You shouldn't really have any problems with i386 version on the AMD. When
centos moved to 4.3 the x86_64 bit version was a mess with trying to install
packages.
Regards
Mark Brooker
T: 02 4959 8670
M: 0415 846 865
F: 02 9882 0947
E: [EMAIL PROTECTED]
W: http://www.mbit.com.au
-Original M
Hi
I am going to install asterisk on an AMD server, did any one had problems
installing it on an AMD server ?
Regards
Kani
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visi
FIXED.
Found and
fixed the problem.
Teach me to
cut code from AAH 2.8
Issue was..
exten =>
s,1,Set(__SIPADDHEADER=Call-Info: answer-after=0) THIS DOES NOW WORK.
exten =>
s,1,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) THIS DOES.
Strange what
a type can do. Also strange why the other w
Scott Gifford wrote:
Colin Anderson <[EMAIL PROTECTED]> writes:
Why not capture the faxes (in or out) in tiff format, instead of audio
format? Setup your asterisk box to relay faxes!
I think in this case the impact on the client would be much greater if you
can show them a recreati
I'm not getting registration from: iax2.fwdnet.net
--
#Joseph
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To UNSUBSCRIBE or update options visit:
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Hi all.
The SPA941 and friends have pretty multicoloured LEDs, but there doesn't
appear to be any support for SUBSCRIBE/NOTIFY as * as implemented for
extension hinting.
Has anyone managed to get the phone to support this?
Thanks!
--
David Zanetti <[EMAIL PROTECTED]>
Team Leader, Systems Admin
On 5/4/06, Arnar Birgisson <[EMAIL PROTECTED]> wrote:
Hi there,
I'm looking into developing an in-house switchboard application. Does
anyone here know of a way to control a hard-phone from such an
application.
For example, the attendant forwards a call with another one in queue.
Once the first
--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:
> >> If Asterisk has a DNS lookup failure it will
> never
> >> retry that lookup.
> > "Never" meaning until the next "reload" command is
> > issued, or until the next "restart" command is
> issued,
> > or until the next time the OS reboots
works for me as well.
On 5/4/06, kevin ling <[EMAIL PROTECTED]> wrote:
Hi,
But it's seems the auto-answer function work on my spa-941. Have you upgrade
to the latest firmware version?
Regards,
kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Tom Engleward wrote:
--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:
If Asterisk has a DNS lookup failure it will never
retry that lookup.
"Never" meaning until the next "reload" command is
issued, or until the next "restart" command is issued,
or until the next time the OS reboots,
Hi,
But it's seems the auto-answer function work on my spa-941. Have you upgrade
to the latest firmware version?
Regards,
kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Friday, May 05, 2006 9:02 AM
To: Asterisk Users Mailing Lis
Asterisk wrote:
Hello all,
I want to report a BUG with the Linksys SPA94X so it is general
knowledge and that we can all make noise about it so it will get fixed
sooner..
The handsets do not work with the SIP flag to make them AUTO-ANSWER.
(As documented is should)
Ie, you cannot use the
Colin Anderson wrote:
That is not a good metric for call completion. Shitty quality will not be
counted in that metric...
Ah, true dat. However, if quality was crappy believe me my users would let
me know. They are salespeople and wholly intolerant of anything that keeps
them from yipping on t
--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:
> If Asterisk has a DNS lookup failure it will never
> retry that lookup.
"Never" meaning until the next "reload" command is
issued, or until the next "restart" command is issued,
or until the next time the OS reboots, or until the
next ti
You can disable the DND button completly. I think that will get you what
you want.
I don't have a 500/501 handy to find out which button it is, but you can
check in Menu -> Status -> Diagnostics -> Test Hardware -> Keypad
Diagnostics.
It's button 9 on my 600 and this disabled it:
--Derek
>
Steve Prior wrote:
> Rich Adamson wrote:
>
>>
>> At least outbound calls still work, even though they changed IP
>> addresses (and probably colo locations).
