Stefan Gofferje wrote:
That's brilliant! And so easy...
Works exactly as supposed. You should put it onto the wiki under section
"tips & tricks".
Regards,
Stefan
PS: Chris, your boss might like this also!
Wow, is it too late to put in for royalties? ;)
I'm glad it worked for you.
--
Andrew Thom
On Tue, 2005-02-08 at 13:53 -0500, Giovanni Powell wrote:
> In most configuration files i see that they comment lines instead of
> adding spaces.
>
> e.g. - correct way
Correct according to who?
> ;
> ;IAX configuration
> ;
> [general]
> blah blah blah..
> ;
> register => ..
>
>
>
Hi,
On Tue, 2005-02-08 at 10:01 +0100, Peer Oliver Schmidt wrote:
> I now experience a lot of drop outs during a conversation. They last 5
> seconds and more, but eventually the sound comes back (if the other side
> has not hang up).
Can you tell us wether you are using ISDN or VoIP phones, and
Hello everybody,
I have an Asterisk box with a TDM04B and would like to connect it to a GSM
Gateway.
Can someone tell me whether i can get the callerid for incoming calls in
this case?
Thanx
Lamine
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There are some SIP firmware images available..
The latest 2.x release SIP/H.323 firmware:
ftp://ftp.rekom.ru/pub/ata18x/ata18x-v2-16-2-030909a-1.zip
http://kvin.lv/pub/Cisco/ata18x-v2-16-2-030909a-1.zip
3.1(0) firmware:
http://kvin.lv/pub/Cisco/ata_03_01_00_sip_040211_1.zip
http://voip-i
> -Original Message-
> From: David Brodbeck [mailto:[EMAIL PROTECTED]
> I finally figured out my extension D issue. The extension
> works fine as
> long as Background() has finished playing. But during
> playback, the "D"
> tone is not recognized. Is there any way to configure this?
I'm trying to use the SPEEX codec with Voicepulse.
Here's what I see in the CLI when I RELOAD:
-- Reloading module 'codec_speex.so' (Speex/PCM16 (signed linear)
Codec Translator)
== Parsing '/etc/asterisk/codecs.conf': Found
-- CODEC SPEEX: Setting Quality to 5
-- CODEC SPEEX: Settin
I have just created a very rough (read hack-ish) version of a Polycom
SIP phone configurator. It allows you to define phones, create
registrations, and such. By describing stuff about users, I am
attempting to divine what the configuration should be. This is a VERY
early first step in that direc
dorian logan wrote:
At the moment my VoIP system is considered to offer a lower quality
call experience than a standard phone ( I am sure it is my equipment and
configs that is the culprit) - I would love to demonstrate what is
possible with audio > 8khz
"standard phone" uses ulaw in the USA/C
Hi,
I used this command line :
sipp -sn uac URL_of_*_server -trace_err
Is there any better ?
Regards,
Rob.
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Hi,
Regarding these high bandwidth CODECs - is it possible to upgrade
asterisk to record at a higher quality bit rate too - is Asterisk based
on a 8Khz system. We would like to stream calls from SIP phones to the
internet at a higher quality than a standard phone, also it would be
great to bu
Funny, when I tried faxing this to myself it didn't arrive either.
Looks like it's blank:)
On Tue, 8 Feb 2005 16:59:28 +0100, Roy Sigurd Karlsbakk
<[EMAIL PROTECTED]> wrote:
> hi
>
> I've been trying to fax digium this agreement for a month or so now
> Any chance they can fix their fax?
>
> roy
Kevin P. Fleming wrote:
I've been considering doing this as well... something like a "dial
list", with a delay before dialing and a timeout for each entry.
Delay before dialing and timeout for each entry? I think I follow your
choice in words there, your saying - forgive possible stupid syntax
I suggest you take a look at your configuration files, especially
ipmid.cfg, in , the attribute "up.oneTouchVoiceMail".
Arrgh. Quite right, David. I guess I'll stop posting my rants, and
actually keep track of my config files. Yet another stupid error on my
part.
Stefan Gofferje wrote:
Andrew Thompson schrieb:
Stefan Gofferje wrote:
[private_huntgroup_day]
exten => s,1,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED],15,rt)
exten => s,2,Wait(1)
exten =>
s,3,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED],20,rt)
exten => s
Hi,
On Tue, 2005-02-08 at 20:29 +0100, administrator tootai wrote:
> question: does this card do PSTN *and* ISDN? What does mean ADSI on the
> digium hardware page? May I mix PSTN and ISDN (eg 2 fxo for ISDN and 2
> fxo for PSTN)?
