. I
am planning to create a web interface to monitor the current sip connected
end points and status of iax channels use.
If i write a code in php to execute this command should it be possible?
asterisk -rx iax show
I'm not a really good C Programmer,
butmanage to get the meetme integrated with mysql. Please visit http://www.mithotech.com/asterisk
.
Some of the functions are :-
- Make room reservations- Authenticate with
MySQL base on room number, pass, maximum allowed users and datetime- With
Does anyone know the status of Australian approval
for the E100P?
Cheers,
Clint.
Hi there,
is there anyone that has a clue as to the effect of TRUE or FALSE for a
swissoice ip10 config.cfg file:
set xgcp PackageNotify TRUE
Cheers, Philipp
___
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[EMAIL PROTECTED]
hi all,
i am getting crazy with my TE410P - it won't work
now we already think that the only thing which can be wrong is the
cable.
At our telecom endpoint we have 1-2 tx, 4-5 rx
So - which cable do i need to get it working
Already tried a cross over cable 1-2 - 4-5 / 4-5 - 1-2 (with this
cable
Group
Following the information located on
http://www.asterisk.org/index.php?menu=download
I get the following error installing the zaptel
Any help would be great!!!
Thanks
[EMAIL PROTECTED] zaptel]# make clean; make install
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg
Hi Glynn,
Just take a look on http://www.voip-info.org/wiki-Asterisk+monitoring
Glynn Condez wrote:
Hi all,
I would like to ask if Asterisk will allow to be monitor via web browser. I
am planning to create a web interface to monitor the current sip connected
end points and status of iax channels
Hi,
I'm not sure if I just missed something somewhere along the way, but I
noticed while I was going through the CVS logs that there is an option
in the wcfxs module to set an opermode - which apparently might help
with echo issues around the globe (like the ones I'm seeing - some
times).
So
Thank you, Ken.
I just asked because one from our clients is using this system. and
I have to configure a few phones for connecting to their network/pbx.
It really disaster to me. no SIP no docs... I just would like to
cry a little :) and see if anyone can say anything good and
Hi!
What's your iax.conf config files look like on both end? And your dial
statements in the extensions.conf file? Also, what version of Asterisk are
you running locally, remotely?
Small note: I had weird IAX2 problems with CVS-HEAD of yesterday -
updated again to current CVS and things are
Modprobe wcfxs opermade=UK is what I was using - if my card worked =)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Yager
Sent: 02 July 2004 1:19 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Zaptel, Line Impedence and Echo
Hi,
I'm not
I think all you need to do is
modprobe wcfxs opermode=AUSTRALIA
after the module is loaded
Chris
- Original Message -
From: Andrew Yager [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 02, 2004 1:18 PM
Subject: [Asterisk-Users] Zaptel, Line Impedence and Echo
Hi,
I'm
I just installed it on handytone, and I cant access web based
administration. any idea how to get it back?
On Fri, 02 Jul 2004 17:33:30 +1000
Master Abi [EMAIL PROTECTED] wrote:
New firmware version at http://www.hellofone.com/downloads.html.
Might fix the no register issue and others.
I
Hi,
Is there any program for ZAPTEL FXO with which I can debug the
signals that are coming from PSTN (Tones, Voltages, Ampers, etc.)?
In case that I have to do this program which is the closest entry
point of the ZAPTEL software?
Best Regards,
Miroslav Nachev
COSMOS
Is there a way to specify that info for the zaptel init.d script?
(as a side note - I'm talking on my Cisco 7960 via the FXS now, and it
sounds fine)
Andrew
_
Andrew Yager
Real World Technology Solutions
Real People, Real SolUtions (tm)
ph: (02) 9945 2567 fax: (02) 9945
Senad Jordanovic wrote:
Brian Weaver wrote:
I have a Sipura 2000 working fine, but whenever
I dial any extension there is a delay of 5-10 seconds before
it starts ringing. However, if I dial the extension and hit
the pound key after the number, it goes through right away.
