I have not studied SIP yet.
I have an Asterisk server, and SIP phones somewhere on the globe, not locally
connected to the Asterisk server.
What is the bandwidth I need to the Asterisk server?
I assume that only the registration is the bandwidth to the Asterisk server,
while the phone call
Hi list,
every now and then I get the following message in my * logs:
chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
D-channel of span 1
As this is only a notice and voice worked quite well, despite the messages,
I didn't bother.
_But_, I wanted to try spandsp/fax
Hi list,
I face 2 problems with a today (11/04/04) CVS version:
H323:
Nufone h323: calling Dial(H323/extension) give in logs
Host: extension Username:
placing outgoing call to :1721 - this is my GK port
h323_make_call failed(H323/extension)
[...]
dialstatus=chanunavail
Calling from a
Bounty bumped to $US1,000 11/04/04
I've been able to secure some pledges from 2 people outside
of the asterisk community that should this feature be made available that they
will contribute $300 and $200 respectively.
This bounty now stands at $US1,000.
So far I haven't received any
I saw a previous post about this but I can't find it,
CVS-HEAD-11/03/04-14:09:34 does not pass ALERT_INFO to the phones. It
used to work but has now stopped. I'm not a coder so I can't look
through the code but someone mentioned ALERT_INFO does not exist in
app_dial if I remember correctly.
Hello everybody
Do you remember I sent a case to the list about a digit 1 phantom I
received when I call the method get_data or stream_file? Fine. I realized
that It does not happend when I omit a subrutine I my code where I open a
TCP client socket by IO::Socket.
I think It is because
Hi Damon,
I have the Eicon Diva Server BRI card working fine on my box.
What is important (if you want to use kernel 2.6) is that you
need to use at least kernel 2.6.9 because a lot of CAPI/Eicon
fixes where added.
Divactrl is needed to load the firmware on the card:
On Thu, 4 Nov 2004, Kurt Bauer wrote:
Hi list,
every now and then I get the following message in my * logs:
chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
D-channel of span 1
As this is only a notice and voice worked quite well, despite the messages,
I didn't
On Thu, 2004-11-04 at 08:56 -0500, david winter wrote:
All,
I was going to mess around with meetme on my cvs install of asterisk, but the
app_meetme does not seem to be there. I did a normal make
clean;make install. ran asterisk, did a 'show applications' and its not there. i
have
On Thu, 2004-11-04 at 11:15 -0400, Victor Cartes wrote:
Hello everybody
Do you remember I sent a case to the list about a digit 1 phantom I
received when I call the method get_data or stream_file? Fine. I realized
that It does not happend when I omit a subrutine I my code where I open
We recently switched to 729. I wouldn't expect that to cause built-in
conferencing to stop working.
Matthew
- Original Message -
From: Kevin P. Fleming [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, November 03, 2004 5:28
All,
What is a good multiline sip phone for an operator? Model and and
manufacturer.
I presume the multiline phone looks like 4 or 8 independent SIP
phones and asterisk would handle that by a call queue.
Then the operator just does her normal routine answering calls etc...
Thanks for the
Hi,
I connect a X100p in a Analog PBX extension.
If I want to call a analog extension (e.g.: using a softphone), the
asterisk pick up the extension and dial perfectly.
If I call the extension where where the X100Pp is connected (inside the
company), the asterisk doesn't answer the call.
I do:
--On Thursday, November 04, 2004 03:19:56 PM +0100 Peter Svensson
[EMAIL PROTECTED] wrote:
On Thu, 4 Nov 2004, Kurt Bauer wrote:
Hi list,
every now and then I get the following message in my * logs:
chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
D-channel of span 1
As
Hi,
Somebody
have any idea how I can config a CISCO IP CONFERENCE
STATION Model 7935 that work with * .
Thanks.
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To
Brian Wilkins wrote:
Sounds cool, but I heard a rumor that CF Cards don't like too many rewrites or
they start losing data.
Brian,
It boots read-only and uses ramdisks for RW stuff.
--
Kristian Kielhofner
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[EMAIL
Hi *,
[this goes to [EMAIL PROTECTED] and
[EMAIL PROTECTED]
i try to setup an VoIP conferencing server within a UML using asterisk
and it's 'MeetMe conference bridge'. I have several UMLs running other
services, but my asterisk know-how is poor (as you will see ;-).
Question #1: did
[EMAIL PROTECTED] wrote:
Without code samples, it is just a guessing game.