No you cannot. The TDM card is PSTN/analog only. For ISDN use a
Can Asterisk-oh323 generate the package BRQ?
Thanks
Megan Willigs
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In most configuration files i see that they comment lines instead of
adding spaces.
e.g. - correct way
;
;IAX configuration
;
[general]
blah blah blah..
;
register => ..
Is this incorrect:
;
;IAX configuration
[general]
blah blah blah...
register =>
I guess what i'm trying to s
Hi list,
question: does this card do PSTN *and* ISDN? What does mean ADSI on the
digium hardware page? May I mix PSTN and ISDN (eg 2 fxo for ISDN and 2
fxo for PSTN)?
Thanks for sharing your knowledge ;-)
--
Daniel
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Aster
> Message: 4
> Date: Tue, 8 Feb 2005 13:09:23 -0500
> From: Noah Miller <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Polycom screwed up Messages button in 1.4.1?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; c
Does anyone know how to get rid of these hung channels?
I am getting this when I do a:
show sip channels
209.82.xxx.xxx0071495217 2591218534@ 00103/1 unknow(d)
209.82.xxx.xxx0041590104 0690231739@ 00103/1 unknow(d)
209.82.xxx.xxx0070259259 3265102826@ 00103/000
> -Original Message-
> From: Andrew Thompson [mailto:[EMAIL PROTECTED]
> Twice in the last week or so, I've received a message similar to the
> attached.
>
> A portion of the attachment that's attached is not in
> English. Is this
> my mail server failing, or someones who's on the list
I finally figured out my extension D issue. The extension works fine as
long as Background() has finished playing. But during playback, the "D"
tone is not recognized. Is there any way to configure this? Is this a bug?
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Twice in the last week or so, I've received a message similar to the
attached.
A portion of the attachment that's attached is not in English. Is this
my mail server failing, or someones who's on the list?
--
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
--- Begin Message ---
Livevoip has 479; iax.cc might be willing to work with you.
> -Original Message-
> From: Kelly Griffin [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, February 08, 2005 10:25 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] VoIP Termination in 479
Hi:
I'm a bit confused about CODEC declarations in IAX.conf.
In the [GENERAL] section, should I declare NO codecs and no
[allow|disallow all] statements in favor of declaring all CODEC rules
within each context section?
If this is the case, I'm wondering then what exactly should remain in
the g
I'm using [EMAIL PROTECTED] with the AMP interface and I'm having troubles
getting incoming calls working properly. In AMP, I have it set to take
incoming calls from PSTN, during regular business hours, to be sent to
extension 201. The include statement for extentions-additional.conf is
uncommented
Is it possible your MoH class isn't set properly for that call? It
should say class 'default' not class ''. But then again, IANAME (I Am
Not A Music-on-hold Expert) ;)
Guills
> -Original Message-
> From: Stefan Gofferje [mailto:[EMAIL PROTECTED]
> Sent: Saturday, February 05, 2005 3:39
In most configuration files i see that they comment lines instead of
adding spaces.
e.g. - correct way
;
;IAX configuration
;
[general]
blah blah blah..
;
register => ..
Is this incorrect:
;
;IAX configuration
[general]
blah blah blah...
register =>
I guess what i'm tryin
Mike Wright wrote:
I was looking for something to connect a couple of POTS handsets to my
asterisk server and found this on ebay
http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&category=162&item=5162868118
&rd=1
The documentation says that it does SIP - therefore will it work in an
asterisk environ
Robert Spielmann wrote:
Hi everyone,
I have a question concerning DNS SRV lookups. The situation is like this:
- one central Asterisk server
- many domains with SRV records, let's say we have bar.com and doe.com
Now the question is: if the SRV lookup is done for [EMAIL PROTECTED] the call is
mappe
Mike:
The ATA will do SIP although I cannot say if this ATA will do
SIP out-of-the-box.
-Steve
Mike Wright wrote:
I was looking for something to connect a couple of POTS handsets to my
asterisk server and found this on ebay
http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&category=162&item=5162868
i installed it the other day but from some reason can only get one of
my budgetone 100's to register...any thoughts? I have tried upgrading
firmare but that didn't seem to work.
thanks in advance,
ken
Steve Rawlings wrote:
Why not try [EMAIL PROTECTED], it only takes about an hour to install a
I was looking for something to connect a couple of POTS handsets to my
asterisk server and found this on ebay
http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&category=162&item=5162868118
&rd=1
The documentation says that it does SIP - therefore will it work in an
asterisk environment.