Is there any way around
Glynn Condez wrote:
Hi all,
I would like to ask if Asterisk will allow to be monitor via web browser. I
am planning to create a web interface to monitor the current sip connected
end points and status of iax channels use.
If i write a code in php to execute this command should it be possible?
Dear Ted
my problem is not related to Voip Provider i use two * box:
- Box1 is my local network PBX with 2 BG phone , TDM400 FXS and E100P
If i can from BG phone to TDM400 FXS i have oneway audio also if i call to
E100P this happen with yestarday CVS HEAD, but if rollback to stable branch
On 02/07/2004, at 10:43 PM, Andrew Yager wrote:
Is there a way to specify that info for the zaptel init.d script?
In answer to my own question - yes there is.
You should modify /etc/init.d/zaptel and find the line that reads
insmod ${x} ${ARGS}
Change it to read:
insmod ${X} opermode=AUSTRALIA
Hi-
Only pins 1-2 and 4-5 are used, so one of the two cables should work.
(probably the straight through cable)
On your zaptel, you should use the phone company as the source for
asterisk's internal clock, like:
span=1,1,0,ccs,hdb3,crc4
(But I don't think this would make a difference on the red
Kevin Walsh wrote:
Leif Madsen [EMAIL PROTECTED] wrote:
Also, from what I have been told (and I've tested this by building
zaptel, but not any of the other sources) is that you no longer need
the sourcecode with the 2.6 kernel. You can create a symlink to:
/lib/modules/`uname -r`/build/
Instead
Dorian Gray wrote:
Kevin Walsh wrote:
Leif Madsen [EMAIL PROTECTED] wrote:
Also, from what I have been told (and I've tested this by building
zaptel, but not any of the other sources) is that you no longer need
the sourcecode with the 2.6 kernel. You can create a symlink to:
/lib/modules/`uname
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
You can access web browser only using its LAN network connection and
getting (or setting) a IP by builtin DHCP.
IMHO its bad.
Telles
Hekuran Doli wrote:
| I just installed it on handytone, and I cant access web based
| administration. any idea how
I work for an Inter-Tel dealer and sorry, but the system that you are
describing is PROPRIETARY.
The current iteration of the Axxess platform does support SIP through a SIP
gateway that Inter-Tel cleverly packages with some additional software to
drive the cost up.
Another alternative would be
After my inquiry on-list I contacted a couple of Zultys resellers as
well as Zultys tech support themselves. What I found about the 4x5 is
as follows:
1. The phone is not yet in widespread distribution. While they may be
taking orders and have a few select beta sites, they are not in general
Vasyl,
Not sure what kind of setup you're trying to do but if its a build out of an existing
system you're two options are pretty much as follows:
1. Proprietary System Integration
In this scenario you would use the Inter-Tel IPC. Supposedly they have a new card
that supports SIP as well as
On Thu, 2004-07-01 at 18:27, chouck wrote:
Thanks robert, But im having a problem trying to add a user that can login,
im using a sipura voip box trying to connect to the server and it always
gives me SIP/2.0 403 Forbidden. Under what config can I allow users and
hows it work exactly? Thanks
Make sure your dial-plan includes the extensions. Example:
(911|1xxx[2-9]xx|2xx)
Allows dialing of
- 911
- 1+10 digit long distance
- 3-digit extensions beginning with 2
Once the dialed digits match any part of the dial plan, they're sent to
asterisk.
-Original Message-
From:
I have been doing so much reading on phones lately that I have completely
lost track of some things. I seem to remember that there was one series of
Cisco IP phones that required Cisco's call manager. Does anyone know if the
7960 will work with Asterisk or does it require call manager?
hi,
Am Fr, den 02.07.2004 schrieb Scott Stingel um 15:31:
Hi-
Only pins 1-2 and 4-5 are used, so one of the two cables should work.
