The combination of a cheap ATA and standard doorphone would seem easier (and maybe
cheaper).
For door phone hardware check out: http://www.vikingelectronics.com/
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: Adthrawn [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
DearSir / Madam,
I am a beginner of implement VoIP
system and need some help.
Weare
interested in developing a small scaled VoIP calling card system with 4 FXO port
in the primary run.
The whole
system will have two PCs, one in local and one in
overseas.
Each PC
will havefour Wildcard
Rich Adamson wrote:
Christian,
The packet trace indicates that when a Snom 200 is configured with
two extns, asterisk sends a Notify (with mwi=yes/no) every eight
minutes. The MWI on the panel displays the last notify received.
In my case, the notify for x3002 is received before the notify for
Jonathan Moore wrote:
I got mine to show the mwi, but when I press the button if gives me an error. I
think snom is now giving out a config guide for snom phones and * in pdf
format. I have a copy somewhere, but it only got me to the point above and it
just hasn't been a high enough priority for
This has been covered before. I think the reason is that asterisk sends
notifications
from [EMAIL PROTECTED] and pressing the button dials that address.
It's not fixed, yet.
For this, i have an voicemail-extension for my snoms:
exten = asterisk,1,GoTo(8518,1)
exten = 8518,1,VoicemailMain2
Hi,
-Original Message-
Does anybody know of a VoIP compatible doorbell or door intercom unit?
I've contemplated buying a cheap SIP phone, ripping it apart, and
putting it inside an IP66 sealed unit...
It would need:
- At least one speed-dial key, or some way to make every
Thomas Dingermann wrote:
This has been covered before. I think the reason is that asterisk
sends notifications
from [EMAIL PROTECTED] and pressing the button dials that
address.
It's not fixed, yet.
For this, i have an voicemail-extension for my snoms:
exten = asterisk,1,GoTo(8518,1)
exten
I am trying to calculate bandwidth needs. Is 1 T1 Line able to provide
488.5 Gigabytes of traffic for 1 month based on a 30 day week? I did my
calculation as follows:
1.544 mps * number of seconds in a minute(60) * number of minutes in a hour
(60) * number of hours in a day(24) * number of
Hello all,
We are having a really wierd problem with asterisk, basically after
about 80 calls, asterisk stops responding. The process itself is still
running, you can log into asterisk and events are still logged, however
all phones lose dial tone and no incoming or outgoing calls work.
Calls
Hi.
I have the same issue with budgetones 102 ( 101) with firmware 1.0.4.30
But happens also with .4.26 , .4.18 and .4.17 .
Doing an ethereal trace, I noticed that the GS isn't answering to OK's
sent by asterisk when the ringed party answers (GS doesn't not send ACK
to the cpnnection
Jim Flagg [EMAIL PROTECTED] said:
Agreed, Guess I should have said traditional computer. Most PBXs would
only use a hard drive for voice mail. A hard drive failure would not cause the
PBX to stop working.
You can stash Asterisk+Linux on a 64Mb flash. Next question please?
(and in fact, I
Jonathan Moore [EMAIL PROTECTED] said:
Any ideas?
Wait until 2am :-).
And of course, with well-managed security procedures around the system,
you probably will not be upgrading kernels with every bug. The last
couple of holes in the Linux kernel are locally exploitable only, and
with a * box
Quoting Ariel Batista [EMAIL PROTECTED]: [Say hi to Steve for me!]
I need to get 2 Asterisk servers working together. I have been reading
I have tried the command in the 2nd box of
[local]
switch = IAX2/redbox2:[EMAIL PROTECTED]/local
What I am looking for is a real example of
On Tue, 2004-01-06 at 21:08, Jonathan Moore wrote:
These are good issues, but I am even thinking of something simpler and more
common than crises. Such as this scenerio.
I need to update my Asterisk server that runs all my phones inorder to install a
kernel update that fixes a security bug.
Hi,
I am using a Cisco 1750 with FXO card in Australia to provide ports into
Asterisk.
I was wondering if anyone out there has a config for the cisco to detect
the disconnect or hangup signal for Australian tones.