Maybe there's a bug where the TCP stuff is writing
to fd(0), which IIRC is STDIN
Guessing is fun!
--
Andreas SikkemaRits tele.com
Scheepmakersstraat 11 3011 VH Rotterdam
t: +31 (0)10 2245544
I've uploaded new packages to fix a couple of problems in the previous
releases. Again, these are for FC1 and are located in the standard place:
ftp://ftp.linuxsys.com/pub/releases/FC1/asterisk-v1.0/
Regards,
--
Andrew McRory - President/CTO
Linux Systems Engineers, Inc. -
Hello,
--- Anand S. Katti [EMAIL PROTECTED] wrote:
register = [EMAIL PROTECTED]/1000
register = [EMAIL PROTECTED]/1001
register = [EMAIL PROTECTED]/1002
Remove these register lines. You dont need them as your UACs are trying to register
with
Asterisk.
[sourabha]
username=1001
The context
Quick Question that I hope someone can answer.
Will Asterisk work with basic PCI FaxModems instead of those expensive
cards listed on the hardware page?
--
Mike Shultz
[EMAIL PROTECTED]
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On Thu, 2004-11-04 at 02:58, Larry Hendrickson wrote:
Hi all,
This is my first post to this list, so I apologize if it is a newbie
question. I did quite a bit of reading and a number of Google searches
for answers and found people with dynamic DNS problems, but not the
same one.
I just
Quick Question that I hope someone
can answer. Will Asterisk work with basic PCI FaxModems instead of those
expensive cards listed on the hardware page?--Mike
Shultz[EMAIL PROTECTED]
Answer is here http://www.voip-info.org/wiki-Asterisk+Hardware
Answers to the next
Hi Mike,
Yes it may work with some modems depending
on which chipset they use (in fact there are some advertised as supporting
asterisk) however a number of us choose to buy from Digium in order to show our
support for how much effort they put in by developing Asterisk.
Cheers,
Dean
I am interested in implementing Asterisk and someday
hope to have it replace my 8 x 24 Nortel switch.
However, I was told by a Telcom friend that my multi
line phones (Nortel 7208s) may not work with Asterisk.
This is a huge concern because in my business we are
constantly jumping back from one
The reason I went with kernel 2.6.9 is that it contains all
the needed CAPI support while a 2.4 kernel needs to be
patched (check melware.de).
I am using Fedora Core 2 with the latest kernel from rawhide
(2.6.9-1.640 iirc). So no requirements other than not wanting
to patch the kernel.
for voice they may or may not work, but the audio quality is bound to
be bad... unless you're really lucky :)
On Nov 4, 2004, at 16:10, Mike Shultz wrote:
Quick Question that I hope someone can answer. Will Asterisk work
with basic PCI FaxModems instead of those expensive cards listed on
the
On Thu, 2004-11-04 at 10:10 -0500, Mike Shultz wrote:
Quick Question that I hope someone can answer. Will Asterisk work
with basic PCI FaxModems instead of those expensive cards listed on
the hardware page?
Did you bother checking the wiki or using google to check past messages?
Or did you
On Thu, 2004-11-04 at 10:10, Mike Shultz wrote:
Quick Question that I hope someone can answer. Will Asterisk work
with basic PCI FaxModems instead of those expensive cards listed on
the hardware page?
No. The wcfxo driver only works with a very specific Intel/Ambient
chipset.
-Seth
--
Seth
You might have better luck with app_conference (see the wiki) under
UML.. It is probably a little more tolerant of loose timing and scheduling.
-SteveK
nils toedtmann wrote:
Hi *,
[this goes to [EMAIL PROTECTED] and
[EMAIL PROTECTED]
i try to setup an VoIP conferencing server within a UML
I know that the 7910 only works with Skinny. We have a possible client that
wants to bring 80 lines to us off his current provider. All 80 of his phones
are Cisco 7910s. Does anyone run 7910s with chan_sccp or the like and find
that it works good?
Thanks,
Matthew
Yes look at Ebay for x100P compatible cards
On Thu, 04 Nov 2004 10:10:50 -0500, Mike Shultz [EMAIL PROTECTED] wrote:
Quick Question that I hope someone can answer. Will Asterisk work with
basic PCI FaxModems instead of those expensive cards listed on the hardware
page?
On November 4, 2004 10:10 am, Mike Shultz wrote:
Quick Question that I hope someone can answer. Will Asterisk work with
basic PCI FaxModems instead of those expensive cards listed on the
hardware page?
Quick answer: No.