>>
>>
>
> Maybe not so much now. I just got a disconnect notice from nufone
> which states that I have a positive balance in my account, but
Tom Engleward wrote:
--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:
Are you specifying the remote Asterisk box by IP or
by hostname. If by
hostname, then specify it by IP. Asterisk's DNS
lookup support has issues.
In the trunk peer details in AMP I'd set "host=" to a
hostname.
>That is not a good metric for call completion. Shitty quality will not be
>counted in that metric...
Ah, true dat. However, if quality was crappy believe me my users would let
me know. They are salespeople and wholly intolerant of anything that keeps
them from yipping on the phone. I also run e
hello
I have an asterisk server with a public IP address and a nat address
like alias. I have 20 sip clients with private IP address. I don't Know
why, sometimes, when I try to call between 2 phones, y see the next
menssage in the astersk console:
Can't find address for host 'XX'
What could hap
Rich Adamson wrote:
At least outbound calls still work, even though they changed IP
addresses (and probably colo locations).
Maybe not so much now. I just got a disconnect notice from nufone
which states that I have a positive balance in my account, but still
need to add money to bring i
Yes I believe this is a 1.0.9 bug unfortunately I can't find a reference to
it in Mantis except for this:
http://bugs.digium.com/view.php?id=5971
I run 1.0.9 and I do all of my MeetMe's through the PRI's. This is stupid,
because we are using 2 X # of users / PRI channels but MeetMe runs solid. I
On Thursday 04 May 2006 18:04, Colin Anderson wrote:
> SELECT count( calldate )
> FROM `cdr`
> WHERE calldate
> BETWEEN '2006-01-01 00:00:01' AND '2006-05-04 15:51:00' AND channel
> LIKE '%IAX2%' AND duration < 1 OR calldate
> BETWEEN '2006-01-01 00:00:01' AND '2006-05-04 15:51:00' AND dstchannel
>
I haven't seen this appear on the list, so I thought I would resend
it...
Sorry for the repost if it did appear before...
- Forwarded message from Michael George -
Date: Wed, 3 May 2006 21:48:09 -0400
From: Michael George
Subject: meetme conference latency degrades...
To: asterisk-user
Thanks for all the help! This should go alot quicker then by hand.On 5/4/06, Chad Osmond <[EMAIL PROTECTED]
> wrote:
You can use my script, based on Chris Mason's script, to do most of what
you want, you can feed it your MAC's and Extensions and it will create the
phones.
Be warned, it's n
>Since January I've passed over 37000 calls through these boxes.
SELECT count( calldate )
FROM `cdr`
WHERE calldate
BETWEEN '2006-01-01 00:00:01' AND '2006-05-04 15:51:00' AND channel
LIKE '%IAX2%' OR calldate
BETWEEN '2006-01-01 00:00:01' AND '2006-05-04 15:51:00' AND dstchannel
LIKE '%IAX2%'
--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:
> Are you specifying the remote Asterisk box by IP or
> by hostname. If by
> hostname, then specify it by IP. Asterisk's DNS
> lookup support has issues.
In the trunk peer details in AMP I'd set "host=" to a
hostname. I've switched it t
On Thursday 04 May 2006 17:24, Tom Engleward wrote:
> You said I should help rather than whine. Am I not
> helping by announcing that I, too, am experiencing a
> problem which somebody else has mentioned, thus
> providing verification that the problem isn't just an
> isolated quirk caused by somebo
On Thu, 4 May 2006 23:29:37 +0200, Louis-David Mitterrand <[EMAIL PROTECTED]>
wrote:
> On Thu, May 04, 2006 at 10:31:17PM +0500, Vahan Yerkanian wrote:
>> Same, here, two asterisk 1.2.7.1 boxes connected to the same switch...
>> Over a week I see at least one case of one of the boxes becoming
>> u
> Is anybody on this list actually using iax2 for
> anything mission-critical?