--
No vir
Hello,
I downloaded asterisk 1.0.5, and the cvs version, and none of them
has the attended transfer option (x and X). I see a patch here:
http://bugs.digium.com/bug_view_page.php?bug_id=0002460
but this patch doesn't applies well in cvs version or 1.0.5. How can
I make at
Stefan Gofferje wrote:
[private_huntgroup_day]
exten => s,1,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED],15,rt)
exten => s,2,Wait(1)
exten =>
s,3,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED],20,rt)
exten => s,4,Voicemail(u810920)
exten => s,5,Hangup
exten
I'm using Asterisk 1.0.4 with AMP and Broad Voice.
I have that with only 5 XTen Lite phones.
I'm able to call / etc with internal phones just fine.
I can call outside Vonage Numbers, and other
BroadVoice Numbers. I have vonage where I live (626)
and can call that fine. However, other 626 num
Greetings all!
astfax v1.0 has been released, and is now available at our website.
You can check out the package at http://www.inter7.com/?page=astfax
astfax allows you to create an email to fax gateway.
It processes incoming emails with attached fax images, and sends
them outbound via the txfax
Why not try [EMAIL PROTECTED], it only takes about an hour to install and be up
and running with softphones like x-lite. This takes care of the os and
asterisk in one cd.
Steve
- Original Message -
From: "Shaoul Jacobson - TELLINK" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List -
asterisk-oh323-0.7.0 is for Asterisk CVS.
How did you manage to compile it with Asterisk-1.0.3?
Use Asterisk-1.0.3 with asterisk-oh323-0.6.5.
Michael.
Roger Schreiter wrote:
Hi,
which is currently a stable combination of asterisk and
asterisk-oh?
The combination of asterisk-1.0.3 and asterisk-oh-0.
Chris Wade wrote:
Really I think that the best solution would be to have Dial/RetryDial
have an optional DELAY for each channel listed in the dial-string. Such
that SIP/101&SIP/102[5]&SIP/103[10]&SIP/104[15] would result in SIP/101
being rung immediately (with retry options to continue attempti
Why?
When you register with another provider, the use or not of MD5 Auth is
up to him/her...
When other clients are registering with you, you may require MD5 auth
(Auth=MD5)... If you don't want to have a cleartext password, you can
use md5secret=. instead of secret=
Greetings
Julian.
On
Title: Asterisk performance monitoring
Hello,
Has anyone used any 3rd party web based software to get performance information out of Asterisk?
Looking for CPS, call setup times, voicemail database utilization etc…
Cheers
Keith.
___
Asterisk-Us
I think Polycom has added another feature that nobody wants.
With MWI configured, and a phonexxx.cfg that has this:
Under 1.3.4 and earlier, the phone would immediately connect you to
extension XXX.
Under the 1.4.1 firmware, when you press the messages hard button on
the phone, t
On Tue, 8 Feb 2005 12:30:36 -0500, Noah Miller <[EMAIL PROTECTED]> wrote:
> > I installed a tdm400p into a old p2 machine.
> > I'm not able to see it under /proc/interupts or using lspci..
> > we removed all other cards. changed slots, forced irq to that slot..
> > etc etc.
> >
> > what is the min
Stefan Gofferje wrote:
Hi Folks,
on my home asterisk, I have a "huntgroup" for incoming calls on the
private line which first let ring my phones in my office and living
room, after a while then office, living room and bedroom.
I do this by simply putting two dial statements in sequence:
[private
Combing thru the Wiki, I did find that Asterisk does have some secret
sauce with respect to sorting out what the caller is dialing...
It is covered in the wiki page "Asterisk Extension Mapping"
http://www.voip-info.org/tiki-index.php?page=Asterisk+Extension+Matching
This page does an excellent j
I run Debian, and it's not hard to get a base install running. If you
want a GUI and such, then it'll be more than "follow the screen
prompts." I've been writing some Debian documents, if you're
interested, email me off-list.
Anyhow, on pretty much any distro, you can make your own packages
(RPM,
Hi,
which is currently a stable combination of asterisk and
asterisk-oh?
The combination of asterisk-1.0.3 and asterisk-oh-0.7.0 is
not stable at all and crashes approx once the hour when
having approx 3 simultanious calls.
Thanks for telling me your experience!
Roger.
