On Thu, 31 Mar 2005, Steve Underwood wrote:
Eric Bishop wrote:
True. I think Digium's USA bias is clearly demonstrated by their lack
of a BRI ISDN product. Most of the rest of the world use it in
abudnace yet Digium do not see fit to service this market because it
is not big in the US
to me like your connection doesn't have enough throughput for the
data rate you are trying to send.
Perhaps there is bandwidth shaping so that the IAX port doesn't get much
bandwidth? Or, your ISP is well oversubcribed?
Steve
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card.
(PSTN)(old PABX)---===(4 ports asterisk)
Just make sure that hangup-detection on the Voicetronix will be compatible
with whatever it is that the PBX does at the end of a call.
Steve
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- but rather integrated, and (some might
say ;-) ) not the world most perfect SIP implementation.
You can find a bunch of options at
http://www.voip-info.org/tiki-index.php?page=Protocol%20Stacks%20and%20Development%20Services
A number of them are open-source/free (Vovida's is one I know about).
Steve
don't want native bridging, you need to disable it in chan_iax2.c
by undefining BRIDGE_OPTIMIZATION. If you do that, then your box will
probably hear and act on the # transfer request.
Regards,
Steve
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I can tell you that they work fine
alongside both the Junghanns and Sirrix quad-BRI boards. Would you like
to buy a TDM04B ;-)
Regards,
Steve
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only does native bridging when both sides of the bridged call
are IAX. So I don't think your problem is the same.
I think you need to check carefully that the options on the Dial() calls
are right, that your SIP phone and Asterisk are configured with the
matching dtmfmode and so forth.
Steve
On Thu, 14 Apr 2005, Paul Seymour wrote:
Thanks Steve, have done as you suggested and it works perfectly. Would
this be considered a bug since the T or t directive in the dial plan
probably should preclude native bridging if the end result is to prevent
a transfer?
Yeah - I think I'd
CDR records. In my experience, h lines in
your extensions.conf will result in your dialled-number info being lost in
your CDRs and being replaced by h.
Its bad when you notice this weeks after what you thought was a minor
change
Steve
at this and some other
things. But I think I know what's wrong from your description...
Are you using SNOM phones? Go to the advanced setup and turn off the
bridge calls on hangup option.
Regards,
Steve
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occured with that as well. We are getting more
calls now, so it's beginning to be more of an issue.
Hi,
I think you need to use SetCallerPres to allowed.
show application setcallerpres will give you the info you need.
Steve
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this
packet loss?
How can I get the file recorded as I hear in the
handset,ie , with the deteriorated audio?
Hi,
Asterisk just dumps the arriving audio into the Monitor file as it comes -
IE missing packets just disappear. This accounts for the speedp and the
different distortion.
Steve Kann has
else experiencing this?
Paul
DHCP timeouts ??
Didn't somebody say that the IAXy doesn't renew its DHCP lease (ie its a
BOOTP client). In which case, your DHCP server needs to give it an
infinite lease.
Steve
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/RTP.
Steve
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in and not receiving calls :) ?
CVS Asterisk can already do this, using the Asterisk database.
Steve
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eventually realizing
they're no longer logged in and not receiving calls :) ?
CVS Asterisk can already do this, using the Asterisk database.
Steve
Care to give an example?
-Matthew
From configs/queues.conf.sample:
[general]
...
; Persistent Members
;Store each dynamic agent
) Your extensions.conf syntax is messed up.
So they should look like:
exten = 550,1,Dial(IAX2/user1)
exten = 551,1,Dial(IAX2/user2)
Steve
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that someone can vouch for
with the TE410P?
Thanks,
Steve
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) IRQ assigned, though I did
see that the actual IRQ that Linux sees in the end is not the one I
assigned (even with noapic).
Steve
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On Mon, 3 Jan 2005, Chris Modesitt wrote:
I have 3 DL380 G4's in production, only difference that I can tell is that I
am running a 2.4.22 kernel.
Great info - thanks. I'll try a 2.4.22 kernel in the morning.
Steve
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On Mon, 3 Jan 2005, Scott Stingel wrote:
Hi Steve-
My customer has five DL320's and ten DL360's - all running asterisk with
TE410P's with no problems - no 380's though. I'm using Fedora core 1
and 2.4.xxx kernel..
I'm assuming that you've done the correct initialization for the card
vanilla 2.6.10.
Steve
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.
Not me.
Steve
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the DTMF tone in a phone?
Their space in the IAXy memory is very constrained.
Steve
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...
Maybe even if I don't have the hotplug riser I still need this driver...
Steve
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-- Forwarded message --
Date: Tue, 4 Jan 2005 17:12:32 -0800
From: Bill D'Anjou [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: FW: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
Can you post this to the group for me? I've joined I am receiving
e-mails from the
[dialout-telkom]
exten = _0.,1,Dial(telkom...)
