Dean Collins wrote:
Yep, POE has turned out to be a real fizzer.
Whilst a great idea for Access Points (particularly ceiling mounted AP's
They are *far* more useful for simplifying phone wiring.
so you don't need to run power points) but apart from that the whole
concept has just died.
Not
TC wrote:
Why not just keep it simple use dial with Macro argument
and this std macro-screen
like this
http://lists.digium.com/pipermail/asterisk-users/2005-March/098257.html
Thank you so much!
I was not familiar with this option since we only run STABLE and this
feature is only available
Is there any other way to connect multiple tables and fields to read and
write in the dialplan? (simple inserts queries).
Perhaps via app_dbodbc or res_sqlite?
Rafal
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.10 -
After I run it, I get the following:
99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.00%
99.987793%
Just for reference, I'm running a PIII-800Mhz and I get (with no
particular load on CPU)
-Best: 100.00 -- Worst: 99.987793
100.00% 100.00% 100.00% 100.00%
I suppose by this you mean some sort of client software installed on
the client PC that listens to events targeted at a particular port
this software is listening to. If this is the case, how do you make
Asterisk communicate with this client software?
I use yac and system() with the nc comand
Date: Sun, 15 May 2005 15:17:53 -0700
From: trixter http://www.0xdecafbad.com; [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 911 Options
To: Ira Burton [EMAIL PROTECTED], Asterisk Users Mailing List -
On Sun, 2005-05-15 at 15:55 -0600, Ira Burton wrote:
I am curious if anybody has
I just registered ID 0004283 at http://bugs.digium.com for the problem
described in subject (found when using a Linksys PAP2-NA). I don't know
where the proper forum is to discuss, so I'm hoping anyone interested will
read the bug and let me know your thoughts, either at bugs.digium.com,
John Daragon wrote:
Yep - down in openh323/src/transports.cxx there's a method
H323TransportAddress::GetIpAndPorts() which is called (eventually) by
MakeCallLocked(). This in turn calls GetPortByService() and
GetHostByAddress().
My guess is that the 60 second wait is caused by a request
OK I see the ponit (although I never said that second page is interrupted -
I said that in some combinations of resolutions and TIFF options receiving
fax spits another blank sheet of paper beside the clearly received first
page).
I have read someware (some faxing tutuorial) that there is some
How do you make yac open a webpage?? Or what are you doing with yac on the
client pc?
Is there any way to configure yac with a diff. skin or something? Or plain
old black small screen is ugly :)
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
Hi Waldo, I would be money on your problem being related to the accuracy
of zttest. One way of checking IRQ's is to run cat /proc/interrupts,
but it is a lot more accurate to run lspci -v and lspci -vb.
I would recommend Googling the lspci command, although the output is
pretty self
Hi,
I'm seeking to change my service provider (after ten months, I've had it
with broadvoice), but I would like to keep my 310 number. I've been
digging through the lists of other providers and am considering telasip
(good plans and support number transfers).
My concern is what precisely happens
Hi,
in my attempt to install ISDN BRI card, I loaded asterisk-addons.
I think I went to fast and buggerd up the locations of the files and
directories.
cant load asterisk again, getting:
[res_config_mysql.so] = (MySQL RealTime Configuration Driver)
asterisk: relocation error:
How do you make yac open a webpage??
Don't know, since I'm not triying to open a webpage
Or what are you doing with yac on the
client pc?
The CID info pops up so someone working on their PC can see who's
calling. Especially nice for people with older phones that don't have
CID at all.
Is
Personally, I'd like to see this changed so there are two 'general'
sections--one for default parameters to use unless overridden when there
*is* a peer section below, and a different one to describe parameters to
use when the remote peer is not previously known. I know there are ways
to
Hi,
From the chan_misdn readme:
Now I use Kernels 2.6.9 and it works perfect. with kernels = 2.6.10
there is a very litle bug in hfc_multi.c which causes the module not to
compile, it can be easyly fixed by changenging pci_findsubsys to
pci_getsubsys in code.
Hope this helps
Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?