(probably the straight through cable)
ok
On your zaptel, you should use the phone company as the source for
asterisk's internal clock, like:
why 1 ? - the
It'll work, either as a SIP phone with the SIP image, or as skinny using
wither chan_sccp or chan_skinny (check the wiki).
Steve
-Original Message-
From: Matt Davies | MattDavies.Net [mailto:[EMAIL PROTECTED]
Sent: 02 July 2004 15:46
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
The 7960 works perfectly with Asterisk; I have them (and the 7940s)
running the SIP image with no problems whatsoever.
-Shaun
On Fri, 2 Jul 2004 08:45:44 -0600, Matt Davies | MattDavies.Net
[EMAIL PROTECTED] wrote:
I have been doing so much reading on phones lately that I have completely
On Fri, 2004-07-02 at 10:45, Matt Davies | MattDavies.Net wrote:
I have been doing so much reading on phones lately that I have completely
lost track of some things. I seem to remember that there was one series of
Cisco IP phones that required Cisco's call manager. Does anyone know if the
7960
On Fri, 2004-07-02 at 09:56, Wolfgang Pichler wrote:
hi,
Am Fr, den 02.07.2004 schrieb Scott Stingel um 15:31:
Hi-
Only pins 1-2 and 4-5 are used, so one of the two cables should work.
(probably the straight through cable)
ok
On your zaptel, you should use the phone company as
Does anyone know how to change the source IP address/Source Interface of
RTP packets? Changing the SIP source IP address in sip.conf has no
apparent impact on RTP. RTP traffic still uses the address assigned to
the outbound interface.
___
Cool, did you just use the standard ebuilds in portage (although the
'unstable' versions) ror did you build from cvs?
I have just received my hardware and want to build asterisk on a gentoo
box too :)
On Fri, 25 Jun 2004, Kevin Walsh wrote:
Ed Brady [EMAIL PROTECTED] wrote:
I am about to
Can someone help me, im using latest CVS, asterisk and cdr_mysql, when I
make a bridge call (using .call files in outgoing/) I always get
'billsec=12' in the cdr, both mysql and Master file even if the call lasted
longer, watching the Master file while making a call I see it updated at 12
seconds
Have you tried connecting an ordinary phone into the
line and
making/receiving calls?
I have, and it works fine. In fact, If I plug the
card into the line, and then a phone into the 'phone'
jack on the card, the line works fine (though I can't
imagine why that would make a difference).
Hi, I have an asterisk running great, with 2 cisco 7912 phones
converted to SIP, and a cisco 2600 xl w/ E1 and SIP. I´m thinking to expand the
test adding more 7912, but I prefer not convert all the 7912 to SIP, so I´m
tying to put CHAN_SCCP to work. I´ve get the sources from
Brian Weaver wrote:
I have a Sipura 2000 working fine, but whenever
I dial any extension there is a delay of 5-10 seconds before
it starts ringing. However, if I dial the extension and hit
the pound key after the number, it goes through right away.
Are you using pattern matching in your
Just starting to do the research on this oneI've got a customer who
is showing interest in replacing any older Panasonic unit providing
service to 96 tip/ring lines from a single PRI. Does anyone have any
recent experience with a decent (as in, plays nice with * and has a
reasonable per-port
ken-
it is worse than you think.
you have to have an IPRC(16 channels only) card in the Axxess cabinet with
current Call Processing software and then an additional server running
Inter-Tel's SIP gateway server (MS windows based just like their Call
Processing Server). And yes it is quite
On Fri, 2004-07-02 at 10:17, Morgan Gilroy wrote:
Can someone help me, im using latest CVS, asterisk and cdr_mysql, when I
make a bridge call (using .call files in outgoing/) I always get
'billsec=12' in the cdr, both mysql and Master file even if the call lasted
longer, watching the Master
I finally figured out how to create accounts and
actually login with some sip boxes and softphones, but how do I make it so they
have designated numbers just internally, err extensions i should say. So
if I had a user bob and bill, how would I set them up in config or how do I
setup the
Wolfgang-
If all of those check out, it really does seem like a protocol error of some
kind.