Yes. 2K inbound / outbound calls a day to 30 remote locations, aggregated to
2 PRI's tied together with IAX2. All with IP address specified rather than
hostname. All with Asterisk 1.0.9. All with 99.9% completion rate,
On Thu, May 04, 2006 at 10:31:17PM +0500, Vahan Yerkanian wrote:
> Andrew Kohlsmith wrote:
> >On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote:
> >>I've got this low-ping 100%-up dsl connection between two asterisk
> >>1.2.7.1 servers. And oftentimes one of them would declare its opposit
--- Andrew Kohlsmith <[EMAIL PROTECTED]>
wrote:
> No, you are supposed to realize that a) this
> software cost you nothing. Not
> one penny. b) this software is user-supported.
> This means that in order to
> make it better you need to help. and c) we don't
> owe you anything. Not a
> thing
Please give me some heads up.
I'm having troube setting up my asterisk connecting to my SIP provider (SP). Here's the setup.
in sip.conf
I register asterisk to SP using
register => 15551234567:[EMAIL PROTECTED]
everything was ok for incoming call until I want to dial out using the same line
At 12:24 PM 5/4/2006, you wrote:
Customer
service may be stellar but when clients are actually trying to save money,
that's a damned hard sell.
My first thought too, but somewhere he said, call them for wholesale
rates which are reasonable.
Ira
Hello again..
Disregard my last message. I recompiled Asterisk and all is well.
I guess all the messing around with trying to get Zaptel to compile
correctly screwed something up with the Asterisk configuration.
Thanks.
-Ben
Original message:
This has got to be a stupid error I'm making..
Are you specifying the remote Asterisk box by IP or by hostname. If by
hostname, then specify it by IP. Asterisk's DNS lookup support has issues.
2) What is your qualify= set to. Set it to "yes" (2000), or don't set
it at all. Also look at the qualify smoothing options in iax.conf.sample.
try http://sourceforge.net/projects/web-meetmeChris Blunt <[EMAIL PROTECTED]> wrote:Hi List, Is it possible to store meetme config in a MySQL table? If so, any pointers would be appreciated. Thanks Chris -- Chris Blunt Entropy IT Ltd __
I have asterisk
1.2.7.1 running on Fedora core 5. Everything looked like it compiled
OK.
When a call is
bumped to voicemail, the message prompts sound fine to the user. However,
when the voice message is retrieved, it sounds "compressed" or speeded
up.
I have checked this
against t
couple of things... was asterisk compiled after zaptel? from the cli
try "load chan_zap.so" and see what you get
Ben Gore wrote:
This has got to be a stupid error I'm making...
I have been experimenting with different hardware and software
configurations before I decide what to use as a produ
On Thursday 04 May 2006 15:51, Tom Engleward wrote:
> Am I supposed to make a cron job to automatically tell
> asterisk to reload every so often, since iax2 likes to
> periodically die? Or maybe am I supposed to make a
> cron job to place a phone call every so often from an
> external phone into my
You can use my script, based on Chris Mason's script, to do most of what
you want, you can feed it your MAC's and Extensions and it will create the
phones.
Be warned, it's not pretty, my perl book was in storage so I did a lot of
kludging. Feel fee to update.
http://holburn.com/poly/pol
Edit your config files to enable persistance
Will remain across multiple calls, but not reboots
On May 4, 2006, at 2:51 PM, Jim Freeze wrote:
We are using the polycom 501 phones, and are having some challenges
with the volume setting. When a phone call comes in, the user ups the
volume for th
Title: Message
I have a conflict problem with the eth0
card and wct2xxp digium board. The PRI can
receive calls but my network connection is gone.
When I "cat /proc/interrupts" I get the
following:
1
..
1 ..
..
..
..
169 0 IO-APIC-level wct2xxp,
eth0
..
etc.
even before I "modprobe
wct2xx
This has got to be a stupid error I'm making...