When I add a SIP Trunk with my Sipgate settings and use a pattern of
8|. to place all
Hi,
I have installed AreskiCC on Slackware 10.1 with Asterisk
latest CVS and Postgres 7.4. First of all the instructions are very confusing
and hard to follow if you are not an expert. But, I managed to install it
andobviously t doesnt work. The other instructions I found on
wiki are
hello
i am trying to make a callback solution.
client will call callback number and call is
terminated.
now callback server will create a call for that
client.
actually i have a problem in this process. that server
is creating call to client (UA) when previous call is
not disconnected yet.
Hi all
I have a following problem. I want to use sjphone to connect to asterisk sip
server and then I want asterisk to do a conversion to h323 and send this to
h323 gateway.
sjphone---sipASTERISKh323-GATEWAY
Example:
if someone from plane PSTN line dials 123456 the gateway will
Hello all
I am in the process of trying to create a more fault tolerent HW setup
for my asterisk platform, its all going well and I intend to do a
wiki about it once its seen to be working.
One thing gets me, and hopefully someone here can confirm my suspision
- why is zaptel.conf not with the
Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?
When I add a SIP Trunk with my Sipgate settings and use a pattern of
8|. to place all
This is relatively straight forward, you can either use Nufones
Implementation or the OH323 package. Both work relatively well.
However, I've had issues presenting a GateKeeper ID from Asterisk to
carriers that authenticate based on that in the past.
Regards,
Sahil Gupta
VoiceValley
On Mon,
Steve Maroney wrote:
The cheapest I have found was a 3COM 24 Port for $799.00.
Thank you,
Steve Maroney
Be warned, we are a 3Com house, and I ordered a 4400 PWR to test it
would work with our Siemens hard phones. Lucky I did, because it turns
out they are not compatible! It seems the 3Com POE
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=51268item=5774375303
rd=1ssPageName=WDVW#MyDescription
I found these on Ebay, what do you think? They are certainly cheap enough.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
Has anyone got chan_misdn working with passive BRI cards yet? I've tried
both hfc (hfcpci.ko) and w6692 (w6692pci.ko) cards, but when I start
asterisk I get the following when chan_misdn is loaded:
[chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri))
== Parsing
Hello,
I am looking for how to increase the pickup timeout. If a call is not
picked up in 20 seconds asterisk automatically hang it up indicating the
message:
Nobody picked up in 2 ms
How can I increase this timeout?
Thank you very much.
Regards,
Alberto
Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?
When I add a SIP Trunk with my Sipgate settings and use a pattern of
8|. to place all
Im looking for a zaptel type device with one (or more) FXO and one
(or more) FXS port. Basically this guy would sit in-line of your
phone line (PCI card). Any suggestions? TDM400 would be overkill.
Your only choice for zaptel type is the TDM card.
Probably the next best choice
I need to connect up to sixteen phones per building, I can use a cheap hub,
but POE would be useful. Is there a cheap POE hub available? Everything I
have seen has been expensive.
Hope you really meant a cheap switch... you don't want to use hubs
of any sort in the asterisk environment since
I'm not overly familiar with the Skype API. Last I heard the API is
missing the necessary features to make a full client, this was
obviously done on purpose by Skype. I think there are some solutions
to get a third party tool to run along with Skype.
On 5/15/05, Wessel de Roode [EMAIL
In article [EMAIL PROTECTED],
David John Walsh [EMAIL PROTECTED] wrote:
One thing gets me, and hopefully someone here can confirm my suspision
- why is zaptel.conf not with the other asterisk files
(I assume it is because its responsable for bringing up the hardware,
not strictly part of
Single port 3com injectors are really cheap. Like $20 a piece.
Granted no one wants to have a MASS of POE injectors. For small 8
installations it might be manageable though.
I haven't tried them with things outside of my 3com NJack.. I'll have
to test it on the Polycom before I buy more.
On
Hi all,
I am using chan_h323 from Asterisk CVS to interconnect with GNUGK
v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on
Asterisk. However, I only heard ringing when the call was answered on
SIP side. Below is the debug from chan_h323. Any help is welcome.
Thanks.
*CLI ==
Hi Waldo, it really depends on who you ask - Digium say that anything
less than 99.99% is going to result in problems, but ours regularly runs
at around 99.98% and we don't have any problems.
One of our boxes was running at around 99.96% and we had major issues
with the voice quality
I am using Sipgate with [EMAIL PROTECTED] and this is how I have set mine up to
have it working perfectly. Using the AMP Interface my trunk is setup as
follows..