_
On Mon, 7 Feb 2005 21:24:53 -0800, Paul Crick
<[EMAIL PROTECTED]> wrote:
> Maybe something will change in a future software release..
The terminology between VOIP and POTS is different and unclear. If I
go in to an office supply store and buy a 2 line phone, it plugs in to
two wall jacks and I can
I installed a tdm400p into a old p2 machine.
I'm not able to see it under /proc/interupts or using lspci..
we removed all other cards. changed slots, forced irq to that slot..
etc etc.
what is the min specs needed to get one of these cards running?
PCI 2.2
I don't think the TDM cards will work on
Hi all
I installed a tdm400p into a old p2
machine.
I'm not able to see it under /proc/interupts or
using lspci..
we removed all other cards. changed slots, forced
irq to that slot.. etc etc.
what is the min specs needed to get one of these
cards running?
thanks
L
Do you Yahoo!?
Ya
Roy Sigurd Karlsbakk wrote:
are there any codecs around that allows high quality as in "studio
lite"? it may consume high bandwidth, and hopefully allow some packet
loss.
I'm not sure what "studio lite" means to you. Maybe hard figures would
be more precise.
G.722 might be interesting : 64 kbps, 7
how to make g.729 preferred, but failover
to gsm
I've purchased a few g.729 licences,
and would like to set up iax.conf such that g.729 is used if they are available,
but then it fails over to gsm.
I'm not sure how to specify such a preference.
I'll let the server transcode from ulaw (from the s
Get a converter to Q.931, we use one called
an IQ200 from (I vaguely recall) Teltrend, search the web, works fine, easy to
setup, we've used them at two customers now with no problems at all.
Steve
From:
Stephen Owen hosted [mailto:[EMAIL PROTECTED]
Sent: 08 February 200
Voip to voip (in whatever form that takes sip-sip sip-iax iax-iax) is
what I am using asterisk for.
I would have thought mandrake would have been ok - but haven't used it
for a while. I'm running FC2 (fedora core2) and asterisk complies and
runs without any problems.
Dont fear make. Apps, for
On 05:14, Tue 08 Feb 05, Mazhar Hussain wrote:
> Hi to all,
>
> I and using asterisk with following setup.
>
> 1. TDM400p card with four FXS modules,
> so there are four analog phone lines on four zap channels,
> My setup is working fine.
> And version is like such
> Asterisk CVS-v1-0-11/27/04-20
479-254
479-636
479-795
479-359
479-451
479-442
479-751
---
Kelly D Griffin
Network Engineer
Tantella Wireless
http://tantella.com
800.636.0306 Voice
479.464.8998 Fax
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Digital
Support Technologies
Sent: Wedn
Hi,
I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards)
1. the distro
I downloaded a "free mandrake 10.0 - 3 CD's) but some packages seem missing
(some C or C++ or python ...)
(buy the full version )
maybe the latest fedora is more complete ?
or easier to complete w
On Mon, 7 Feb 2005 23:16:05 -0600, Eric Rees <[EMAIL PROTECTED]> wrote:
> Has anyone seen this message trying to install an TDM400.. spurious
> 8259A interrupt: IRQ7
>
> This error happens after I do a modprobe wctdm and then the system
> hangs. I am installing this in an Asus motherboard with a
The stable tree from cvs includes any patches since release that was
also commited for the v1-0 tag since some issues were found after the
release but not major enough for a new tar ball release.
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Yes, We offer that stuff we can get numbers in most U.S area's
Contact us
800-508-1251
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kelly
Griffin
Sent: Tuesday, February 08, 2005 11:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
S
On Tue, 8 Feb 2005 11:56:18 +0200, Yousri Farouk <[EMAIL PROTECTED]> wrote:
> Hello all,
>
> i would like to configure TDM11B with Asterisk, if any one have the
> configuration steps please provide me it.
>
> Thanks in advance
Have you tried looking at Digium's site??
http://www.digium.com/i
is the v1-0 CVS branch supposed to be stable as in STABLE, or should
one use releases?
v1-0 is the tag used for the latest changes to the stable branch.
Releases are still your best bet, but if you are monitoring the CVS
mailing list for commits to v1-0 stable, then you may see a patch go
in that f
how can I tune SIP jitter? is it possible today in asterisk?
I assume you are asking for how to alleviate the effects of jitter on
the
RTP audio streams initated by SIP? Asterisk currently only has a jitter
buffer for IAX, not for RTP streams. There are pland for the next
generation jitter buffer
I am looking for termination of numbers in the 479 area code. I would like
to either port them through my * box or direct SIP connection from the
customer. I am in need of over 100 DID's. Anyone know of anyone that has
this service besides Vonage or Packet8?
