I have been looking around some and I can't seem to find anything which will
answer my question. If I have two Asterisk boxes in different locations which
are linked to each other over the internet, can I configure the boxes to use
each other's lines as local?
In other words, let's say Site A
[EMAIL PROTECTED] wrote:
Yes you are correct... by default asterisk will send the call to priority
N+101... what is your point?
You asked about turning off call waiting.
In the example that I provided,
if the amount of active calls is 1 then
it will forward to VM without
dialing the
What is the best way to include a whole group of exchanges into a dial
plan? I want to route local toll free by exchange (first three) and I will
have a bunch. Can they be stored somewhere and compared as a group to that
position in the dialplan?
Doug
Hi all,
Anyone can give me a reference for setting the asterisk clustering?
Thanks,
unplug
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I am using [EMAIL PROTECTED] v 2.6
I want to active or deactivate voicemail from command line
Like database put AMPUSER/VOICEMAIL/111 ENABLE this command is applicable at
version 2.8 but it don't work at 2.6
Any one can help me ??
Regards
*
No
Hi! I' ve this problem:I've 2 asterisks box, asterisk11 and asterisk12, and one wi_fi phone.I call from wi_fi to a X-lite phone on a windows xp.I've setuped the X-lite to my vrrp IP(vrid IP) and the call is ok, I call from the wi_fi to X-lite and from the X-lite to wi_fi.
In asterisk panell is all
The call is rejected by Asterisk, so it looks like your dialplan
has no rule for accepting calls to '99546476'.
Armin
On Mon, 5 Jun 2006, Esteban Guana-Jarrin wrote:
I have a problem receving calls via the ISDN line, using the followin
components
Asterisk 1.0.9 with [EMAIL PROTECTED]
On Sun, 2006-06-04 at 02:02 -0500, Erick Perez wrote:
[snip]
CFLAGS = -pipe -Wall -Wmissing-prototypes -Wmissing-declarations
$(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE
#CFLAGS += -O2
#CFLAGS += -O3 -march=pentium3 -msse -mfpmath=sse,387 -ffast-math
# PERF: below is 10% faster than -O2 or
Hi Calvis,
Its good if I can help you in any why with this project.
thanks
../ArunOn 6/2/06, calvis [EMAIL PROTECTED] wrote:
Has anyone done any integration with Asterisk Microsoft Dynamics CRM?Ijust wanted to check with the list before I pursue a project with the aboveintegration.In addition,
On Sun, 2006-06-04 at 17:46 +0800, Ronald Wiplinger wrote:
*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# ##
Attended Transfer *2
One
On Sun, 2006-06-04 at 08:49 -0400, Matt Florell wrote:
I kind of assume from all of the mentions of speex in the code that it
is required.
Reading the Makefile posted by Erick Perez in an earlier posting I see:
# 0 = OFF 1 = astdsp 2 = speex
SILDET := 2
I guess if you specify 0 or 1 you don't
Hi
For whatever reason we've getting 2 or 3 CDR lines logged for each call,
often in different formats:
as1:~# grep test-89-1e2c /var/log/asterisk/cdr-csv/*.csv
/var/log/asterisk/cdr-csv/67.csv:67,88,89,test-context,88,SIP/test-88-2dae,SIP/test-89-1e2c,Dial,SIP/test-89|20,2006-06-05
Stephen Bosch wrote:
Hi:
I'm using a TDM-400 to terminate PSTN lines at my Asterisk server with
kewlstart signalling.
When an outside caller calls the server, the TDM-400 goes off-hook and
provides a ringing tone to the caller. If the caller hangs up before the
receiving party answers the
Yes... it is very easy to do...
; on box a
exten = _NXXNXX,1,DIal(IAX2/boxb/${EXTEN})
;on box b
exten = _NXXNXX,1,Dial(IAX2/boxa/${EXTEN})
you just need to make sure that the context on the each side will have a
match for passing in ${EXTEN} to the other side
[from-boxa]
exten =
Andrew D Kirch wrote:
I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N
Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x
quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is
any registerable incoming volume from these lines.
Thanks Armin
The call is rejected by Asterisk, so it looks like your dialplan
has no rule for accepting calls to '99546476'.