If the calling party hangs up while leaving a voice mail for example, it
takes
Hello,
I can not seem to be able to get StripMSD and Prefix to work for me in
extensions.conf. This is an example of what I have:
exten = _050.,1,StripMSD,1
exten = _50.,Prefix,01051
exten = _001051.,1,Dial(${TRUNK2}/${EXTEN})
exten = _001051.,2,Busy
exten = _001051.,102,Busy
What I want to
Hi again,
Sorry, completely forgot about setting the priorities - all is OK
Regards,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dawid Mielnik
Sent: Wednesday, January 07, 2004 1:06 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] manipulating
On Wed, 7 Jan 2004, calvis wrote:
I am trying to calculate bandwidth needs. Is 1 T1 Line able to provide
488.5 Gigabytes of traffic for 1 month based on a 30 day week? I did my
calculation as follows:
1.544 mps * number of seconds in a minute(60) * number of minutes in a hour
(60) *
We have setup an asterisk box to let everybody call into the university
internal network, but I get unexpected hangups when doing an outbound call
from SIP to the ISDN interface, and it happens from 20 seconds to some minutes into
the
call.
--the dial and the problem---
--
On Tue, 2004-01-06 at 21:08, Jonathan Moore wrote:
These are good issues, but I am even thinking of something simpler and more
common than crises. Such as this scenerio.
I need to update my Asterisk server that runs all my phones inorder to install a
kernel update that fixes a
On Wed, Jan 07, 2004 at 01:06:12AM -0800, calvis wrote:
I am trying to calculate bandwidth needs. Is 1 T1 Line able to provide
488.5 Gigabytes of traffic for 1 month based on a 30 day week? I did my
calculation as follows:
1.544 mps * number of seconds in a minute(60) * number of
The packet trace indicates that when a Snom 200 is configured with
two extns, asterisk sends a Notify (with mwi=yes/no) every eight
minutes. The MWI on the panel displays the last notify received.
In my case, the notify for x3002 is received before the notify for
x3008, and in later
Hello,
I read an interesting article on 2cpu.com that talks about the performance
differences between a P4 processor with Hyperthreading(HT) enabled and
disabled.
http://www.2cpu.com/articles/ht_explored/index.html
The bottom line of the article is that processor intensive applications
greatly
Hi Guys,
is there a client which can be used on the SonyEricsson P800/P900...?
IAX would be cool, but i take anything that can connect (via bluetooth)
to an asterisk-server ;-).
The phone is Symbian, and can also execute java-stuff...
Greez
Andreas
Hi Matteo,
Send me the Ethereal SIP Trace and I will take a stab at it.
Regards,
Andres.
- Original Message -
From: Matteo Brancaleoni [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 4:50 AM
Subject: Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
I am trying to calculate bandwidth needs. Is 1 T1 Line able to provide
488.5 Gigabytes of traffic for 1 month based on a 30 day week? I did my
calculation as follows:
1.544 mps * number of seconds in a minute(60) * number of minutes in a hour
(60) * number of hours in a day(24) * number
Hi,
does anyone have any recommended (read tried and tested) way of making asterisk
be able to handle incoming faxes.
I've a PC running asterisk with a digium E1 card in and simply want to be able
to route a call to some application which will take a fax call and save the fax
as an image.
I
On Wed, Jan 07, 2004 at 01:42:30PM +, Andreas Anderson wrote:
Hi Guys,
is there a client which can be used on the SonyEricsson P800/P900...?
IAX would be cool, but i take anything that can connect (via bluetooth)
to an asterisk-server ;-).
The phone is Symbian, and can also execute
We are getting reports from the receptionist that callers are having dtmf
problems (131 gets read as 113, 117 read as 111) - first digit recognized
twice, third digit not read at all).
I have tried 6 different analog phones, two cell phones (on two different
networks), and a digital phone behind
Hi,
-Original Message-
is there a client which can be used on the SonyEricsson P800/P900...?
IAX would be cool, but i take anything that can connect (via
bluetooth)
to an asterisk-server ;-).
The phone is Symbian, and can also execute java-stuff...
I must agree, as a P800
On Wed, 7 Jan 2004, Sjur Eivind Usken wrote:
We have setup an asterisk box to let everybody call into the university
internal network, but I get unexpected hangups when doing an outbound call
from SIP to the ISDN interface, and it happens from 20 seconds to some minutes into
the
call.
You need to compile HDLC support at kernel. Prefer static then module.
Daniel
[EMAIL PROTECTED] wrote:
hello,
I have been using the T1 card with my asterisk for a while now, but an
attemp to upgrade the system to use a TE410P card ( using the T1 option)
i have a 3.3V motherboard. but when i
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Tuesday, January 06, 2004 2:25 PM
To: Asterisk-a-users-list; Christian Stredicke
Subject: [Asterisk-Users] Heads up v2.03h on snom 200
Christian,
Just a quick
Hi Matteo,
I see a problem here. And its the same as the trace I examined from
Wipeout.