Read up on the Wiki about this -- you can likely get it to work but
What is a good multiline sip phone for an operator? Model and and
manufacturer.
The list that I came up with for multiple line/presence SIP phones is:
Snom 190, 200 (5 lines)
Snom 220 (Expandable number of lines)
Cisco 7940 (2 lines)
Cisco 7960 (6 lines)
Polycom IP 500 (3 lines)
Polycom IP 600 (6
On Thu, 4 Nov 2004, Kurt Bauer wrote:
Is your timing source set correctly? If you are connecting to the pstn
the pstn connection should be the primary timing source.
connection is to a Ericsson MD110 wich is set as network, * is set as CPE.
Have you set the span as the timing source?
HI all,
I have a question and I cant seem to find the answer
anywhere. Is there a way to limit the
amount of digits dialed? For example I
have a * box set up for the department of corrections for prisoners to call
home. It has the Digium
2 FXO/ 2 FXS card in it. I have two
Lines brought
On Thu, Nov 04, 2004 at 07:18:38AM -0800, Richard Reina said:
I am interested in implementing Asterisk and someday
hope to have it replace my 8 x 24 Nortel switch.
However, I was told by a Telcom friend that my multi
line phones (Nortel 7208s) may not work with Asterisk.
This is a huge
This may be a no brainer for some of you out there, simply put it seems
that we have a problem passing DTMF from IAX to SIP. The digits cannot
be heard coming from the IAX side nor do they seem to register in
Asterisk. This seems to happen with any Codec we use so that part has
been ruled out.
I sent this to the list earlier but I never saw the post
show up, I apologize if this is a repeat post.
___
HI all,
I have a question and I cant seem to find the answer
anywhere. Is there a way to limit the
amount of digits dialed? For example I
have a * box
Sorry, problem solved, it's my
mistake..
- Original Message -
From:
Serge
To: [EMAIL PROTECTED]
Sent: Thursday, November 04, 2004 9:38
AM
Subject: [Asterisk-Users] res_features.so
Segmentation fault
Have anyone
some idea ?Asterisk - latest
The list that I came up with for multiple line/presence SIP phones is:
Snom 190, 200 (5 lines)
Snom 220 (Expandable number of lines)
Cisco 7940 (2 lines)
Cisco 7960 (6 lines)
Polycom IP 500 (3 lines)
Polycom IP 600 (6 lines)
ipDialog SipTone (2 lines)
Zultys 4x4, 4x5 (4 lines)
aastra
I'm assuming nobody has experience with running ISDN / BRI over H.323...
-Original Message-
From: Huddleston, Robert [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 03, 2004 8:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] H323 ISDN
Hi,
That's problaby a easy question to solve but I couldn't figure out how to do
what I need.
My PSTN line is connected to a phone and a FXO card. What I need is when
someone calls me, and I don't answer in 3 or 4 rings, * makes a VoIP call to
my office, where I'll pickup that call. Or I want to
On November 4, 2004 10:56 am, Henry Devito wrote:
I have a question and I can't seem to find the answer anywhere. Is there a
way to limit the amount of digits dialed? For example I have a * box set
up for the department of corrections for prisoners to call home. It has
the Digium 2 FXO/ 2
Does anybody know how to stop the AGI process after the user Hang-Up? 'Cause
it stills running if the transaction is ended by the user.
Thanks
Víctor
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On Thu, 2004-11-04 at 16:57 +0100, Serge wrote:
Sorry, problem solved, it's my mistake..
Please share more information. Asterisk shouldn't segfault on simple
user problems.
--
Steven Critchfield [EMAIL PROTECTED]
___
Asterisk-Users mailing list
Hello,
For those of you who have a CAPI card with an on-board DSP (like some Eicon
Diva Server), this patch allows you to receive faxes.
If you want to answer a channel in fax mode, use capiAnswerFax() instead of
Answer()
If you use Answer(), you will be in voice mode. If the hardware DSP detects
Hi List
I recently got one of those boxes (without wlan).
Works as ata device with my local asterisk.
Just tested the basic stuff like call the box and make call from the box.
It uses sip with alaw/ulaw/g726 codecs.
Runs on linux, kernel 2.4.17, mipsel.
HardWare:
-wan (UR2/annexB)
-ethernet
I'm in the process of building up a small (1x1) test Asterisk box
based on a 1Ghz VIA C3 Mini-ITX box with one PCI slot (and a FX100P).
Anyone have suggestions as to the best Linux distribution (or kernel)
to base the system on?