Under Trunk:
Outbound caller ID is your full sip number including area code.
Peer Detail:
allow=ulaw
authuser=539 (your sip
How do you disable hyper threading (what's the command and where is it
placed)?
Hyper-threading is a BIOS feature available on some Pentium 4 Xeon
processors. If you have hyper-threading enabled your system may appear
to have more processors than are physically in the system. Typically
Rich Adamson wrote:
I need to connect up to sixteen phones per building, I can use a cheap hub,
but POE would be useful. Is there a cheap POE hub available? Everything I
have seen has been expensive.
Hope you really meant a cheap switch... you don't want to use hubs
of any sort in the
On 06:37, Mon 16 May 05, Rich Adamson wrote:
Hi Waldo, it really depends on who you ask - Digium say that anything
less than 99.99% is going to result in problems, but ours regularly runs
at around 99.98% and we don't have any problems.
One of our boxes was running at around
Hope you really meant a cheap switch...
Yes, I am going to use 16 port Linksys switches if I can't get POE units at
a reasonable price.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
___
Asterisk-Users mailing list
I have an asterisk server behind NAT - no audio on the test external calls I
have tried making so far.
Read http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - No solution
evident from there, sounds like I have case 9. I would have thought that all I
would have to do is port foward and
The latest version of IPSwitchBoard has been released:
Version 0.116 - 16. may 2005
* Call Data Records can be charted (number of calls and duration of calls by
the hour).
* Hotel Application can now print the calls and charges
* Many minor bug fixes
FREE Download:
Could you please give me an example of such configuration?
Thank you!
Regards
On Monday 16 May 2005 12:28, Sahil Gupta wrote:
This is relatively straight forward, you can either use Nufones
Implementation or the OH323 package. Both work relatively well.
However, I've had issues presenting
I'm experimenting (using for the first time) with using a cisco3620 to
connect to the PSTN via a channelised E1 interface, with * handling all
of the SIP calls.
If anyone has any installation tips / help / documentation I would be
most appreciative :)
However, my first question is this: when
this was posted before:
On 5/12/05, Colin Anderson [EMAIL PROTECTED] wrote:
They instantly got us to look at the output of zttest and we found that
this was (in their words) 'extremely low', with 'best' and 'worst'
readings of 99.975586% and 99.963379% respectively.
Might want
Thanks for getting back to me,
the only reason that I see to move it (and more importantly to move it
to /etc/asterisk)
is that I am intending to use DRDB to make the machines as identical
as possible, and to ensure that the configs of the two machines are
kept in-sync.
My mount points for the 3
I need to connect up to sixteen phones per building, I can use a cheap hub,
but POE would be useful. Is there a cheap POE hub available? Everything I
have seen has been expensive.
Hope you really meant a cheap switch... you don't want to use hubs
of any sort in the asterisk
-- Executing Dial(SIP/201-fcb3, SIP/sipgate/###) in new stack
-- Called sipgate/##
Paul I apreciate why you've the dialled digits out there, but
would you be good enough to include the first few, as if your asterisk
box is sending extra / unwanted / too few digits to
Hope you really meant a cheap switch...
Yes, I am going to use 16 port Linksys switches if I can't get POE units at
a reasonable price.
Looks like the 8-port Netgear FS108P is about $103 to $141 right now, and
it supposedly supports poe.
___
On May 16, 2005, at 14:37, Rich Adamson wrote:
Hi Waldo, it really depends on who you ask - Digium say that anything
less than 99.99% is going to result in problems, but ours
regularly runs
at around 99.98% and we don't have any problems.
One of our boxes was running at around 99.96% and we had
D-Link makes a whole line of them.
http://www.provantage.com/YDLNS046.htm
On 5/15/05, Chris Mason [EMAIL PROTECTED] wrote:
I need to connect up to sixteen phones per building, I can use a cheap hub,
but POE would be useful. Is there a cheap POE hub available? Everything I
have seen has been
I cannot email them, I cannot call them, I do not get an answer, but the
credit card is still charged, although NO phone calls are possible
anymore, ...