I have been experimenting with different hardware and software
configurations before I decide what to use as a production platform. Up
until just recently things were going well.
But now it appears I'm unable to get access to my TDM400p from Asteris
sip.cfg
voice.volume.persist.headset="1" voice.volume.persist.handsfree="1"/>
Jim Freeze wrote:
We are using the polycom 501 phones, and are having some challenges
with the volume setting. When a phone call comes in, the user ups the
volume for the handset, but they have to repeat that for eve
That is what I thought too, but what about this:
http://lists.digium.com/pipermail/asterisk-commits/2006-February/001461.
html
???
>No. iSAC is a codec from GIPS. Likely the coded used by Skype.
>
>Michael
>
>On Thu, 4 May 2006 21:35:07 +1000, J
Hi Bruce,
We create a CSV file of our phone setup and then use shell scripts to
parse them and generate .cfg, phone.cfg, sip.conf,
voicemail.conf and entensions.conf entries.
Contact me off list if you would like a copy now (they're not quite
ready for prime-time yet) - the rest of you w
Hrmmm.
I thought there was already an option in the queue.conf or agents.conf
file (Though can't remember off hand what) that would set an agent
logged out or on 'pause' if they did not answer a call. No?
On 5/4/06, Christopher Mayfield <[EMAIL PROTECTED]> wrote:
it is two scripts an empty_queu
--- Vahan Yerkanian <[EMAIL PROTECTED]> wrote:
> Andrew Kohlsmith wrote:
> > On Thursday 04 May 2006 11:31, Louis-David
> Mitterrand wrote:
> >> I've got this low-ping 100%-up dsl connection
> between two asterisk
> >> 1.2.7.1 servers. And oftentimes one of them would
> declare its opposite
> >> UN
We are using the polycom 501 phones, and are having some challengeswith the volume setting. When a phone call comes in, the user ups thevolume for the handset, but they have to repeat that for every call.Currently, the volume level seems to reset itself at about 60%.
Is there a way for the user to
Kerry,
You didn't read my entire e-mail. How do I know that? Because if
you re-read it you'll see that I state:
"If you are a wholesole buyer of minutes, talk to them, don't just
take their prices on the main page... those are for residential and
regular customers. Their prices are very compa
Colin Anderson <[EMAIL PROTECTED]> writes:
>>Why is this hard to fake at all? You send a different fax to your
>>system, and replace the Asterisk audio file with the one from the
>>altered fax. Additionally, the client has no realistic way of
>>verifying the correctness of your audio-to-fax tran
it is two scripts an empty_queue.sh and a fill_queue.sh and a members script If you need intructions please tell me1047 $ cat empty_queue.sh#!/bin/bash# a script to remove everyone in the members script located in the same directory as this file
# to the Q 3901# can be called from a script#Local/[
On Thursday 04 May 2006 15:18, Sean Cook wrote:
> http://www.freedomphones.net/polycom/files/polycom.phone1cfg.pl-script
Yep that's the one that reads sip.conf and spits out phone[exten].cfg files.
It does not tie in mac addresses nor generate [macaddress].cfg files, though.
-A.
___
Sounds like a potential business opportunity. Someone could setup a fax
proxy service that provides this sort of digital signing / archiving.
The originator could simply dial a toll-free access number, receive a
2nd dialtone and then dial the destination. Meanwhile the proxy is
recording the call,
Hard to believe you arent associated with calleveryone.com as I find it hard
to believe that you would be extolling the virtues on one of, if not the
most expensive companies around. $7 a month plus 3.9 cents a minute
domestic, that's pretty much double the cost of anyone else. Customer
service may
Hi Bruce,
We've written software to do this as a service for our
customers. I can't give you the program, but we'd be willing to program
your phones for you. Contact me off list.
Michael Crown Managing Partner
www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED]
Something I made might help.
http://www.horanappraisals.com/asterisk/polycom_addphone/ -- there is a
script, addphone, and a folder called defaults that contains the
templates.