After * sends back the 183 Session Progress message, it should also send
the STATUS 200 OK once the call is answered (the only STATUS 200 OK I see
is the response to an INFO Message). Since the GS never
Nicolas Bougues wrote:
On Wed, Jan 07, 2004 at 01:42:30PM +, Andreas Anderson wrote:
Hi Guys,
is there a client which can be used on the SonyEricsson P800/P900...?
IAX would be cool, but i take anything that can connect (via bluetooth)
to an asterisk-server ;-).
The phone is Symbian, and
Well, to do an upgrade on a traditional system you have the same issues, perhaps even
worse as everything is physically wired to one system. To develop for production you
must have a dev environment, a beta test and a scheduled release right?
Todd
Jonathan Moore [EMAIL PROTECTED] wrote:
I have 2 questions regarding asterisk logs that I really hope someone
can help me with.
Jan 7 09:40:14 WARNING[1009517568]: File chan_iax.c, Line 3537
(registry_rerequest): Received unsolicited registry authenticate request
from '209.242.15.34'
I get this IAX message every minute or so. I have
There are anyone using * with freeside (http://www.sisd.com/freeside/) for
billing ?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I have a problem with my asterisk and IAX2. It seems that I do not have
it or it's broken. I have been trying to connect to another asterisk
server and I was thinking I am setting it up wrong. But I am getting
this on the asterisk CLI for dialing out via IAX2 even to IAXTEL which
is now not
On Wednesday 07 January 2004 06:06, Dawid Mielnik wrote:
Hello,
I can not seem to be able to get StripMSD and Prefix to work for me
in extensions.conf. This is an example of what I have:
exten = _050.,1,StripMSD,1
exten = _50.,Prefix,01051
exten = _001051.,1,Dial(${TRUNK2}/${EXTEN})
I am just wondering if this is normal. I have
my * running for a week now and I'm still testing its interoperability with
other voip provider (in sip using codecs other than g711). yesterday, i changed
my linux's (RH9). and since the new ip i assigned is located on a different
site, i have
Hello,
My clients usually work the regular 8 - 5 day, however they
would like to have control of the night time context.
Is there any way, say a receptionist, can dial a 4 digit extension, to
toggle on/off the night time context?
Thanks in advance,
Brent
Are you doing these items before trying to
start asterisk:
loadmod driver (In my case its wcfxo)
After that is completed:
Ztcfg
This configures all of the zaptel harware in the system.
You might want to place this in init.d to get it to do it automatically.
- Brent
On Wed, 2004-01-07 at 06:59, Rich Adamson wrote:
On Tue, 2004-01-06 at 21:08, Jonathan Moore wrote:
These are good issues, but I am even thinking of something simpler and more
common than crises. Such as this scenerio.
I need to update my Asterisk server that runs all my phones
On Wednesday 07 January 2004 10:36, Jess Magnaye wrote:
I am just wondering if this is normal. I have my * running for a
week now and I'm still testing its interoperability with other voip
provider (in sip using codecs other than g711). yesterday, i
changed my linux's (RH9). and since the new
Have a question about implementing Call Rollover with my
current extensions.conf configuration.
[macro-stdexten]
exten = s,1,Dial(${ARG2},20) ; Ring the
interface, 20 seconds maximum
exten = s,2,Voicemail2(u${ARG1}) ; If unavailable,
send to voicemail w/ unavail announce
exten =
hmm.. i did the modprobe wct1xxp. but i didn't do
ztcfg bec i thought i will only need it if there's change in the zaptel
config. let me try it and i'll let u know.
thanks.
- Original Message -
From:
Brent
Franks
To: [EMAIL PROTECTED]
Sent: Wednesday, January 07,
On Wed, 2004-01-07 at 07:24, mattf wrote:
Hello,
I read an interesting article on 2cpu.com that talks about the performance
differences between a P4 processor with Hyperthreading(HT) enabled and
disabled.
http://www.2cpu.com/articles/ht_explored/index.html
The bottom line of the article
On Wed, 7 Jan 2004, Tilghman Lesher wrote:
On Wednesday 07 January 2004 06:06, Dawid Mielnik wrote:
Hello,
I can not seem to be able to get StripMSD and Prefix to work for me
in extensions.conf. This is an example of what I have:
exten = _050.,1,StripMSD,1
exten =
May be it's due to a kernel patch ... ??