Are they still in business? (except charging credit cards)
bye
Ronald
___
Asterisk-Users
The 8 port would only be 7 port after uplink, so even two of them is not
going to give me 16 ports, so they are not suitable, I don't have room for
three devices. Shame.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
-Original Message-
From: [EMAIL
On 13:24, Mon 16 May 05, David John Walsh wrote:
Thanks for getting back to me,
the only reason that I see to move it (and more importantly to move it
to /etc/asterisk)
is that I am intending to use DRDB to make the machines as identical
as possible, and to ensure that the configs of the
Moreover, The FS108P can only power 4 ports simultaneously.
I'd prefer something like this:
http://www.netgear.com/products/details/FSM7326P.php
Or a Cisco equivalent.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chris Mason (Lists)
Sent: 16
For example, if you use an Point-to-Multipoint ISDN connection (not
'Anlagenanschluss'), then you won't get an immediate 'BUSY' on SIP
Busy/Congestion.
It's not possible to signal the caller 'Busy' or 'Reject', because there is
a timeout on the ISDN-Bus for ANY OTHER device which may answer
How about this...
Replace the old text in /usr/src/zaptel/zaptel.conf.sample:
# span=span num,timing,line build out (LBO),framing,coding[,yellow]
#
# The timing parameter determines the selection of primary, secondary, and
On 5/13/05, Kaj J. Niemi [EMAIL PROTECTED] wrote:
How did you get it to compile?
Do you have to have a strictly 64 bit compile environment?
On RHEL4 it compiles just fine out of the box. Some of the locations are
not strictly correct (things get sent to /usr/lib instead of /usr/lib64..)
This is a portion of code out of a callback program I'm using:
if ($response eq 1) {
verbose(CALLBACK: Callback to $clidnumber confirmed.);
$out = new Asterisk::Outgoing;
$out-setvariable(Channel, $channel . $clidnumber);
$out-setvariable(MaxRetries, 1);
$out-setvariable(context, $context);
Your configuration is OK. Cisco is counting from 0, so Serial 0:15 is 16th
channel (D-channel) of first E1 (if you don`t have serial interfaces
also...).
zaptel/asterisk is counting from 1, so 1-16 is D-channel of first E1
interface.
See archive for thread named Asterisk and Cisco AS5300 or
list wrote:
Hi,
in my attempt to install ISDN BRI card, I loaded asterisk-addons.
I think I went to fast and buggerd up the locations of the files and
directories.
cant load asterisk again, getting:
[res_config_mysql.so] = (MySQL RealTime Configuration Driver)
asterisk: relocation error:
On Mon, 16 May 2005, Michael Welter wrote:
Where is the clock source that the T1/E1 board, with 0 for timing,
uses to generate the tx data stream? Is there a PLL on each board? Or
is some central source used?
For example, I have one system with two separate T100P cards--one for a
Good day all
How do i set a connection between 2 asterisk servers via PRI
In Bri I would set one to NT and TE
How shoud the zapata.conf and zaptel.conf look
And how should the cable be?
All I got on the web was to set one to pri_net...this cant be all?
And the cable
pin1 -- pin4 pin2 -- pin5
Peter Svensson wrote:
On Mon, 16 May 2005, Michael Welter wrote:
Where is the clock source that the T1/E1 board, with 0 for timing,
uses to generate the tx data stream? Is there a PLL on each board? Or
is some central source used?
For example, I have one system with two separate T100P
hello
he is still not replying after correct time
this is the sip debug
May 16 21:41:02 WARNING[3902]: chan_sip.c:730
retrans_pkt: Maximum retries exceeded on call
76fa142e2805cc9a5d44ba4564165b1e@ for seqno 102
(Critical Request)
May 16 21:41:02 NOTICE[3902]: pbx_spool.c:234
attempt_thread:
Broadvoice turned out to be a very frustrating and disappointng service. I
gave up on them a few weeks ago and cancelled my account. I signed up with
VoicePulse, but the olny e-mail I received from them stated there were
problems signing me up and to call a number. I have not and probably will
Can we get this looser bumped, this has been happening for the last 2
weeks now.