To use, I put the defaults folder and its contents and the addphone
script in my ftp or tftp root. I would make su
Try this one:
http://www.freedomphones.net/polycom/files/polycom.phone1cfg.pl-script
Sean
Andrew Kohlsmith wrote:
On Thursday 04 May 2006 14:45, Bruce Reeves wrote:
I am getting read to roll out close to 100 polycom phones and wondered if
any one knows of a program to take a list of MAC ad
>Why is this hard to fake at all? You send a different fax to your
>system, and replace the Asterisk audio file with the one from the
>altered fax. Additionally, the client has no realistic way of
>verifying the correctness of your audio-to-fax translation tool; it
>could just as easily output a
On Thursday 04 May 2006 14:45, Bruce Reeves wrote:
> I am getting read to roll out close to 100 polycom phones and wondered if
> any one knows of a program to take a list of MAC addresses, extensions, and
> names and generate the configuration files?
You can do this relatively easily with Perl. T
Hi there,
I'm looking into developing an in-house switchboard application. Does
anyone here know of a way to control a hard-phone from such an
application.
For example, the attendant forwards a call with another one in queue.
Once the first call has been forwarded (by keyboard shortcuts or
dragg
All, this doesn't appear normal to me, it appears as if ast is ignoring
the itignoreexpire variable.
sip.conf snippet:
rtignoreexpire=yes
asterisk -r
CLI>sip show settings
--snip--
Ignore Reg. Expire: No
--snip--
Does this look like a problem? :-)
Thanks, Matt
__
Colin Anderson <[EMAIL PROTECTED]> writes:
>>Why not capture the faxes (in or out) in tiff format, instead of audio
>>format? Setup your asterisk box to relay faxes!
>
> I think in this case the impact on the client would be much greater if you
> can show them a recreation of the image from the r
I am getting read to roll out close to 100 polycom phones and wondered if any one knows of a program to take a list of MAC addresses, extensions, and names and generate the configuration files?-- Bruce
Nortex Networks
___
--Bandwidth and Colocation provid
On Thursday 04 May 2006 14:08, Steve Underwood wrote:
> I would have no problem decoding a FAX, doctoring the images, then
> creating modified audio from them. During decoding, the FAX modems
> produce a channel estimate, so reproducing the characteristics of the
> original audio path wouldn't be h
Can I mix these in a single system... having problems getting
the tor2 driver or the wct4xxp drivers to load, although they
seem fine if alone in the system.
span=1,0,0,esf,b8zsbchan=1-23dchan=24
span=2,0,0,esf,b8zsbchan=25-47dchan=48
span=3,0,0,esf,b8zsbchan=49-71dchan=72
span=4,0,0,esf,b8zsbch
>I would have no problem decoding a FAX, doctoring the images, then
>creating modified audio from them. During decoding, the FAX modems
>produce a channel estimate, so reproducing the characteristics of the
>original audio path wouldn't be hard. I think it would be pretty easy to
>create fresh
> 'recognize'? The phone cannot know that the external IP has
> been changed, unless it is using a STUN server and
> periodically re-doing the STUN queries (which I doubt any phones do).
Thanks for clearing up my misunderstanding as to the point of STUN. :-) I
thought the phone would query the S
I am trying to use QSIG to interoperate with legacy PBXs.
I am looking to see whether any one knows whether Call Hold, Call Transfer, MWI
works with QSIG support in Asterisk.
Thanks in advance.
--Pillai
On 5/4/06, Olivier Krief <[EMAIL PROTECTED]> wrote:
2006/5/3, Marco Mouta <[EMAIL PROT
Anyone use this thing?
http://www.dlink.com/products/?pid=426
The fab sheet is totally useless for tech info. How does it work? By ToS?