Try to recompile zaptel, asterisk, ...
Van: Jess Magnaye [mailto:[EMAIL PROTECTED]
Verzonden: wo 7/01/2004 17:36
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] * crashed
I
On Wed, 2004-01-07 at 07:59, Stephen J. Wilcox wrote:
Hi,
does anyone have any recommended (read tried and tested) way of making asterisk
be able to handle incoming faxes.
I've a PC running asterisk with a digium E1 card in and simply want to be able
to route a call to some application
On Wed, 2004-01-07 at 10:38, Brent Franks wrote:
Hello,
My clients usually work the regular 8 - 5 day, however they
would like to have control of the night time context.
Is there any way, say a receptionist, can dial a 4 digit extension, to
toggle on/off the night time context?
Greetings...
I am trying to get Asterisk working with a Voicetronix OpenLine4 card.
Searching through the archives it seems that some have managed this
before - although I get the impression it's not a widely used card and
its use is not highly documented.
My vpb drivers were compiled
Does this mean I can run with G711 between ATA and *, and GSM between * and
voip-provider?
Example:
ATA - g711 via SIP - * - gsmfr via SIP - Cisco-VoIP - Pots
I am wondering because I am just getting silence. It looks like rtp coming
back from the voip-provider is not matching with my *
On Wednesday 07 January 2004 10:53, Steven Critchfield wrote:
Interestingly enough, when I nmap my primary 2 asterisk boxes I
don't even see the IAX ports. Need to think about getting nmap
patched for the VoIP ports.
Note that IAX and IAX2 are both UDP.
Anyways, I only have ssh,
aww BS, if you had any intention of sharing you would post it to
bugs and let others out there at it, REGARDLESS of if it makes it
into CVS.
shaking head
Chris Albertson wrote:
It's a chicken and egg type thing. I'd contribute some work
I've done. But why bother? It will just sit there
On Wed, 2004-01-07 at 11:01, Ryan R. Fligg wrote:
Have a question about implementing Call Rollover with my current
extensions.conf configuration.
[macro-stdexten]
exten = s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
exten = s,2,Voicemail2(u${ARG1})
my debug is
./asterisk -vcg
extension.conf is OK becaouce form ATA186 calls are working, aleso from
cisco7905 and cisco as5300. And of course insde * it is working to.
It doesn't matter, that the first is WAIT(1). It crashes any way.
Peter Hudec
Jeremy McNamara wrote:
Peter Hudec
On Wed, Jan 07, 2004 at 10:56:14AM -0600, Tilghman Lesher wrote:
WARNING[1074412224]: File loader.c, Line 312 (ast_load_resource):
chan_zap.so: load_module failed, returning -1 Segmentation fault
(core dumped)
Did you remember to reload the kernel modules for your specific
hardware?
On Wed, Jan 07, 2004 at 04:28:44PM +0100, Peer Oliver schmidt wrote:
BTW: Nicolas, are you thinking of finishing up your SyncML tool
(http://nicolas.bougues.net/syncml/)
As of now, my codebase is being used (and further developed) by
various people. But I really did it as a proof of concept,
Using modprobe+ztcfg didn't work.
Kernel patch is ok.
I just re-installed aterisk and now it's working. :(
Anyway, now that my * is remotely placed. My ATAs are behind NAT. (setup
is: private ATA - router+nat - public internet - public*). I configured
my sip.conf to have host=172.30.200.27,
Hi,
I'm trying to intall zaprtc on my machine (I don't have UHCI_USB nor Zaptel)
and I have a strange behaviour during loading.
When I do make load
I have :
./zaprtc.o: init_module: Input/ouput error
With a quick dmesg got:
rtc: I/O port 112 in not free
Is someone have any idea to solve this ?
--On Wednesday, January 7, 2004 5:24 pm + WipeOut
[EMAIL PROTECTED] wrote:
The GS phones have a setting for Voice Frames per TX with a default
value of 10.. This causes the phone to use a 100ms packet size and
Asterisk is set to use a 20ms pachet size.. The result is a choppy sound
when
Terence, Thank you for sharing your thoughts on our judicial system. I
am glad you are there and I am here.
(i'm not under jurisdiction of a ridiculous judicial system)
Anyway, I work in the 911 arena and in the US many states mandate that
you have E911 (identify the persons location and call
mattf wrote:
Is Asterisk's high memory usage canceling out most of the performance gained
by using HT?