I'll just have one FXO/POTS line and then a Grandstream Budgetone
Following on from the message below I have discovered that the X100P
causes the SNR on my ADSL line to drop even with the Asterisk box
**switched off** and the power unplugged... This seems very strange..
Why should a card in a switched off PC cause noise on a line meaning
that it drops out
Does anyone that the source for app_valetparking.c
Thanks
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To UNSUBSCRIBE or update options visit:
On Thu, 2004-11-04 at 17:31, Carl Sempla wrote:
Hello,
For those of you who have a CAPI card with an on-board DSP (like some Eicon
Diva Server), this patch allows you to receive faxes.
[snip]
A fix for a dead lock issue is also included (Oct 22 18:06:00
WARNING[11275]: channel.c:472
On November 4, 2004 11:39 am, Bill Bradford wrote:
Anyone have suggestions as to the best Linux distribution (or kernel)
to base the system on?
I'm a fan of a trimmed-down slackware but there are smaller distros yet. My
entire * setup fits into less than 500MB and that's without really
Im getting the following error when I do an strace -p # for asterisk
. Its taking alot of resources.
ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, 0xb290) = -1 EIO
(Input/output error)
write(1, voip1*CLI , 11) = -1 EIO (Input/output error)
I checked the WIKI and the list and
On Thu, 2004-11-04 at 10:11 -0700, kyle Hagan wrote:
Im getting the following error when I do an strace -p # for asterisk
. Its taking alot of resources.
ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, 0xb290) = -1 EIO
(Input/output error)
write(1, voip1*CLI , 11) = -1 EIO
Thank you very much for your thoghtful and thorough
response.
I guess I don't wan't to set up * to behave like a key
system, thank godness, I just want to be able to
juggle calls which it sould like Asterisk can do fine.
Just to clarify though, can the polycom IP 500 / 600
work on analog lines?
Im not running strace constantly, I just run it to see whats going on.
And that error it coming in on the Asterisk PID at a rate of about 10
per second, from what I can see.
Kyle
Steven Critchfield wrote:
On Thu, 2004-11-04 at 10:11 -0700, kyle Hagan wrote:
Im getting the following error
Im not running strace constantly, I just run it to see whats going on.
And that error it coming in on the Asterisk PID at a rate of about 10
per second, from what I can see.
here is my TOP:
Cpu(s): 86.3% user, 13.7% system, 0.0% nice, 0.0% idle
Mem:904204k total, 862688k used,
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that companies
DISA to complete calls. So in order to stop that I have to suppress dtmf
after so many digits are dialed. Any idea's?
-Original Message-
From:
I'm using RH9 and Mandrake 10, because they were easy to install. I
heard good things about Gentoo when properly built and tweaked, but it
requires some effort. Search the list archives for previous
discussions.
-Original Message-
From: Bill Bradford [mailto:[EMAIL PROTECTED]
Sent:
You have to use DeadAGI instead of AGI to call your script, for example:
exten = 77,1,Answer
exten = 77,2,DeadAGI(astcc.agi)
exten = 77,3,Hangup
Regards..
Nahuel Ramos.
On Thu, 4 Nov 2004 13:14:56 -0400, Victor Cartes
[EMAIL PROTECTED] wrote:
Does anybody know how to stop the AGI
Sorry for the OT message but I'm very curious to see if anyone on this
list uses pointone for long distance sip call termination?
We've been having an off and on problem with them saying they do not
support sip message with a fqdn in the from field.. which to me appears
to be a breakage of the
On Thu, 2004-11-04 at 11:30 -0600, Henry Devito wrote:
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that companies
DISA to complete calls. So in order to stop that I have to suppress dtmf
after so many digits
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
I'm using Fedora RC1 without problems (kernel 2.4.22)
Anyone here have tried Whitebox Respin 1 with asterisk ?
Maybe nest week i'll install a small Asterisk Server on it.
Asterisk + Apache only.
Bill Bradford wrote:
| I'm in the process of
On November 4, 2004 12:30 pm, Henry Devito wrote:
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that companies
DISA to complete calls. So in order to stop that I have to suppress dtmf
after so many digits are
Karl Brose wrote:
The REGISTER requests that your SIP UAs are sending as listed are not
requests to *register*, but request to *unregister*
The contacts are '*' and expirations are '0'
Granted that Asterisk doesn't do registrations correctly, but it does
need a proper registration request with
João Amaro wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
I'm using Fedora RC1 without problems (kernel 2.4.22)
Anyone here have tried Whitebox Respin 1 with asterisk ?