Port number? Is it programmable? Can I prioritize an arbitrary port or ToS
bit?
tia
___
--Bandwidth and Colocation
http://www.soonr.com/web/front/features.jsp
Just saw this on the Always On Top 100 webcast (if you aren’t
familiar – click the url below)
http://deancollinsblog.blogspot.com/2006/05/always-on-awards-top-100-of-2006.html
Soonr looks like it rocks, haven’t tried it yet.
Ch
Colin Anderson wrote:
Why not capture the faxes (in or out) in tiff format, instead of audio
format? Setup your asterisk box to relay faxes!
I think in this case the impact on the client would be much greater if you
can show them a recreation of the image from the raw data; you could alw
Andrew Kohlsmith wrote:
On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote:
I've got this low-ping 100%-up dsl connection between two asterisk
1.2.7.1 servers. And oftentimes one of them would declare its opposite
UNREACHABLE.
Same, here, two asterisk 1.2.7.1 boxes connected to the sa
>Why not capture the faxes (in or out) in tiff format, instead of audio
>format? Setup your asterisk box to relay faxes!
I think in this case the impact on the client would be much greater if you
can show them a recreation of the image from the raw data; you could always
claim that a TIFF file wa
No. iSAC is a codec from GIPS. Likely the coded used by Skype.
Michael
On Thu, 4 May 2006 21:35:07 +1000, James Harper wrote:
>
>I assume you mean this:
>http://en.wikipedia.org/wiki/ISAC
>
>but maybe you are referring to one of the controller chips on BRI
>adapters?
>
>James
>
>> -Original
It has been approved. We started out trying to use
CVS on SourceForge, but
it appears that there have been major issues with CVS, so
we just switched to
SVN.
We need to
checkin a baseline, and start integrating patches.
Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On B
Hi,
Did anybody succeed remapping soft-keys on polycom 301 ?
I am having some problems with it.
I was trying to remap Transfer button as the first option while being in a
call.
It works but The name of the soft key is still HOLD and while I am not
in a call I see button "NewCall" that sudd
On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote:
> I've got this low-ping 100%-up dsl connection between two asterisk
> 1.2.7.1 servers. And oftentimes one of them would declare its opposite
> UNREACHABLE.
I see this happen on occasion as well -- same type of setup here, static IPs,
n
Why not capture the faxes (in or out) in tiff format, instead of audio
format? Setup your asterisk box to relay faxes!
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Thursday, May 04, 2006 10:07 AM
To: Asterisk Users Mailing Lis
I have tried using the autologoff in the agents.conf and it sort of works. I
set it to 5 seconds to test it and it has taken anywhere from 35 to 60
seconds to actually do something at which point it does indeed log out the
agent.
I don't want to be pestering agents with test calls to see if they a
I've got this low-ping 100%-up dsl connection between two asterisk
1.2.7.1 servers. And oftentimes one of them would declare its opposite
UNREACHABLE.
Why can this happen? The host stanzas in iax.conf have raw IP's, so no
DNS monkey business here.. An inquiring mind wants to know.
___
Just wanted to add my 2 cents. We were with voipjet, and do still use
them for occassional backup.However, their lack of personal
service and inability to get ahold of someone drove us away.After
several total blackouts (like what happened yesterday), and no
responce we finally put out an
> In the PAP2's setup there are all of these "Vertical Service Activation > Codes" that start with star and "Outbound Call Codec Selection Codes", > also the setup menu is accessed by pressing star four times, could they > be intefering with dialing numbers that start with a star? And is there > an
Yapp, timeout is set to 1500ms.
What kind of dtmf mode? As far as i know there are just 2.
Relaxdtmf yes or no
Or am I wrong?
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Mark Ackroyd
Gesendet: Donnerstag, 4. Mai 2006 16:52
An: 'Asterisk
Before I go nuts trying to figure this out, is anyone using DISA in this manner?
exten => s,1,DISA(X|context|callerid)
Everything works except the caller ID part. What I had wanted to do is to setup up a file of authorization codes where each code was associated with a context and caller i
Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different
mobile network providers. Nothing.
I played with the rx/txgain values from hearing nothing to too loud...