Armin
My dial plan as shown below is,
[capi-in]
exten = s,1,Dial(Sip/123,20)
exten = s,2,Voicemail(123)
exten = s,3,Hangup
I believe I should be able to receive
On Mon, 5 Jun 2006, Esteban Guana-Jarrin wrote:
Thanks Armin
The call is rejected by Asterisk, so it looks like your dialplan
has no rule for accepting calls to '99546476'.
Armin
My dial plan as shown below is,
[capi-in]
exten = s,1,Dial(Sip/123,20)
exten =
My dial plan as shown below is,
[capi-in]
exten = s,1,Dial(Sip/123,20)
exten = s,2,Voicemail(123)
exten = s,3,Hangup
I believe I should be able to receive calls with the above.
With immediate = yes then you should.
I have also tried the following, and i get the same problem and debug
My apologies, I didn't realize I was speaking to someone else.
As far as I know the dialplan does not need to have the j option to do N+101. I'm using 1.2.7.1 without the j option and it jumps fine.
I suppose that will work fine as long as you turn off call waiting on the phone itself. I
On Saturday 03 June 2006 14:06, Rick Smith wrote:
The call gets made, I leave a voicemail, or complete the call in some
manner, and the other side hangs up. I hear a busy signal on the phone
on my end.
I'll bet a donut it's not a busy signal but rather a fast busy which is
known as a
- Gil Kloepfer [EMAIL PROTECTED] wrote:
It would seem that the right behavior would be one of consistency --
if
someone specifies the callerid= option in any of the channel .conf
files,
then it should either set or not set ANI, but not behave differently
for
different channels.
On Saturday 03 June 2006 02:47, Michiel van Baak wrote:
I use subversion for this. Every server has its own branch.
There's also a branch called 'common'
All the server specific branches are svn-copied and svnmerge
init from this branche.
Then the svn automerge thingie Kevin wrote for the
- Osama Kamal [EMAIL PROTECTED] wrote:
I am running asterisk behind nat, and 2 sip phones on 2 different adsl
neted connections, asterisk is staying always in rtp media path, while
canreinvite=yes is configured in both extensions. I need asterisk to
stay away from the rtp media path, what
- mustardman29 [EMAIL PROTECTED] wrote:
I know that Digium and FreePBX were not recommending it awhile back
but I
think that was based on 2.4 Kernel and Digium hardware issues. I am
Can you give me a pointer to any place where Digium recommended against using
hardware RAID cards? I
On Saturday 03 June 2006 16:57, trixter aka Bret McDanel wrote:
but $10 only gets you one license, what if you are vonage sized and need
to support a million customers? What if you accept that you can settle
If you are Vonage and need to support a million customers I will bet you are
not
- Mark Drayton [EMAIL PROTECTED] wrote:
How can I configure asterisk not to log to accountcode.csv at all
and
only log the 18-field line (ie, with uniqueid and userfield) to
Master.csv? I just want everything in one file, one line per record.
You have both cdr_csv and cdr_custom
On Mon, 5 Jun 2006, Andrew Kohlsmith wrote:
I'll bet a donut it's not a busy signal but rather a fast busy which is
known as a congestion signal.
I'll be a jelly filled donut that it's the device he's using and not
asterisk sending the signal :) We have a few ATA's that don't
automatically
- William Piper [EMAIL PROTECTED] wrote:
My apologies, I didn't realize I was speaking to someone else.
As far as I know the dialplan does not need to have the j option to
do N+101. I'm using 1.2.7.1 without the j option and it jumps fine.
This is true in Asterisk 1.2.x, as the default
I
need help with collet call
This my case:
Line PSTN -- ASTERISK MFC/R2 -- PBX
when receive call in the PSTN the asterisk send this call to PBX, but if PBX
its
enabled block collect call, the ASTERISK hang up call;
this block call its not category 8 or
On Saturday 03 June 2006 04:05, Sahil Gupta wrote:
We recently had around 60-80 licenses become useless because Digium
refused to renew the keys on that. That was a bit of money kissed
goodbye.
Ok, that's a great fairy tale. Now tell us the true story.
When you buy licenses from Digium, you
If you need to do a couple differing operations on a list of many
area/country codes, then you may consider using the database to let the dial
plan choose what to do, rather than go through so many extensions.
I mention this to keep your extensions.conf easier to read, not because I
know whether
does thia apply on SIP only or also IAX?On 6/5/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- Osama Kamal [EMAIL PROTECTED] wrote:
I am running asterisk behind nat, and 2 sip phones on 2 different adsl neted connections, asterisk is staying always in rtp media path, while canreinvite=yes is
Hey Doug,
Few things you can do. First off, are the numbers for incoming callers
or for when you are making a call? One way that we do it because our
numbers change a lot is I have a text file with all the numbers on it.