---
Kelly D Griffin
Network Engineer
I have just setup music on hold by downloading and installing mpg123 r
Now I have music on hold but it sounds terrible - clipping, buzzing,
digital distortion, and its too loud (which probably isn't helping) and
I'm just running it thru the 'default' line in music onhold.conf line
default => qui
Steve Blair writes
I can redirect and relay calls to numerous destinations via
SER but because the Octel needs an SMDI interface for mailbox
identification I am stuck, none of the solutions thus far support
SMDI-SIP munging.
I just started thinking about the possibility of using Asterisk
with a f
hi
I've been trying to fax digium this agreement for a month or so now
Any chance they can fix their fax?
roy
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Sounds like maybe you don't have either Postgres installed or PHP
confirgured to use it.
If you use RPMs, check for something in the php-pgsql family (%yum
install php-pgsql)
As a warning, you will also need to enable PHP globals in your php config.
Hope that helps,
J
On Tue, 8 Feb 2005 09:1
I had problems as well. It was do to my sip.conf and extension.conf
Here are my conf files.
sip.conf
[general]
port=5060 ; Port to bind to
bindaddr=0.0.0.0 ; Address to bind SIP channel to
context=default ; Default context for incoming calls
regi
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of thieumS
>Sent: Tuesday, February 08, 2005 10:09 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [Asterisk-Users] ASTCC simultenous calls per card
>
>
>Hi guys,
>do you know if it'
Noah,
Thanks for your input on this. I am not sure if it handles incomng
connections or not - will have to check. I don't think it will work
either - worth a shot to ask though.
Thanks!
- Pedro
On Tue, 8 Feb 2005 10:26:48 -0500, Noah Miller <[EMAIL PROTECTED]> wrote:
> > We have a client that
Hi everyone,
I have a question concerning DNS SRV lookups. The situation is like this:
- one central Asterisk server
- many domains with SRV records, let's say we have bar.com and doe.com
Now the question is: if the SRV lookup is done for [EMAIL PROTECTED] the call
is
mapped to [EMAIL PROTECTE
.
Specifically, X is not a digit, you must either use "" for no
interuptions permitted or use 0123456789 for all digits available to
interupt.
I also 'discovered' that you cannot send a sequence of commands to
asterisk without
reading the results between each command submission. Similar to the
It's probably too late for me to say I don't want to sound like a
jerk. :-P It was late and I get frustrated when people don't use
available resources. I apologize. Anyways, a quick search of
google..
http://www.google.com/search?q=asterisk%20n%20priority
pulls up
http://www.sineapps.com/new
We have a client that wants to bond 2 DSL circuits instead of getting
a T-1 (or similar) at their office to run their VoIP traffic on. We
came across this Multihomed Gateway (MH200):
http://www.cyberpathinc.com/mh200/details.htm
Does anybody think this would work if installed at the client locatio
> -Original Message-
> From: David Brodbeck [mailto:[EMAIL PROTECTED]
> Okay, the problem appears to be that I'm tone deaf. ;)
>
> I finally thought to turn on debugging on the channel. The
> PBX is sending
> "D", not "*". The programmer of the previous voice mail system (whose
> confi
Hi guys,
do you know if it's possible to handle more than 1 call per card
with astcc ?
Thank you.
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Checkout http://www.voip-info.org/wiki-NVBackgroundDetect
I haven't had a chance to try it yet, but supposedly it works on SIP,
ZAP, and IAX.
On Tue, 8 Feb 2005 21:26:28 +1100, Mike Sander
<[EMAIL PROTECTED]> wrote:
> That's all very well, but what do you do if you only have SIP extensions and
>
I got the called-name lookup going using php: http://muware.com/asterisk
If you want to "pop up" additional details, you'll need a client
application to notify a computer near the extension -- this is possible,
but will require quite a bit more work.
> -Original Message-
> From: Mazhar Hu
Pedro,
My understanding is that this will not allow for any balancing on any
connections once they are established. Any connection on the first line
that is already established will continue to stay on that line/ip
address until the connection is dropped and a new one is established.
It would be
On February 8, 2005 09:48 am, David Brodbeck wrote:
> > With you listening in on the same physical 2-wire that the
> > PBX uses and you
> > send *, does Asterisk see it? If you're on a call from the
> > PBX to Asterisk and dial * from the PBX phone, does * see it?