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:MAILER-
[EMAIL PROTECTED]
Sent: Monday, 16 May 2005 12:02 AM
To: Dean Collins
Subject: failure notice
Hi. This is the qmail-send program at
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Giles Coochey
Sent: Monday, 16 May 2005 8:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] POE hub
Moreover, The FS108P can only power
Does any one have any recommendations on an off-the-shelve PC for an
Asterisk Server? This is for a proof of concept, so it needs to be
inexpensive. I have tried 2 different PC's and had problems with the sound
cards. I am thinking of PC's I can buy from local dealers like Best Buy,
Office Depot.
hello
can any one tell me
Channel: SIP/[EMAIL PROTECTED]:5060
MaxRetries: 1
# Retry in 5 min
RetryTime: 60
WaitTime: 30
Context: default
Extension: 6000
Priority: 1
why this is not working
Discover Yahoo!
Have fun online with music videos, cool games, IM and more. Check
Dean Collins wrote:
Can we get this looser bumped, this has been happening for the last 2
weeks now.
I hate this kind of thing as much as anyone, but isn't bumping him off a
bit extreme? :-)
Regards,
Steve
-Original Message-
From: [EMAIL PROTECTED] [mailto:MAILER-
[EMAIL PROTECTED]
How about this...
Replace the old text in /usr/src/zaptel/zaptel.conf.sample:
# span=span num,timing,line build out
(LBO),framing,coding[,yellow]
#
# The timing parameter determines the selection of
Hi All
I had asterisk running on a xercom install, I upgraded the box to a full
debian install and now asterisk is not starting from on boot. I can start
asterisk from the command line fine no problems, but when i type
/etc/init.d/asterisk start it says asterisk PBX started. It doesn't start it
My 2 cents:
If I dont misunderstand it, I would guess youll have to read it
again and type in myroot AND mypassword or what it says and NOT
your_login and your_password.
After that, you can change it to your preferred password etc...
Thats what I guess...But Im not sure.
Regards
//Stefan
On
oh,
Thank you !!
Problem solved.
Juanjo
On 5/13/05, Patrick M. Gray, Jr. [EMAIL PROTECTED] wrote:
Did you dial the 800 number correctly? You need to dial *1800XXX. I
had this problem for a while and then checked out the docs on FWD's website.
Any toll-free number seems to require a *
I am operations vp for a wholesale VOIP network and we have customers
sending us VOIP over satellite that works quite well.Several well
known carriers just do not work for VOIP in my experience.
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi Josiah
Thanks for the info. What I decided to do instead was to modify my own
macro so I could pass the ring type to it. It may have helped me had I
remembered that the default config comes with a dial macro, but then
probably not, as I rewrite things all the time.
I like to reinvent the
Jeffrey Starin wrote:
Jonathan! You don't know how much that simple explanation has helped me
understand Asterisk. Well done. Well said. And to the point clearly.
I would hope this could find it's way onto the Asterisk Wiki and be the
*first* thing someone reads when looking at the
That's a setting of the BIOS (at least on the motherboard we have).
- Waldo
On May 16, 2005, at 8:37 AM, Rich Adamson wrote:
Hi Waldo, it really depends on who you ask - Digium say that anything
less than 99.99% is going to result in problems, but ours
regularly runs
at around 99.98% and we
Dell's entry level line of servers is very Linux friendly.
I use poweredge for some production systems (yes, even with a single drive)
but if is only for a proof of concept, then a $50 Compaq deskpro which
are also Linux friendly might be an option.
Stephen McAllister wrote:
Does any one have
Ronald Wiplinger wrote:
I cannot email them, I cannot call them, I do not get an answer, but the
credit card is still charged, although NO phone calls are possible
anymore, ...
Hmm. I called them twice yesterday to ask questions, the queue wait was
less than a minute in both cases. First time
This is interesting. Do you also have a TE410P?
- Waldo
On May 16, 2005, at 2:46 AM, Wilson Pickett wrote:
After I run it, I get the following:
99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.00%
99.987793%
Just for reference, I'm running a PIII-800Mhz and I get (with no
particular
On Mon, 2005-05-16 at 22:21 +0800, Steve Underwood wrote:
Dean Collins wrote:
Can we get this looser bumped, this has been happening for the last 2
weeks now.
I hate this kind of thing as much as anyone, but isn't bumping him off a
bit extreme? :-)
The account doesnt exist, he cant
Thanks. That gives me something to work on.