Like below:
[localtoolexchange]
exten = _342, 1, Goto(whereever)
etc..
On 09:41, Mon 05 Jun 06, Andrew Kohlsmith wrote:
On Saturday 03 June 2006 02:47, Michiel van Baak wrote:
I use subversion for this. Every server has its own branch.
There's also a branch called 'common'
All the server specific branches are svn-copied and svnmerge
init from this branche.
On 6/5/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
This is true in Asterisk 1.2.x, as the default in the code is to enable jumping (but the default in the sample
extensions.conf file is to have jumping turned off). In Asterisk 1.4 the default in the code will be to have jumping disabled, and
Kevin P. Fleming
[EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
05/06/2006 14:48
Please respond to
Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
To
Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
cc
Hello everybody.I will apreciate your help in this case.I have the environment showed in the picture of the file included in this message.I would like to send trhough internet all the messages received by voicemail. I made all the configuration through the AMP, and all messages are saved properly,
Stephen Bosch wrote:
Jeremy McNamara wrote:
Stephen Bosch wrote:
I don't know. I'll have to check. Is that a requirement?
Yes - Most absolutely.
http://www.digium.com/en/products/hardware/tdm400p.php
I've confirmed that the board supports PCI 2.2.
I've also
So honestly now, Sahil, what did you guys do that was so different? It
*really* pisses me off when people like you give a half-assed, half-baked
digium sucks post. If you've got an honest beef with Digium, then sure,
lay it all out, but don't present half the fucking story and then bitch
Hi,
I also remember reading that.. but i'm not sure if it was Digium's
word ;) It had to do with some SCSI and SATA controllers taking
control of the PCI bus for too much time, and causing frame-slips or
IRQ losses on TDM hardware.
Julian.
On 6/5/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Andrew Kohlsmith wrote:
Nonsense. The license fee that Digium charges is for onesie-twosie stuff. If
you're making a real go of this as a business you will be paying that patent
license fee either through Digium (if you're transcoding on PCs, in which
case you are either doing something
Yrving,How have you verified that your SMTP is working? Can you actually send a message from your Asterisk server to one of your internal addresses?AlexOn 6/5/06,
yrving rivas [EMAIL PROTECTED] wrote:
Hello everybody.I will apreciate your help in this case.I have the environment showed in the
Many ISPs block outbound SMTP except directly to their mail servers. If this is the case, you could try using a mailhop service such as one provided by dnydns.org.
On 6/5/06, yrving rivas [EMAIL PROTECTED] wrote:
Hello everybody.I will apreciate your help in this case.I have the environment showed
Welcome to our world. You will find yourself up nights thinking of all the possibilities and kicking yourself all day because you can't get them all done :)
Good luck.
On 4 Jun 2006 06:02:39 -, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hello everyone.I had heard about this open-source PBX
On Mon, 2006-06-05 at 09:49 -0400, Andrew Kohlsmith wrote:
On Saturday 03 June 2006 16:57, trixter aka Bret McDanel wrote:
but $10 only gets you one license, what if you are vonage sized and need
to support a million customers? What if you accept that you can settle
If you are Vonage and
On 6/3/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- Sahil Gupta [EMAIL PROTECTED] wrote:
We recently had around 60-80 licenses become useless because Digium
refused to renew the keys on that. That was a bit of money kissed
goodbye.
Unless you had been clearly abusing the key
Talk to digium about this on [EMAIL PROTECTED], they might be able to
help you out there.
Zoa
Chris Mason (Lists) wrote:
I have no problem with paying Digium the $10 for G729 licenses,
everyone has to make money. It's the administration of the licenses
that sucks. I experiment with
On Mon, 2006-06-05 at 10:00 -0400, Andrew Kohlsmith wrote:
On Saturday 03 June 2006 04:05, Sahil Gupta wrote:
We recently had around 60-80 licenses become useless because Digium
refused to renew the keys on that. That was a bit of money kissed
goodbye.
Ok, that's a great fairy tale.
Yes. Jon is correct.
$agi[str_replace(agi_,,$s[[0])] = trim($s[[1]);
This line needs some work... Your brackets are mismatched.