Maybe nest week i'll install a small Asterisk Server on it.
Asterisk + Apache only.
Bill Bradford wrote:
| I'm in the
Does anyone in the list have a fully functional ASTCC and
would like to share their CGI, AGI and CONF files/Scripts
and database installation that is customized for:
1) Accepting user input for a Pin for authentication
2) Recognizes the caller id for authentication
3) Has a better GUI to
Hi
I am trying the fine iso at http://www.asterisk.de.ms/ but are having
problems with Capi probably due to having to old Fritz PCI card. Trying
with both non version marked version and version marked V 2.0.
I get following error when booting Astrisk on Debian:
Oct 31 02:26:10 asterisk kernel:
Thanks to everyone who answered my stupid
question. I did not see the wiki nor an answer to my question on their
main site. My main experience is with a
3Com NBX 25 which is small and really simple so I never had to learn
much except the dial plan, which is why most of this is still foreign
to
Hi everyone,
I'm having problems with Playback() on a Grandstream Budge Tone-100.
Every time Playback is used I get the following messages:
WARNING[229388]: file.c:550 ast_readaudio_callback: Failed to write frame
== Spawn extension (from-sip, , 1) exited non-zero on 'SIP/2002-559c'
I
Sounds more like a requirement for custom development since I'm sure
your needs will vary from some others that are also using astcc as a
starting point for their prepaid cards
-- William
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[EMAIL PROTECTED]
On 11/4/2004, Victor Cartes [EMAIL PROTECTED] wrote:
Does anybody know how to stop the AGI process after the user Hang-Up? 'Cause
it stills running if the transaction is ended by the user.
I'm unsure (haven't had a need yet to use AGI), but perhaps you could
use the DeadAgi application in your
I'm using * in a prepaid environment with Whitebox Respin with all the bells
and whistles loaded.
I made the move when Redhat lost their mind and discontinued support of 9.0.
The asterisk is still in test and development mode (not a lot of traffic
yet) but it seems to work okay. I am, however
On Thu, 4 Nov 2004, Luciano Macedo Rodrigues wrote:
That's problaby a easy question to solve but I couldn't figure out how to do
what I need.
My PSTN line is connected to a phone and a FXO card. What I need is when
someone calls me, and I don't answer in 3 or 4 rings, * makes a VoIP call to
On Nov 4, 2004, at 9:23 AM, Richard Reina wrote:
Thank you very much for your thoghtful and thorough
response.
I guess I don't wan't to set up * to behave like a key
system, thank godness, I just want to be able to
juggle calls which it sould like Asterisk can do fine.
Just to clarify though, can
Hi everyone,
I have a question regarding the use of callerID and call
forwarding. When I forward any of my Zap extensions in
the office to an outside line, such as a cell phone, the
callerID info shows up as originating from that office
phone, rather than from whoever actually originated the
Not necessarily. The need is a generic calling card app - take user
Input or recognize the ANI, Allow the calls
The users and pins are stored in Mysql database
In order to make the database easy to manage - as the users and pins are
stored in Mysql database,
PHPMysqlAdmin (which is a generic GNU
Thank you very much for that clarification.
--- Scott Laird [EMAIL PROTECTED] wrote:
On Nov 4, 2004, at 9:23 AM, Richard Reina wrote:
Thank you very much for your thoghtful and
thorough
response.
I guess I don't wan't to set up * to behave like a
key
system, thank godness, I
On Thu, 2004-11-04 at 13:15 -0500, Mike Shultz wrote:
Thanks to everyone who answered my stupid question. I did not see the
wiki nor an answer to my question on their main site. My main
experience is with a 3Com NBX 25 which is small and really simple so I
never had to learn much except the
On Thu, 2004-11-04 at 13:14 -0400, Victor Cartes wrote:
Does anybody know how to stop the AGI process after the user Hang-Up? 'Cause
it stills running if the transaction is ended by the user.
You must handle t his in your AGI application. If you start getting back
broken reads from the STDIN
On Thu, 4 Nov 2004, Chris Goodwin wrote:
I have a question regarding the use of callerID and call forwarding.
When I forward any of my Zap extensions in the office to an outside
line, such as a cell phone, the callerID info shows up as originating
from that office phone, rather than from
ALL,
is it possible to plug a standard analog 56K modem into my
iaxy device and make a modem call out? 9600 baud call would
be fine actually. I just want to make a call out with my iAXy
device and eliminate my PSTN line.