See if you can ask them what link of switch they are using to serve you. If
it's something unusual, perhaps google the digium site for references to
that switch.
Sorry that you're having so much
Gentoos great but everytime I see people talk about it and ask if theres any
special USE flags or crap like that to make somthing compile right I just
cant help but laugh.
and heres why : http://funroll-loops.org/
everyone needs a good laugh..
I'm not insulting Gentoo or anything, I like it.
We're doing some SIP development and have a question on additional parameters
supplied to the register (in this case maddr= and the non-standard clport= in
our example below).
What we're experiencing is the INVITE doesn't included these parameters
and they get dropped when the INVITE is sent to
Isn't this what the externip setting in sip.conf is for?
I have ran numerous tests - examining the SIP headers each time. I'm not
convinced that 'externip=' does anything at all.
With or without externip set in sip.conf, the headers send to my SIP client
(with nat=yes) will look exactly the
Thanks a lot,
What about latency and bandwidth? What codecs it uses? What max latency
is serves?
I am looking for connecting this handsets with the system with
latency about 250-300msec
Ken Wiesner wrote:
Vasyl,
Not sure what kind of setup you're trying to do but if its a build out
I have switched my voicemail configuration over to a mysql database and
everything works fine except the email. When a user gets voicemail
Asterisk reports: Jul 2 11:14:36 WARNING[1251156800]: app_voicemail.c:837
sendmail: E-mail address missing for mailbox.
It forks fine when running from
Oops, never mind that last post. I missed that it was fixed in cvs. Works
great now.
Jody N. Rudolph
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On 2 Jul 2004 at 16:17, Morgan Gilroy wrote:
Can someone help me, im using latest CVS, asterisk and cdr_mysql, when I
make a bridge call (using .call files in outgoing/) I always get
'billsec=12' in the cdr, both mysql and Master file even if the call lasted
longer, watching the Master file
Hello everyone,
I'm using a TE410P, no irq sharing, and all extraneous devices disabled,
such as USB, Parallel etc. I'm getting a few IRQ misses according to
ZTTOOL.
We're running a standard PRI_CPE interface and seem to be getting
dropped calls, and errors on the D-CHANNEL occasionally. The
Jason,
We've got the Axxess and we just have the IPC (8 channel) card with the IP PhonePlus
keysets. You only need to have the IPC for this type of configuration. It works, but
again, not as well as one would hope.
The SIP gateway is only required if you want to attach SIP endpoints to the
Hi,
nobody got any Idea?
;-(
- Original Message -
From:
wendys
To: Asterisk-Users
Sent: Sunday, June 27, 2004 8:43 PM
Subject: [Asterisk-Users] Optipoint 400
Standard Sip
Hi everybody,
I am testing Optipoint 400 Standard SIP (Firmware
2.3.14) with
Ryan Courtnage wrote:
With or without externip set in sip.conf, the headers send to my SIP client
(with nat=yes) will look exactly the same... no difference at all.
But have you tried nat=no? You missed the important point of my previous
message, that since you are using port forwarding, your
Hi,
I have a Cisco 7960 with SIP firmware which works perfectly with
Asterisk. I did get it working using chan_sccp originally. I couldn't make
chan_skinny work. All without callmanager.
HTH
Chris
--
Chris
--
E Mail: [EMAIL PROTECTED]
SIP: [EMAIL PROTECTED]
Hi all
I'm trying to configure a TE410P in Europe with three
E1s and a T1 channel bank (already bought in the US) to receive and make
calls through *. I managed to get the E1s to work, but I'm having
trouble with the channel bank (a Rhino). I have tried the bank in spans
1 and 4
Is there a way to specify that info for the zaptel init.d script?
In answer to my own question - yes there is.