[dialout]
include = dialout-telkom
exten = _0[78]2.,1,Dial(vodacom)
exten = _0[78]3.,1,Dial(mtn)
etc
The reason is that Asterisk only follows the include links when it can't
find a match in the current context.
Regards,
Steve Davies
On Wed, 5 Jan 2005, Eric Bishop wrote:
I will certainly try that. Please also let me know your progress..
Didn't help for me.
I also tried removing one processor with no benefit.
So I've now given up.
Steve
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servers it seems.
Steve
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disconnect should get everything internal to
reset and initialize properly on next power up.
I tried it that way, with no difference.
Steve
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change is the raid
controller integrated on the main board. Also, PCI riser is no longer
hot-swap as standard; that is an option.
Steve
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On Fri, 14 Jan 2005, Steve Hanselman wrote:
Has anyone also logged a support call with Digium, it has to be either the
card, Linux or the Zaptel drivers.
Yes of course - we have a call open.
Steve
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compiling the kernel with it turned off -
both to no avail.
Steve
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: 62572633
ERR: 0
Its an HP ML110.
Steve
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. If you
aren't getting interrupts through from the card, then the timing doesn't
work - everything stalls.
So - do cat /proc/interrupts and look for lots of interrupts from the
card. If you aren't getting any, invetigate why not...
Steve
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=peer/friend entry that
matches.
Steve
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a Local channel.
So - when you handle the incoming call, on your intermediary machine,
rather than Dial() the third box, rather dial a Local/ channel that then
dials to 3rd machine in turn.
Then, chan_iax2 will by bridged to the local/ channel, and will dejitter.
Regards,
Steve
looped calls - though I've never
thought through why that should be.
Steve
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of the RJ series connectors, if I'm not
mistaken.
Steve
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that they will kick out the Asterisk system.
I'm unable to tell whether the PRI alarm is a symptom or a cause. I
think its a symptom of a bug, because once down it doesn't recover, but
restart Asterisk and it comes up immediately.
Can anyone help shed some light on the problem?
Thanks,
Steve
it.
Steve
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/mediaxphone.php ;)
But it only works on Windows for now.
Works pretty good on my Mac.
Steve
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PSTN and the
Internet.
GS is configured:
Software V 1.0.4.30
Static IP
SIP Server is Asterisk's IP
SIP user ID is the extension of GS
Authenticate ID as user ID
No pw
Name is Steve
Codex are: PCMU, PCMA, PCMU, G729, G726, G728 - note no 723
723 Rate is 6.3
Silence Suppression is Yes
Voice Frames
On Thursday 15 January 2004 12:03 am, [EMAIL PROTECTED] wrote:
sounds like one of those pesky auto dialers the simpsons make fun of.
It sure does...
--
Steve
__
You actually need to constantly be alert
and willing to handle things, or life
On Thursday 15 January 2004 01:30 am, Chandra wrote:
hi i am not talking about * behind NAT. its * outside NAT and GS inside
NAT.
Why leave a host to defend for itself? At least behind a firewall you got some
layers of protection.
--
Steve
On Monday 09 February 2004 01:20 pm, Steve Kennedy wrote:
Probably a dumb question, but what's the best Linux variant to use to
build/run an Asterisk server.
Hardware is Compaq DL360 with a Widcard 410.
Debian/Fedora Core ?
Steve
Nah, go with good ol' DOS 3.3!
You will not have any
.
Without the ACK, the Dialog is not in acorrect state.
snip
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and willing to handle things, or life
will find a way to get you good!
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kernel that I've used since
before 9.0 came out. For me it's been the thing you have to live with
if you want to use *.
I've used completely different h/w except for the same Digium card
(TDM400P w one port.) Many versions of * too.
--
Steve
On Monday 16 February 2004 08:51 pm, Jamin W. Collins wrote:
Do any of you know of a cost effect device that could be connected to
an Asterisk station port to provide room monitoring? I'm looking to
replace the wireless baby monitor we currently have, since there is
too much interference
On Monday 23 February 2004 12:56 pm, Khalid Yaseen wrote:
Hello,
I am interested in running small busines in telecommunication with
minimum expenses and investment. Can Windows operating be used for
this purpose. Thank you all.
Regards,
Yaseen
Haha, that's funny!
Unless of course you
On Wednesday 25 February 2004 08:26 am, Peer Oliver schmidt wrote:
Good day,
I am in the middle of getting my self some hard phones. Anyone care
to comment on the *voice* quality of the following phones:
Cisco 7960
Siptone II
SNOM
Budgetone
I have seen a few reviews, but none go to deep
On Wednesday 25 February 2004 05:35 pm, Matthew B Marlowe wrote:
How'd you get the # transfer feature working? :)
My transfer button worked out of the box. Then when I upgraded * in
Nov-Dec it stopped working I have to use #.
Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
exten = s,3,Goto,2
exten = s,4,Hangup
exten = 1,1,Voicemail,u100
exten = 2,1,Goto,s|4
exten = i,1,Goto,s|2
exten = t,1,Goto,s|2
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howto/doc version 2! You have cut out a huge chunk of confusion!
This will prompt you with how dial works. Dial will jump to n+101
if n+101 exists and the channels you specify are busy.
exten = 2,1,Hangup
exten = t,1,Goto,s|3
exten = i,1,Goto,s|3
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:
#!/bin/sh
# convert wav recordings to gsm
InFile=$1
sox -V $InFile.wav -w -r 8000 -g $InFile.gsm
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.
Did you installed them also. If you installed them, then add
the path in /etc/ld.so.conf and also run ldconfig to refresh
the path of your libs.
Which files are you talking about?
I assume the .so files which are copied to /usr/local/lib.
Is that not enough?
--
Steve Szmidt
On Wednesday 11 June 2003 08:08 pm, Dylan VanHerpen wrote:
Hi everyone,
I'd like to use Asterisk to build a phonetree (www.phonetree.com) type
of application, like this:
1. Read a text-based name/phonenumber file.
2. Call every number and play a recorded message.
3. If a beep is detected,
Hi,
I just made a fresh install on a new box and at the end I got this message:
make: warning: Clock skew detected. Your build may be incomplete.
I had all the various libs added to a default install of RH 9. Though its
possible that I'm short on developer tools. Any clues anyone?
--
Steve
On Thursday 12 June 2003 08:57 am, you wrote:
I just made a fresh install on a new box and at the end I got this
message: make: warning: Clock skew detected. Your build may be
incomplete.
I had all the various libs added to a default install of RH 9. Though
its possible that I'm
On Thursday 12 June 2003 08:49 am, julian green wrote:
Steve wrote:
Hi,
I just made a fresh install on a new box and at the end I got this
message: make: warning: Clock skew detected. Your build may be
incomplete.
I had all the various libs added to a default install of RH 9. Though
On Thursday 12 June 2003 09:06 am, Steve wrote:
On Thursday 12 June 2003 08:57 am, you wrote:
I just made a fresh install on a new box and at the end I got this
message: make: warning: Clock skew detected. Your build may be
incomplete.
I had all the various libs added
address but it does not wrk on e400 or e800,
so I'm stomped.
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) with no luck. What address should it be using?
On Thursday 12 June 2003 11:03 am, Steve wrote:
(I'm producing a writeup on all the things one run into building an *
box. Which after a couple of boxes is showing up aplenty.)
So this uses the working h/w and config's from another box. All I do
of new messages I used a high quality setting with
'record' and then converted it to gsm and it's just fine.
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http
and readline-devel.
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is harder than office.
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Hi,
I see that there's been some very light discussion on having a standard time
and date stamp in VM. How can I implement it today? (About to offer a
system to a customer but they need the stamp to tell when people called.)
Thanks,
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On Monday 04 August 2003 11:40 pm, Steve Meyers wrote:
Where can I find a good tutorial on how channel banks work? I need to
get a 6 port (or so) channel bank for FXO. I need to find some
information on which ones are supported well under Linux and with
Asterisk, how to configure them, what
for workplace injuries is both contemporary
and not unusual.
The way to handle it is to drop the monitor into the desk in a low angle,
kinda like a keyboard. This allows for one in the desk and one on top.
Looks cool too!
- --
Steve
They that would give up essential liberty for temporary safety deserve
looking for and then start receiving), or create a
call spoolfile and initiate a seperate outbound call.
I think you should evaluate the spandsp fax stuff for yourself, though.
It works nicely for me in my limited testing but does seem to have open
issues with some fax machines.
Steve
, maybe with dynamic addresses). Is TDMoE the thing I have to look at
to set this up? The corresponding wiki article says that I must have a zaptel
interface ?!
You'll use IAX to link your two servers.
Steve
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[EMAIL
, and with the filtering abilities you can be as specific as you
want.
As far as following a conversation it can also follow a network session.
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Steve
They that would give up essential liberty for temporary safety deserve
neither liberty nor safety.
Benjamin Franklin
the tables.
From that you can understand what needs to go into the tables.
- You need to add some cards into the database
It might help to go to your /etc/asterisk/logger.conf and add debug to the
console= line. Run asterisk with -v -g.
Steve
On Thu, 10 Jun 2004, Storm D. J. Petersen wrote:
The biggest thing to consider when you are doing a prepaid system is, what
if the person with the same account in/out calls twice?
A simple check for this is included in the standard app_prepaid.
Steve
Nufone numbers.
You can send as many outbound calls as you like to Nufone - just
Dial(IAX2/...) away.