On 6/2/06, Jon Farmer [EMAIL PROTECTED] wrote:
Yes you have a parse error in your PHP when I saved it locally and run it from the command line I got
syntax error,
On Mon, 2006-06-05 at 10:46 -0400, Paul wrote:
I really doubt that Digium would insist on the $10 fee for a quantity buyer.
no they do give some discount for quantity, people have mentioned that
when they bought a bunch. However I think they said it was close to
$8/license for 672 channels.
Hi
I am trying to use Asterisk as a backend to send and receive faxes over
analog channels connected to a Siemens HiPath 3550 switch and a TDM400P
card.
Receiving faxes, including multipage ones, works really fine, we have no
issues at all. But when it comes to send faxes using the app_txfax
On Monday 05 June 2006 10:46, Paul wrote:
I really doubt that Digium would insist on the $10 fee for a quantity
buyer.
I have no idea (I do not work for Digium) but if you want to buy quantity
g.729 codecs I'd be strongly looking at hardcore NAS equipment to do it for
you. The PC only has so
Hi,
I couldn't quite understand what was so wrong if someone was moving a bit
of hardware around and requested key changes. After all, the keys have
been paid for and the registered person was requesting for the keys to be
reset.
It was a while back... All good otherwise.
Regards,
Sahil
Hi,
I am trying to set up a dial plan und I have a few problems to realise some
functions.
The dial plan should look like this:
123,1,Answer()
123,2,Queue(1stlevel,t)
123,3,Queue(2ndlevel,t)
123,4,Queue(3rdlevel,t)
123,5,Hangup()
If a member of the 1stlevel-Queue can answer the call it should
On Monday 05 June 2006 11:03, trixter aka Bret McDanel wrote:
Again, 10k channels you'll have a half dozen MaxTNT boxes terminating
DS3s. Your fixed costs will already be significantly higher and that
little $10 license fee is included in that.
Its not $10, which also goes along with
David:thanks for your answer. I´m afraid is not the case.I can tell you this because I have an email server inside my network and it works fine.I can send emails from the [EMAIL PROTECTED] to itself (the root) but if I try to send an email to internet it gives me a time out.Thanks
I use it all the time! now I can redirect calls to cell phones with no
problems! I´ve tried a
SE T637 and a Motorola V3 and they worked really well. I´m still trying to
receive calls from the
asterisk server to get into it from my cell phone, but I still can´t.. so
I´m just able to
call other
Alex:I verified my SMTP with "telnet ([EMAIL PROTECTED] ip) 25" and sent an email to root.I have another mail server inside my network and, as happens trying to send an email to internet, it gives me a time out.I can give you all the details if you would.Thanks for your help.yrvingAlex Robar
On Monday 05 June 2006 11:12, trixter aka Bret McDanel wrote:
Well it may be it maybe not, I wouldnt call people liars without proof
however. What I will do is state that I have written a tool that allows
Fair enough.
That may be, which goes with what he said that you said was a faery
tale.
Hi all,
I have some problems configuring the right behaviour of Zaptel devices
in asterisk. Especially with three way call, call waiting and call
transfers.
Having one party on hold and the other on line I would like to have
- Flash Hook + 3 to activate three way call
- Flash Hook + 2 to swap
Asterisk support the concept of configuration engine, this means
that you can write a configuration engine to get the data from
anywhere. The default configuration engine is text_file_engine, that
reads the configuration from text files. This engine does not have any
limit in the code, so the
Have anyone experienced mixed meetme conferences?
Im running a 12 seat call center outbound only. Asterisk 1.2.8,
SIP/ulaw at the phones, SIP/ulaw to the SIP terminator.
Machine: pentium Dual core 2.0ghz (533 fsb) , 1 GB RAM (533mhz ddr) ,
2 SATA 80GB DISK in RAID 0 via software RAID driver. Two
On Mon, 2006-06-05 at 12:01 -0400, Andrew Kohlsmith wrote:
On Monday 05 June 2006 11:03, trixter aka Bret McDanel wrote:
Again, 10k channels you'll have a half dozen MaxTNT boxes terminating
DS3s. Your fixed costs will already be significantly higher and that
little $10 license fee is
Have you checked the validity of the mailcmd setting in voicemail.conf? Make sure that the command supplied there is the exact location to the application that will process your mail.
ie. mailcmd=/usr/sbin/sendmail -t
Also, make sure that the variable emailbody doesn't have something funky
On Mon, 2006-06-05 at 11:24 -0500, Moises Silva wrote:
Asterisk support the concept of configuration engine, this means
that you can write a configuration engine to get the data from
anywhere. The default configuration engine is text_file_engine, that
reads the configuration from text files.