In my experience, I've found that hyperthreading is a major problem on
Linux Kernel 2.4. If I turn on hyperthreading and start to load up an
Asterisk box we get ratty sounding audio
Pardon me if this is a faq--I could not find the answer.
I am trying to build an asterisk-based solution with approximately 200
users.
I have a 2.8 GHz P4 with 512 MB RAM.
Any comments re this being sufficient/insufficient?
Any pointers to sizing guidelines will be really appreciated.
Thanks
Hey Steven,
Sorry to bother you yet again w/ a question on my seemingly endless
quest to get DID trunks setup for a customer.
If you don't know anything about this issue or would rather I looked
elsewhere (including the Asterisk list, I suppose), please just let me
know right off the bat.
Peter Hudec wrote:
my debug is
./asterisk -vcg
extension.conf is OK becaouce form ATA186 calls are working, aleso
from cisco7905 and cisco as5300. And of course insde * it is working to.
It doesn't matter, that the first is WAIT(1). It crashes any way.
If it crashed you should send
he... this is a kernel question, not * one :)
btw, making make menuconfig in kernel src
dir, will bring a nice ncurses menu, where you will
find RTC support (don't rember now if under devices
or char devices... search it)
then disable RTC support, rebuild install kernel
and you're done.
most
On Wed, 2004-01-07 at 12:46, Yannick DESSERTENNE wrote:
Hi,
I'm trying to intall zaprtc on my machine (I don't have UHCI_USB nor Zaptel)
and I have a strange behaviour during loading.
When I do make load
I have :
./zaprtc.o: init_module: Input/ouput error
With a quick dmesg got:
rtc:
Hello
Is there any way to limit amount of disk space available per user for the voicemail.
Regards
Tony
I've been looking at Asterisk for a replacement for our phone system and I'm
hoping someone can help validate my assumptions.
We have 4 analog lines coming into the building. These lines are simple POT
lines and we have them in a hunt group with Verizon so that when a single
phone number is
In extensions.sip, I am using the following within the [sip] context:
exten = _9.,1,Dial(vpb/1/1/${EXTEN:1})
exten = _9.,2,Congestion
If any one can think of any suggestions to address any of these
problems, please let me know - I appreciate any comments received.
Thanks!
Terence
I am the network administrator at a small (20-30 employee) financial
company. We are in the process of moving offices and will be obtaining
a VoIP phone system when we do. Right now, it's down to the 3com nbx100
series and *. Having lurked on *-user for a few weeks and having seen
the nifty
Yea we still found some issues with the line assignment. I think you should
stay with 2.03f or whatever is loaded; I guess we will have a real good
version by the end of the week.
Christian
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf
Test Post
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Javed Ikbal wrote:
Pardon me if this is a faq--I could not find the answer.
It is, but hey, let's go over it again.
I am trying to build an asterisk-based solution with approximately
200 users. I have a 2.8 GHz P4 with 512 MB RAM.
Any comments re this being sufficient/insufficient?
This
Hi John-
I'll try and give you a brief outline of DID signalling:
Many businesses have several incoming telephone numbers used for different
purposes, for example customer service, sales, etc. Some have individual
telephone numbers for each user in the system. In a home setting on the
other
-Original Message-
From: john lawler [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 07, 2004 1:38 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] yet another question on DID trunks
Hey Steven,
Sorry to bother you yet again w/ a question on my seemingly endless
quest to
Hi,
I am trying to connect * E100P directly to Cisco using an ATM circuit.
The ATM circuit is ok (I can connect to Ciscos). I do not understando
too much about Cisco syntax, please help me.
#
Cisco Conf:
isdn switch-type primary-net5
isdn voice-call-failure 0
controller E1 3
tony banks wrote:
Hello
Is there any way to limit amount of disk space available per user for the voicemail.
Regards
Tony
Or easier a way to limit the number of messages a voicemailbox can have??
I just use the script that is in the cvs to delete messages after a
number of days..
Is anyone experiencing random hang-ups while in a call? I'm using Cisco
7940s with Catalyst 2950s, QOS enabled. The PRI channels (Zap/g1) are
connected to a Digium T100P.
In the call, asterisk is issuing a hang-up, not the other end.
example 1:
-- Executing Dial(SIP/Derek-1-5a13,
Peer Oliver schmidt wrote:
Nicolas Bougues wrote:
On Wed, Jan 07, 2004 at 01:42:30PM +, Andreas Anderson wrote:
Hi Guys,
is there a client which can be used on the SonyEricsson P800/P900...?