Set the callerid on the outbound calls the way you want them and that's
what the callee will see.
Steve
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merge the two - or maybe drop one and keep the other
entirely.
Steve
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hopefully will go into
CVS. But I think further tweaking is also desirable.
I see on bugs.digium.com stevek has also submitted some adjustments which
have stimulated discussion.
So check asterisk-dev, check bugs.digium.com and I think we'll get the
jitter buffering right.
Steve
bumping both systems up to current cvs Head, add the statement,
and eval the result.
jitterbuffer=no turns off that dynamic jitter buffer function.
People recommend to turn that off because it doesn't work 100% at the
moment.
Steve
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to create another for this specific
phone.
Like you say, put that phone in its own context which doesn't have the
exten= entries for dialling out.
Steve
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On Tue, 22 Jun 2004, Tony Nichols wrote:
I'm trying to get two * boxes to talk no matter what variation I try
I get No Authority Found and connection refused from 192.168.1.5
I've googled, I've site searched to no avail.
I think you need to match a peer at one end to a user at the
dialing them again with another provider.
Why don't you put that on bugs.digium.com - perhaps someone will implement
it.
Steve
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.
Older versions of chan_capi don't initialise an important timestamp in
audio frames - with the result that capi originated calls forwarded over
IAX will probably end up with no audio.
Steve
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On Fri, 25 Jun 2004, Fletcher Bonds wrote:
1. A general will this work? (vmware linux, same pc as phone, NAT'd
addresses,etc)
You'll probably be the first person to try it. I'd guess that it will
work, but expect call quality to be impacted because of all the extra
scheduling and
...
Regards,
Maron
Well
cat sound1.gsm sound2.gsm sound3.gsm
is easier.
- --
Steve
They that would give up essential liberty for temporary safety deserve
neither liberty nor safety.
Benjamin Franklin
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 06 July 2004 07:53 pm, Steve wrote:
On Tuesday 06 July 2004 03:00 pm, Maron Kristófersson wrote:
Also, I need a Linux tool to splice a series of gsm audio
clips together in order to use one 'get_data' instead of multiple
cat
-Users mailing list
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- --
Steve
They that would give up essential liberty for temporary safety deserve
neither liberty nor
are also agents. The same thing...
- --
Steve
They that would give up essential liberty for temporary safety deserve
neither liberty nor safety.
Benjamin Franklin
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
iD8DBQFA7LlwljK16xgETzkRAnPXAJ4rE7Kr
the box pretty tight with
ongoing tests.
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They that would give up essential liberty for temporary safety deserve
neither liberty nor safety.
Benjamin Franklin
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux
/asterisk-users
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Steve
They that would give up essential liberty for temporary safety deserve
neither liberty nor safety.
Benjamin Franklin
-BEGIN PGP SIGNATURE
that *s jitter handling and
packet-loss-concealment can't be improved - I've been working on that and
I'm still busy).
I'm about to ask for some help in gathering jitter stats from a bunch of
users - perhaps you'd like to help with that.
Steve
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I used to be able to run asterisk -rx 'stop gracefully' on stable.
But now with CVS-HEAD-07/07/04-20:09:43 it's returning:
'Broken pipe'
Any ideas why, or how to fix it?
- --
Steve
They that would give up essential liberty for temporary
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Anyone have any experience with either of these, I 'd appreciate some
feedback? Plus it seems pretty easy to steal a connection with this.
Zyxel Prestige 2000W
WiSIP
thanks,
- --
Steve
They that would give up essential liberty for temporary
but...
(It's for a project, not for serious production).
Well - my UK phones' hook-flash always looks like a loop-disconnect 1 to
my Asterisk box. So it looks like loop-disconnect dialling is detected.
Looking at the code definitely pulse dialling is handled.
Steve
] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve
Sent: Tuesday, July 13, 2004 11:01 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Broken pipe in remote exeute
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Hi,
I used to be able to run asterisk -rx 'stop gracefully
love to be able to do this - basically to be able to have a
context eavesdropping on each side of the call - ordinary pbx.c logic
could match digits heard on the call. What fun could be had!
Steve
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Asterisk-Users mailing list
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http
installation.
Thanks in advance
Simon
The best bet would be to read up on the wiki:
If you got to http://voip-info.org/wiki-Asterisk you'll find a link which says
Asterisk dimensioning. There you can read about real life hardware choices.
The wiki is really your friend.
- --
Steve
They that would
.
You should have at least SP3, or even better, SP4 on Win2k. I believe it
will run on Win9x, but I have not tested it and can make no guarantees.
Steve Besch
The bad part is that starting with SP2 on w2k ms EULA has changed to include
your agreement to let microsoft not only see, what you have
... : )
- --
Steve
They that would give up essential liberty for temporary safety deserve
neither liberty nor safety.
Benjamin Franklin
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