- Waldo
On May 16, 2005, at 4:59 AM, Damian Funnell wrote:
Hi Waldo, I would be money on your problem being related to the
accuracy of zttest. One way of checking IRQ's is to run cat /proc/
interrupts, but it is a lot more accurate to run lspci -v
http://www.pbs.org/cringely/pulpit/pulpit20050512.html
interesting comment this week about the Xbox any intelligent
thoughts here?
I know the price point puts it above most users Asterisk
outlay (I run mine on a $100 P3 -800)
But interesting to see what happens if people start
Rich Adamson wrote:
Im looking for a zaptel type device with one (or more) FXO and one
(or more) FXS port. Basically this guy would sit in-line of your
phone line (PCI card). Any suggestions? TDM400 would be overkill.
Your only choice for zaptel type is the TDM card.
Probably the
Hi,
I'd be interested in comments from any users of the vonage service in the UK?
http://www.vonage.co.uk is the website.
Where are the servers located, traceroute would be useful.
What is the general reliability like?
Thanks
Mike
___
Asterisk-Users
On Sun, 15 May 2005 23:37:28 -0500, Jon Gabrielson wrote:
On Sunday 15 May 2005 09:53 pm, Paul wrote:
Do you have the clout to get a handytone for evaluation and not have
salespeople calling you every day to ask how it's going? :)
Why not just buy one? You can buy one for less than $100 and
Steve Underwood wrote:
I hate this kind of thing as much as anyone, but isn't bumping him
off a bit extreme? :-)
Hell no. Its a permanent error. It won't go away. Plus, this wastes
digium's server time having to send back all the bounces.
Bounce him off the list. Most mailing list
On Mon, 16 May 2005, Rich Adamson wrote:
It doesn't make any difference. The pcm data that arrives from the telco
is buffered in the zaptel and/or asterisk code, and sent out the second
T1 card as soon as it can. That buffering reduces (or eliminates) the
need to sync one T1 card to another.
Lol - yeh and at $1300 I prefer some power plugs.
That's how I feel
Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Thanks for that - I've managed to configure the cisco box following
various examples on the web, but come stuck at the following:
dial-peer voice 100 pots
application session
max-conn 30
destination-pattern 0.
translate-outgoing called 1
no digit-strip
direct-inward-dial
port 0:D
trixter http://www.0xdecafbad.com wrote:
How do you create them?
There is a race condition with asterisk and the spool where if you
create the file or copy it into the queue directory asterisk tries to
read and parse the file before you have finished writing it. A
suggested method instead is to
Stephen McAllister wrote:
Does any one have any recommendations on an off-the-shelve PC for an
Asterisk Server? This is for a proof of concept, so it needs to be
inexpensive. I have tried 2 different PC's and had problems with the sound
cards. I am thinking of PC's I can buy from local dealers
Calling him a loser is a bit extreme. Maybe they fired him but he got a
job that pays twice as much.
Anyway, bumping him is not extreme at all. IIRC - some lists are setup
to automatically unsubscribe people after N days of delivery failures.
We only see this individually when we post but the
Michael Manousos wrote:
Alistair Cunningham wrote:
I have a customer who wants to do large volumes of H.323 to H.323
hairpinning. We haven't tested this scenario for large volumes before;
maybe someone on asterisk-users has.
If they buy a top of the line PC, how many concurrent calls are we
If you shop that netgear option, you can get it under $1000, plus its
managed so you can do things like VLANs and QoS which could come in
handy.
Another upside is that the Netgear will autodetect Cicso PoE vs. IEEE
PoE (espeically important to me because I have a mix of 7900 phones
and IEEE
Hi all,
Is anybody else experiencing problems with voicepulse? Today and over
the weekend? I've had problems with both gateways, but one usually
works when the other doesn't. I'm trying to eliminate my network from
the problem.
Sean
___
Well, I dont know the model numberr of the 3com poe hub that I used but it
worked just fine with the polycom ip phones.
Thank you,
Steve Maroney
On Mon, 16 May 2005, Chris Hills wrote:
Steve Maroney wrote:
The cheapest I have found was a 3COM 24 Port for $799.00.
Thank you,
Steve Maroney
On Mon, May 16, 2005 at 03:51:28PM +0100, Mike Dent wrote:
Hi,
I'd be interested in comments from any users of the vonage service in the UK?
http://www.vonage.co.uk is the website.
Where are the servers located, traceroute would be useful.
What is the general reliability like?
No idea re
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