I have no more ideas.
Marc
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] I
Marc Scheuffler wrote:
Hi all,
I am trying to detect DTMF keys from a mobile when asterisk make an
outgoing call to the mobile.
The DTMF detection on incoming calls (also FROM mobiles) is working very
well.
The only problem is if asterisk called the phone... Nothing is detected.
I am using a
>From my recent problem on this sort of thing, I'd suggest you set the
timeout to around 1500ms in the feature.conf file. This is of course if your
using the DTMF digit's to activate any of the features.
also make the devices both sides of the call are using the same DTMF mode.
Mark
>
> Hi all
Hi all,
I am trying to detect DTMF keys from a mobile when asterisk make an
outgoing call to the mobile.
The DTMF detection on incoming calls (also FROM mobiles) is working very
well.
The only problem is if asterisk called the phone... Nothing is detected.
I am using a digium te205p with PMX/PST
I have a client that 'NEEDS' (his words not mine) to make sure that all
faxes, emails, calls, and mail are archived. Phone and email are simple,
Mail depends upon the integrity of the mail room, Faxes however can be
sent from anyone. They would like this as they recently had an issue
with a fax sen
In the PAP2's setup there are all of these "Vertical Service Activation
Codes" that start with star and "Outbound Call Codec Selection Codes",
also the setup menu is accessed by pressing star four times, could they
be intefering with dialing numbers that start with a star? And is there
any way to
Well, yes and no. I tested that before and it causes a silent ring
instead of a call rejection. I actually want to disable the entire feature. So
the phone always rings unless you're actually on the phone.
Thanks for the reply though!
Regards,Jan
Från: [EMAIL PROTECTED]
[mailto:[EMAIL P
QSIG was just the protocol communication between Legaccy PBX and Asterisk.My users connect to Asterisk through SIPOn 5/4/06, Olivier Krief <
[EMAIL PROTECTED]> wrote:
2006/5/3, Marco Mouta <[EMAIL PROTECTED]>:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
I've made some tests using thi
Yes, sorry I was wondering if anyone is working on ISAC voice codec
I have seen a patch
http://lists.digium.com/pipermail/asterisk-commits/2006-February/001461.
html
But not seen anything anywhere else...
trond
I assume you mean this:
http://en.wikipedia.org/wiki/ISAC
but maybe you
> This is a very KGB / NSA / InterPOL / CIA type
> question, but if I have a
> recorded file (G.711, no compression) can I feed it
> into standard in of
> an application and have it recreate the fax that was
> send?
What is the specific reason as to why you want to
record it to a file and send it
Attribute Values Default Interpretation call.rejectBusyOnDnd 0, 1 1 If set to 1, reject all incoming calls with the reason “busy” if do-not-disturb is enabled. Have not used, but looks like it may ignore the key if this is 0Let us know...On May 4, 2006, at 2:22 AM, <[EMAIL PROTECTED]> <[EMAIL PROTE
2006/5/4, Craig Guy <[EMAIL PROTECTED]>:
If you have both sides of the call it is possible. It may not be practicalthough. If one side was using spandsp then it is both possible andpractical.CraigCould you elaborate ?
And if a fax is recorded with Asterisk voicemail application (in case an error
How can you store "pauses" in speed dials for the GXP-2000? I used
something like 8005551212,,,1,7890 to dial the toll free number, wait
6 seconds (I'm used to the commas being a 2 second delay), pressing
1, waiting 2 more seconds and then entering 7890. However, when I
press the speeddial
2006/5/3, Marco Mouta <[EMAIL PROTECTED]>:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
I've made some tests using this in Portugal and seems to work:---
switchtype=qsig ; you may try this in your zapata.conf-
2006/5/3, Asterisk User <[EMAIL PROTECTED]>:
Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking?
Do you mean something like ECMA 336 ?http://www.ecma-international.org/publications/files/ECMA-ST/Ecma-336.pdfRegards
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