>
> Yes, in both cases.
How shor
Okay, the problem appears to be that I'm tone deaf. ;)
I finally thought to turn on debugging on the channel. The PBX is sending
"D", not "*". The programmer of the previous voice mail system (whose
configuration I was cribbing from) seems to have made the same mistake.
_
On Tue, 2005-02-08 at 06:27 -0600, Rich Adamson wrote:
> Looking for some advanced thoughts relative to exten number assignments.
>
> We're in the planning stage for rolling out asterisk at multiple small
> US telco/isp operations. Their typical voip customer has had their
> pstn line for a l
On Tue, Feb 08, 2005 at 02:58:01PM +0100, Roy Sigurd Karlsbakk wrote:
> >>are there any codecs around that allows high quality as in "studio
> >>lite"? it may consume high bandwidth, and hopefully allow some packet
> >>loss.
> >>
> >
> >I'm not sure what "studio lite" means to you. Maybe hard figur
Does the order in which you allow codecs matter? cuz i've found that
somethings work better if you allow them in a particular order.
Alot of warnings and errors can also be eliminated.
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> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
> On February 8, 2005 09:28 am, David Brodbeck wrote:
> > What puzzles me is it works fine if I dial *, but if I hang
> up instead and
> > the PBX sends *, Asterisk doesn't seem to get it.
>
> With you listening in o
> Which linux is prefereable ? for asterisk ?
As long as you know how to rebuild your kernel, how to install modules, and
how to manage basic unix security, the best Linux for Asterisk is the one
you're most comfortable with.
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On February 8, 2005 09:28 am, David Brodbeck wrote:
> What puzzles me is it works fine if I dial *, but if I hang up instead and
> the PBX sends *, Asterisk doesn't seem to get it.
With you listening in on the same physical 2-wire that the PBX uses and you
send *, does Asterisk see it? If you're
Hey gang,
I'm trying to work out all possible scenarios using SER & Asterisk in our
upcomming deployment. The example scenario is 50 different customers, all
with different numbers of SIP UAs. All UAs would register with SER; This
will help keep any inter-office conversations off our bandwidth sin
I know Q931 cards are supported, does anybody know
how to
go about supporting DASS II ?
Thanks
Stephen
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-Original Message-
Good day all.I get the warning message on my system,this is for a snom
220,it repeats this message a few times,please help Feb 8 09:29:26
WARNING[1093445952]: chan_sip.c:683 retrans_pkt: Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 105
(Non-critical Reque
> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
> On February 8, 2005 08:44 am, David Brodbeck wrote:
> > The sequence I hear on the extension, when I plug in an
> analog phone, is
> > the click of the phone at the other end being hung up,
> followed immediately
>
at first it was not answering (there was complete
silence after 200 Ok and ACK). i dont know what was
the reason. but now it is answering me(asking for
mailbox then password). but the problem that is is not
authenticating me to check mailbox i have defined
mailbox and 1234 password (it is sayi
I have a question regarding to OS platform.
As I see on Wiki -s homepage there are many type of linux version.And in some of them there are reported errors (regarding to asterisk ) for exemole in rad hat .
Can you tell me what is the best linux paltform ,( version ), which is supported by digiroom
-- Forwarded message --
From: Carlos Gabriel Drach <[EMAIL PROTECTED]>
Date: Tue, 8 Feb 2005 11:20:01 -0300
Subject: Re: [Asterisk-Users] Record() cut off after 40 sec
To: Steven Critchfield <[EMAIL PROTECTED]>
On Mon, 07 Feb 2005 15:35:46 -0600, Steven Critchfield
<[EMAIL PROTECT
Need Help ..
I am trying to install AreskiCC Calling Card application but each time
I tried to login as root -- I recieved this error
Fatal error: Call to undefined function: pg_pconnect() in
/var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 67
Please help me - I am stuc
I put dtmfmode=rfc2388 into the sip.conf definitions for each sip client
and now asterisk is recognising the # key press - guess it wasn't
hearing the dtmf tones...
Now blind xfer works - how do I do attended xfer? I have read posts
about it being in the CVS version - I am running the 1.0.3 rel
We have a client that wants to bond 2 DSL circuits instead of getting
a T-1 (or similar) at their office to run their VoIP traffic on. We
came across this Multihomed Gateway (MH200):
http://www.cyberpathinc.com/mh200/details.htm
Does anybody think this would work if installed at the client locat
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