You should modify /etc/init.d/zaptel and find the line that reads
insmod ${x} ${ARGS}
Change it to read:
insmod ${X} opermode=AUSTRALIA ${ARGS}
(or appropriate
Did anybody get this error message before:
chan_zap.c:5044 do_monitor: select return -1: Bad file descriptor
When it's happening, Asterisk gets freezed and talkers can not hear each
other. This message appears like in a loop at the server's screen.
thank you
Oz
12 ms is what I saw when I did * to * IAX.
In iax.conf, set:
notransfer=yes
That prevents IAX from transferring call to remote Asterisk, so it will
stay in path.
Sathya
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Friday,
Lenny Tropiano / asterisk.org Mailing list wrote:
We're doing some SIP development and have a question on additional parameters
supplied to the register (in this case maddr= and the non-standard clport= in
our example below).
What we're experiencing is the INVITE doesn't included these parameters
On Jul 2, 2004, at 9:43 AM, [EMAIL PROTECTED]
wrote:
Mediatrix only goes up to 24 port, as far as I can tell, which puts me
around 13k of just their hardware. And it just doesn't seem quite as
carrier class as a traditional channel bank to me.but I'm just
going
on gut feeling
From: Marc Spiegelman [EMAIL PROTECTED]:
Does anyone know how to change the source IP address/Source Interface of
RTP packets? Changing the SIP source IP address in sip.conf has no
apparent impact on RTP. RTP traffic still uses the address assigned to
the outbound interface.
You can't. It's
Hi!
I'm trying to compile the gastman´s sources with the
vc++(6.0) and to the Borland C++(1.0), but when I
compile these sources shows many erros like, there
aren´t Libraries and funtions necessaries. I tried to
get these at the Internet but I didn´t get all and
somethings with erros.
I would like
Hi!
I'm trying to compile the gastman´s sources with the
vc++(6.0) and to the Borland C++(1.0), but when I
compile these sources shows many erros like, there
aren´t Libraries and funtions necessaries. I tried to
get these at the Internet but I didn´t get all and
somethings with erros.
I would like
My question pertains to the use of IAE..
I would like to setup 2 Asterisk boxes. One would be located in our
office behind the firewall and hooked up to our analog lines. The other
would be located in a remote datacenter and used for our remote employees
to connect to. I would like to be able
Bryan:
I would like to setup 2 Asterisk boxes. One would be located in our
office behind the firewall and hooked up to our analog lines. The other
would be located in a remote datacenter and used for our remote employees
to connect to. I would like to be able to accept calls on the Office
Dimitri: Strange...I haven't come across your problem before. Have you
tried bugging the list or checking the Bug Tracker?
Greg: Honest answer? I have the same problem entering FWD numbers in an
FWD gateway (DTMF issue). I'm sort of passively looking for a fix for
that, as it's not the most
On Fri, 2004-07-02 at 15:49, Bryan Brannigan wrote:
I would like to setup 2 Asterisk boxes. One would be located in our
office behind the firewall and hooked up to our analog lines. The other
would be located in a remote datacenter and used for our remote employees
to connect to. I would
Hi,
Here it is when I start it with /etc/rc.d/init.d/asteriskd found in asterisk
source contrib/init.d/rc.redhat.asterisk
It started without problem and when i issue stop now It freezes, please
see below,
tai*CLI
add debug dontdumpextensions
Depending on what you are planning to do in the datacenter you could
just put SIP phones/ATAs there rather than a full Asterisk server but
that would require some care in configuring your firewall.
Actually the users are will be remote to the datacenter. The IPs in our
office are dynamic so I
Same problem here - with latest 0.6.3a oh323. Locks up on exit. Had to
kill -9
This didn't happen with 0.6.2a, but that's on a different machine. Maybe
you could try this older version which worked fine (same PwLib and OpenH323)
Regards
Scott
Scott M. Stingel
President,
Emerging Voice
All,
I have spent the last couple of days looking through the mail archives and
the documentation on the Wiki, but have not been able to find a solution to
the problem. The version of code I am running is from CVS as of 6/30/04.