IAX would be cool, but i take anything that can connect (via bluetooth)
to an asterisk-server ;-).
Steve,
If your E1 card has another port, you might consider investing in a fax
software package that can drive a digital fax board, like a Brooktrout
TR1034 of an Eicon Diva server. Then you're ISDN-PRI end-to-end. These cards
come in fractional E1 port densities, so you could wire only 8 bearer
I have an application that I want to be able to do with asterisk. The
scenario is this:
Caller 1 is a user on asterisk. He/she calls mom dialing her phone number
from his phone. Mom flashes and calls a third person and then bridges
Caller 1 and Caller 3 through her phone. I do not want
On Wed, 2004-01-07 at 13:38, john lawler wrote:
Hey Steven,
Sorry to bother you yet again w/ a question on my seemingly endless
quest to get DID trunks setup for a customer.
If you don't know anything about this issue or would rather I looked
elsewhere (including the Asterisk list, I
M. Matt Colgin wrote:
I've been looking at Asterisk for a replacement for our phone system and I'm
hoping someone can help validate my assumptions.
I'll try.. :)
We have 4 analog lines coming into the building. These lines are simple POT
lines and we have them in a hunt group with Verizon so
See below
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Cees de Groot [EMAIL PROTECTED]:
Jim Flagg [EMAIL PROTECTED] said:
Agreed, Guess I should have said traditional computer. Most PBXs would
only use a hard drive for voice
While you approached the community in a very polite way and all, your
few weeks of *-users list should have told you that most answer should
be able to be found on the wiki.
http://www.voip-info.org/wiki-Asterisk+hardware+recommendations
This has a few listings of working systems.
Also, that
Another concern I have on this front is that it seems like some updates require
an asterisk restart rather than just issueing a reload command from the *
console. This that correct, or I am just not running the system correctly? For
instance it seems like I couldn't get zapata.conf changes to go
On Wed, 2004-01-07 at 14:16, M. Matt Colgin wrote:
Test Post
Ohh noo, I read it. Anyone have the antidote?
--
Steven Critchfield [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Jeffrey Paul wrote:
I am the network administrator at a small (20-30 employee) financial
company. We are in the process of moving offices and will be obtaining
a VoIP phone system when we do. Right now, it's down to the 3com nbx100
series and *. Having lurked on *-user for a few weeks and
I'm still trying to examine a DID solution for this customer but
don't understand how a single trunk (whatever that is--I assumed
just a single pair of wires like a POTS line, but I'm thinking
now it must not be) can support multiple (incoming) phones
calls.
As another attempt tohelp
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Andrew Thompson [EMAIL PROTECTED]:
- Original Message -
From: Jonathan Moore [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, January 06, 2004 9:54 PM
Subject: Re:
In your example 1, it shows the hangup started on the PRI channel. Do
you have busydetect or callprogress set in your
/etc/asterisk/zapata.conf file? If so, comment them out as they are not
necessary for your PRI lines.
Not to rub it in, but that isn't the normal behavior for asterisk. My
PRI
On Wed, 2004-01-07 at 15:27, Todd Wallace wrote:
I have an application that I want to be able to do with asterisk. The
scenario is this:
Caller 1 is a user on asterisk. He/she calls mom dialing her phone number
from his phone. Mom flashes and calls a third person and then bridges
Caller
I really hope that someone can help me with this one.
DTMF tones are not working for certain places that I call, specifically
1-800-882-8880 which is the AA advantage line. It works for almost
everyplace else. If I bypass asterisk, the call works fine.
Network looks like:
SPA-2000 --SIP--
Now imagine this person having his SIP phone in IOWA talking
to the the telephone switch in New York via VPN and dialing
911. The call will go to NYPD.
Why is it the theoretical VoIP user in such examples always seems to be from
Iowa or Nebraska? I feel compelled to state that not all people
Greetings all. I am new to the Asterisk world! Found it very impressive so far!
In relation to the below.
I have worked with Alcatel PBX's for the last 3 years. Alcatel OxE supports SIP and
H323 as well.
As far as SIP goes I have also found the Xlite to be good for soft phones. I am using
one
On Wed, 2004-01-07 at 15:41, Jonathan Moore wrote:
Another concern I have on this front is that it seems like some updates require
an asterisk restart rather than just issueing a reload command from the *
console. This that correct, or I am just not running the system correctly? For
instance
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