I thought I saw a guide at voip-info that described how to set up and asterisk
to run in a chrooted environment. Now, I can't seem to find it. Anyone know
where such a guide may be?
Doug
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Andrew Kohlsmith wrote:
On Monday 05 June 2006 11:03, trixter aka Bret McDanel wrote:
Again, 10k channels you'll have a half dozen MaxTNT boxes terminating
DS3s. Your fixed costs will already be significantly higher and that
little $10 license fee is included in that.
Its not $10,
On Monday 05 June 2006 12:29, trixter aka Bret McDanel wrote:
you win, let it go
I'm not looking to win anything here.
I got into this thread because of Brian and Lee's dialogue on Friday. What
set me off was Lee's yes, be a good colonist and don't dump any more tea
into the harbour.
You
Hi List,
After quite a bit of struggle, it looks like I'm all ready to roll out
prepaid cards on my small island. I now have a 4 E1s with a bit of spare
capacity in order to accept incoming calls, and I can route Reunion
Island mobile and fix through my own installations.
For all other
If by database you are referring to an external database, such as MySQL, you
have to address failover, redundancy and performance issues if you go in that
direction.
-Original Message-
From: Moises Silva [mailto:[EMAIL PROTECTED]
Sent: Monday, June 05, 2006 10:24 AM
To: Asterisk
- Osama Kamal [EMAIL PROTECTED] wrote:
does thia apply on SIP only or also IAX?
It has nothing to do with the VOIP protocol, it is the nature of how NAT and
port translation works.
--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.
___
- William Piper [EMAIL PROTECTED] wrote:
By gone forever in 1.6... do you mean that even the j in the dial
plan won't work either? Will it just go to the next priority in the
event of a congested or busy signal?
That is correct. All the 'j' options will go away, in favor of
I don't think you can do what you want. The Zap custom calling features
work very much like Centrex service in the USA.
Henry Margies wrote:
Hi all,
I have some problems configuring the right behaviour of Zaptel devices
in asterisk. Especially with three way call, call waiting and call
-Original Message-
From: Michiel van Baak [mailto:[EMAIL PROTECTED]
Sent: Monday, June 05, 2006 8:03 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Config Revision Control
On 09:41, Mon 05 Jun 06, Andrew Kohlsmith wrote:
On Saturday 03 June 2006 02:47,
- Mark Drayton [EMAIL PROTECTED] wrote:
Okay. Which one writes the 18-field line (with uniqueid and
userfield)?cdr_custom.conf has fields for these two but the wiki docs
also say that cdr_csv will write uniqueid and userfield if configured.
Can I just unload whichever one I don't need to
On Mon, 2006-06-05 at 10:44 -0600, Douglas Garstang wrote:
I thought I saw a guide at voip-info that described how to set up and
asterisk to run in a chrooted environment. Now, I can't seem to find it.
Anyone know where such a guide may be?
http://www.voip-info.org/wiki-Asterisk+non-root
Stephen Bosch wrote:
Hi:
I'm using a TDM-400 to terminate PSTN lines at my Asterisk server with
kewlstart signalling.
When an outside caller calls the server, the TDM-400 goes off-hook and
provides a ringing tone to the caller. If the caller hangs up before the
receiving party answers the
Girls, girls, you're both pretty...
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- Henry Margies [EMAIL PROTECTED] wrote:
I know that some of these features are configured in features.conf,
but
always totally without the use of flash hook.
No, flash-hook features are handled by chan_zap itself.
Is there any way to configure this behaviour in asterisk?
Not without
- Jon Lewis [EMAIL PROTECTED] wrote:
IMO, locking the licensing to a piece of system thats often built-in,
has
been very annoying. I think I'd be happier if it was locked to some
sort
of dongle (parallel, or more likely today, USB). At least that way,
we
could easily move the key
What part of NON-COMMERCIAL do you not understand?
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It would help if you included some more information, maybe like the
output from show channels concise from Asterisk and then a summary
of which channels and/or meetmes are mixing audio.
I have not run into any issues with audio from one meetme bleeding
into another, but since you mention meetme
Hi Folks,
I'm trying to test out Asterisk overall.