What happens is that when I make an IAX call to another IAX client
On Fri, 2004-07-02 at 16:22, Bryan Brannigan wrote:
Depending on what you are planning to do in the datacenter you could
just put SIP phones/ATAs there rather than a full Asterisk server but
that would require some care in configuring your firewall.
Actually the users are will be remote
Folks!
Netweb Group, Inc. fully supports connectivity to any Asterisk PBX systems you have
and can provide A-Z termination with immediate effect.
Any volume is good enough for us, even an amount as small as $1.00 a day will do for
us.
We will provide connectivity from our Softswitch IP
Dear All,
I don't know but I tried all 0.6.x version of OH323 and normally I use
safe_asterisk to start asterisk, and everytime when I use 'stop now' to
terminate asterisk, it does not do anything, and you are rite, I have to use
kill -9 to kill the PIDs and threads. However, if I use asterisk
Brought to you by the fine folks at citigroup.com?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu
Sent: Friday, July 02, 2004 3:44 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Termination for Asterisk Users -
Hi,
I'm kind of a newbie myself. I've had similar problems and it can be
very frustrating. I did get them all resolved so I'll share some of what
I did in hopes that it will fix your issue.
To get some of my phones to work (Grandstream BT100) I had to add a line
nat = yes in my sip.conf under
iH
i received the card earlier this week but could not get the FXS port to
work. The FXO port did work for a short time but now fails. (no lights
on the card) and reports a number of errors to syslog concerning power
up failures.
would like to return the card and get a replacement
thanks
-
[EMAIL PROTECTED] wrote:
Does this only work with the new fxo modules?
Yes.
Richard
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This is what keeps my (CVS-HEAD) server happy..
bindaddr = 192.168.0.200 ; Local interface
externip = 80.63.xxx.xxx ; Public IP address
localnet = 192.168.0.0/255.255.255.0 ; Local LAN, internal clients etc.
(localnet can be repeated for each local LAN segment)
Server
Hi there
I am pretty close on giving up on Asterisk :-/
I am (still) trying to make a call from a H323 phone to an Asterisk
provider using AIX. But H323 does not route the number to AIX. All it is
transmitting is an s.
*CLI -- Executing Dial(OH323/R27865,
IAX2/demo:[EMAIL PROTECTED]/s)
from the zaptel sample config:
# dacs: The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a colon
# dacsrbs : The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a
A trace to the IP gives valuenet : http://www.valuenet.net/ a colo provider on
Level3.
I have a dislike for this kind of targeted spam on mailing lists, and are they
harvesting email addresses from their subscription .. I suggest nobody contact
them else they will think this is acceptable.
I
On Fri, 2004-07-02 at 18:17, Stephen J. Wilcox wrote:
A trace to the IP gives valuenet : http://www.valuenet.net/ a colo provider on
Level3.
I have a dislike for this kind of targeted spam on mailing lists, and are they
harvesting email addresses from their subscription .. I suggest nobody
Here is some code to do authentication by caller ID for DISA through AGI.
My original code had a bug in the Mysql query code, and there was a hangup
in the wrong place
[that's what I get for coding something at 2:00am], but the attached code
works correctly.
Take note of the REGEXP for the
You can connect two PRI devices together using a T1 crossover cable.
from the zaptel sample config:
# dacs: The zaptel driver cross connects the channels starting at
# the channel number listed at the end, after a colon
# dacsrbs : The zaptel driver cross connects the
On 04:53 PM 7/2/2004, Jason Garland wrote:
You can connect two PRI devices together using a T1 crossover cable.
but that defeats the purpose of putting the asterisk box in the middle.
The idea here is to install an asterisk box between the carrier and the PBX
and initially just pass through the
On Friday 02 July 2004 20:09, Chris A. Icide wrote:
On 04:53 PM 7/2/2004, Jason Garland wrote:
You can connect two PRI devices together using a T1 crossover cable.
but that defeats the purpose of putting the asterisk box in the middle.
As does DACS.
The idea here is to install an asterisk
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