THanks,
Jerry
___
Asterisk-Users
On 11/4/2004, Henry Devito [EMAIL PROTECTED] wrote:
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that companies
DISA to complete calls. So in order to stop that I have to suppress dtmf
after so many digits are
One of my customers use Grandstream for ASTCC and it suddenly stopped
recognizing
DTMF for my ASTCC Application.
When ASTCC asks to enter destination number, and when the the digits
Are entered, the phone keys does not take any of them. They are dead.
Any suggestions
Seshu Kanuri
from 172.16.3.13
Date: Thu, 4 Nov 2004 18:55:30 -
MIME-Version: 1.0
Content-type: text/plain; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable
Hi all,
I hope someone can shed some light on the following: - I came across
a thread with a similiar problem but it didnt fix the
Andrew Kohlsmith wrote:
On November 4, 2004 12:30 pm, Henry Devito wrote:
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that companies
DISA to complete calls. So in order to stop that I have to suppress dtmf
after
Hey Folks,
I am trying to light a MWI located on a remote SIP phone. In other words,
the phones register to one server but the voicemail app lives on a different
one.
I am guessing it has something to do with passing a user command in the
voicemail.conf file.
Of course I would also need to
Maybe they switched to a codec that doesn't support inband DTMF and it isn't
configured to use SIP INFO or likewise?
-Michael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu
(Company IT)
Sent: Thursday, November 04, 2004 12:53 PM
To:
On Thu, 2004-11-04 at 13:59 -0500, Andrew Thompson wrote:
Andrew Kohlsmith wrote:
On November 4, 2004 12:30 pm, Henry Devito wrote:
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that companies
DISA to
On 11/4/2004, Andrew Thompson [EMAIL PROTECTED] wrote:
In any case, chan_sip would be much more likely to be hackable to make
DTMF quit working.
Possibly, but his working configuration most likely doesn't use SIP (I
would presume):
It has the Digium 2 FXO/ 2 FXS card in it. I have two Lines
On Thu, 2004-11-04 at 13:50 -0500, Jerry Geis wrote:
ALL,
is it possible to plug a standard analog 56K modem into my
iaxy device and make a modem call out? 9600 baud call would
be fine actually. I just want to make a call out with my iAXy
device and eliminate my PSTN line.
Depends on
On 11/4/2004, Ashling O'Driscoll [EMAIL PROTECTED] wrote:
[general]
port =3D 5060 ; Port to bind to (SIP is 5060)
bindaddr =3D 0=2E0=2E0=2E0 ; Address to bind to (all addresses on machine)=
diallow=3Dall=20
allow=3Dulaw
context =3D from-sip ; Send SIP callers that we don't know about here
is it possible after a meetme call to keep going on in the context, like
the meet me is priority 2 I want it to hit priority 3 (after the party
disconnects)?
thanks
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Flynn wrote:
Possibly, but his working configuration most likely doesn't use SIP (I
would presume):
It has the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in
to the fxo ports and 2 standard 2500 analog sets for the prisoners to
use to dial out.
Yeah, I saw that, but the replies I'd
Hi Flynn,
Feel free to contact me offline if you feel this isn't suitable
conversation for online but I read an article about this a few weeks ago
about how you were freezing out other carriers for offering cheaper
calls to inmates than the inflated prices you charged. And I don't
understand how
Hi,
I am living a wear thing. I am using my asterisk with all kind of
NAT / PAT / NPAT, with multiple ports on the same network address and
all works perfect.
The problem I have been trying to solve is with a Cisco ATA behind a
Linksys NAT. The ATA's register work ok, but when I execute sip
Hi,
Thanks for the reply. Yes I had left out the 's'(as I had copied from
the previous thread) but that is not the problem. I still have the
'call leg transaction does not exist' error. I have included the
debug sip messages below if that will help any bit. I read that this
error should have
On Nov 4, 2004, at 10:28 AM, Chris Goodwin wrote:
Hi everyone,
I have a question regarding the use of callerID and call forwarding.
When I forward any of my Zap extensions in the office to an outside
line, such as a cell phone, the callerID info shows up as originating
from that office phone,
steve,
Thanks, do you recall what config commands you gave the modem
to drop it down and only connect at lower speeds? I'm not a modem guru.
Thanks,
Jerry
On Thu, 2004-11-04 at 13:50 -0500, Jerry Geis wrote:
/ ALL,
//
// is it possible to plug a standard analog 56K modem into my
// iaxy device
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