I'm having some problems with DTMF. Currently I'm playing with DISA,
but I'm worried this will happen when I get to implementing AAs etc.
I have a free SIP trunk from IPKall that I'm trying to make work.
I'm able to receive calls, and I've
While I was playing with svn, it was driving me nuts. It would ALWAYS re-create
the current directory, even if I said to check out all files from inside that
directory. Means if you went to /etc/asterisk and checked out asterisk, you'd
get /etc/asterisk/asterisk. Yuk.
Doug.
Ahem.
cd
Hi
Is there an easy way (without writing a C app) to make asterisk call 2
numbers and then bridge them into one conversatoin (preferably without
using meetme).
Regards
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Thanks Patrick, but thats for non-root Asterisk, not chroot Asterisk.
Doug
-Original Message-
From: Patrick [mailto:[EMAIL PROTECTED]
Sent: Monday, June 05, 2006 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk chroot
What are the reasons that people/companies/manufacturers use G729
instead of comperable codecs like GSM or Speex?
Microsoft and Apple both support GSM in their software, and Speex is
the same compression ratio as G729 yet is BSD-like licensed so no cost
whatsoever.
MATT---
On Monday 05 June 2006 13:03, Martin Joseph wrote:
Girls, girls, you're both pretty...
But I'm prettier, right? :-)
-A.
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Greetings,
I'm looking to purchase 10 Cisco 186 ATAs. Please send me quantities available, payment methodsand out the door pricing (shipping +tax +unit costs).
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To
On Monday 05 June 2006 13:05, Kevin P. Fleming wrote:
I have proposed that a number of times internally, only to be told
(vehemently) that customers would never go for it. That includes responses
from our distributors and channel partners, among others. It would also
dramatically increase the
svn trunk 31497
For the life of me, I can't get this :) I want to be able to catch the
situation where the calling party hangs up *before* the call is
connected to the called party. My dialplan is thus:
macro DialExternal(exten) {
Dial(Zap/G3/${exten},120,g,M(connected));
goto
On Mon, 2006-06-05 at 12:05 -0500, Kevin P. Fleming wrote:
I have proposed that a number of times internally, only to be told
(vehemently) that customers would never go for it. That includes responses
from our distributors and channel partners, among others. It would also
dramatically
On Mon, 2006-06-05 at 13:47 -0400, Matt Florell wrote:
What are the reasons that people/companies/manufacturers use G729
instead of comperable codecs like GSM or Speex?
Microsoft and Apple both support GSM in their software, and Speex is
the same compression ratio as G729 yet is BSD-like
Yrving,You can send to a local user on the system, but can you send to an external account? AlexOn 6/5/06, yrving rivas
[EMAIL PROTECTED] wrote:Alex:I verified my SMTP with telnet (
[EMAIL PROTECTED] ip) 25 and sent an email to root.I have another mail server inside my network and, as happens
DTMF problems happen at the point where the PSTN call is converted to
VoIP. EXCEPT where you are using inband DTMF and ulaw or alaw codec.
inband DTMF does not work with any other codec.
Mr. Jones wrote:
Hi Folks,
I'm trying to test out Asterisk overall.
I'm having some problems with DTMF.
-Original Message-
From: Michiel van Baak [mailto:[EMAIL PROTECTED]
Sent: Monday, June 05, 2006 8:03 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Config Revision Control
On 09:41, Mon 05 Jun 06, Andrew Kohlsmith wrote:
On Saturday 03 June 2006 02:47,
Kevin P. Fleming wrote:
- Jon Lewis [EMAIL PROTECTED] wrote:
IMO, locking the licensing to a piece of system thats often built-in,
has
been very annoying. I think I'd be happier if it was locked to some
sort
of dongle (parallel, or more likely today, USB). At least that way,
we
James,Please send this type of inquiry ONLY to the Asterisk-biz group, not the non-commercial discussion group.AlexOn 6/5/06, James Ching
[EMAIL PROTECTED] wrote:
Greetings,
I'm looking to purchase 10 Cisco 186 ATAs. Please send me quantities available, payment methodsand out the door pricing
Anyone know how to
direct sip calls in a dial plan to a specific proxy if * is registered with more
than one proxy?
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Lewis:This is what the logs says regarding to de mails on a test I made:Jun 5 14:28:59 DEBUG[27498] app_voicemail.c: Sent mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t'Later will come with an error like this:Date:
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