I have not noticed any issues with quality, just with caller volumes
being way different when mixing 2 channel types (ZAP and SIP
specifically). Here's my custom script for processing the recording
files. Make sure you use option m on your monitor command so that the
custom script will run. My
NORMALIZE=nice -n 20 /usr/bin/normalize --no-progress -a 1.0 --peak
Which package comes this normalize from?
Elmar
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Thanks very much Armin.
After migrating to chan_capi-cm, the issue now is, everytime a dial statement is
executed, it fails and restarts asterisk! The restart I believe is due to safe_asterisk
script. So, in my opinion chan_capi-cm terminates asterisk process abruptly.
When I replace the
Hi,
I configured a pair of asterisk box, one of them (the first) ISDN connected
to the PSTN.
I configured an integrated dial plan, and I SIP connected the two pbx.
No problems in dialing extensions defined in one of the two pbx from any
pbx.
My problem is this: How can I make the users logged in
On Mon, 19 Sep 2005, Voicomm User wrote:
Thanks very much Armin.
After migrating to chan_capi-cm, the issue now is, everytime a dial
statement is
executed, it fails and restarts asterisk! The restart I believe is due to
safe_asterisk
script. So, in my opinion chan_capi-cm terminates
The best place for Astri Con 2006 would definatly be
Naples, ITALY
__
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http://mail.yahoo.com
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Matthew Boehm wrote:
I currently not use it due to some limitations in * realtime .
Such as?
-Matthew
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On Mon, 2005-09-19 at 14:27 +1000, [EMAIL PROTECTED] wrote:
Hi, all
Did anyone tried TDM400 card on old main board (Intel 440LX chipset -- PII)?
The reason I am asking is because TDM400 needs PCI2.2 and main board is PCI2.1
I do not want to upgrade yet.
Since Digium is so clear about using
Shaun Ewing wrote:
These options have been tested with Asterisk 1.0.8 and CCM 4.1(2)sr1.
If this is something that people would be interested (and you made it
this far), I'd be quite happy to whip up some instructions and add it
to the wiki.
Please do!
Doug
I connected
my * box to two ISDN BRI lines using two ISDN pci cards (w/ the Cologne
chip-set, hfc) and set up with mISDN.
so using chan_mISDN.
When I set-up the first ISDN pci card for testing everything worked
just fine.
Now that I have two cards, things are working fairly OK with the
Thanks,
I do realise that I will have to upgrade, just did not want to do it so
soon.
Digium says that it is not recommended to use main mainboard with PCI less
than 2.2. They did not say it will not work. I just wanted to find out if
anyone had any experience with this particular chipset. So
Hi, all,
I got my * to work with voipbuster service. And it works quite well when I
am calling USA or Europe. However, for local calls, I am experiencing long
delays (About 1s). As far as I know, voipbuster application does not have
this problem.
I am using IAX and gsm codec.
Any ideas on
Please,
send us zaptel.conf and zapata.conf and say us what card you
have(TE110P, TE410P...). And what is your country.
Regards,
srsergio
-Mensaje original-
De: manish kumar [mailto:[EMAIL PROTECTED]
Enviado el: lunes, 19 de septiembre de 2005 6:32
Para:
Rudolf Ladyzhenskii wrote:
Hi, all,
I got my * to work with voipbuster service. And it works quite well
when I am calling USA or Europe. However, for local calls, I am
experiencing long delays (About 1s). As far as I know, voipbuster
application does not have this problem.
I am using IAX
Please do!
Doug
Here it is:
http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Voicemail+Integration
It needs some cosmetic work, but I think that gets the point across :-)
-Shaun
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I'm new to asterisk and need some help with ideas to handle this
configuration question.
I am trying to establish a termination point/DID number in another
country. I am currently running Asterisk CVS-HEAD. My foreign provider
uses SIP and authenticates via IP address. I am not required to
You probably have satellite connection between you and voipbuster. So,when you call local then your voice passing twice that connection (from you to voipbuster, and from voipbuster to your neighborhood)
On 9/19/05, Paul [EMAIL PROTECTED] wrote:
Rudolf Ladyzhenskii wrote: Hi, all, I got my * to
Thanks, Paul
I have tried to set tos=0x18 and this improved it a lot.
Also, looks like I have to seriosly look into setting up QoS on my firewall.
At the monet I shut down P2P application running on one of the PC and this
has helped too. Not sure why it was affecting delay. I would expect
Obviously setting this for each and every phone on Callmanager was not
an option for any wide deployments, and Paul Davidson investigated
some of the other options. Paul discovered that it was possible to
setup a voicemail pilot, tick the voicemail box, etc. but you would
lose the ability to have
Hi all,
I have configured Asterisk to call to PSTN phone from
our IP phone, But I am unable to call my IP phone from
a PSTN phone (If I called any number between 21494350
and 21494399, the card should route my call to my IP
phone, IF my configuration was correct). I have done
my research and
Did you find a way to turn-on voice-mail lamp on Cisco phone connectedto Cisco Call Manager, when there is new voice mailin Asterisk mailbox?
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On 9/19/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Did you find a way to turn-on voice-mail lamp on Cisco phone connected to
Cisco Call Manager, when there is new voice mail in Asterisk mailbox?
Yes, this is described in the wiki document I created.
-Shaun
I'm getting these messages from this Asterisk version:
Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h
Sep 19 07:11:18 WARNING[99391]: chan_sip.c:2084
sip_new: Unable to allocate channel structure
Sep 19 07:11:18 NOTICE[99391]: chan_sip.c:7523
handle_request: Unable to create/find channel
When I kill the
Yes, but I'm asking for CM3.3 connected via H323.
I can set everything, including voice mail button, automatic forwarding to voice-mail... but only I didn't find a way to turn-on voice-mail lamp on Cisco phone (MWI)
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I don't have CCM4..., but if somebody know how I to get one...
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To UNSUBSCRIBE
Yes. I used a script that I believe was from Shawn and modified it to
suit my needs:
In /etc/asterisk/voicemail.conf:
externnotify=/etc/asterisk/vm.sh
vm.sh:
if [ $3 -gt 0 ]; then
# TURN LITE ON
CALLFILE=$(cat -EOF1
Channel: Local/11$2
MaxRetries: 1
# Retry in 2 min
RetryTime: 120
Hi all,
I'm currently doing a french canadian (quebec) translation of the
prompts. Almost all the about 140 default prompts are done, but there is
one I can't find...
In the directory, when the user found the good persor it press '1'. Then
Please whait while I try extension... prompt is played.
Since we are all trading secrets, check this site out
http://members.dandy.net/~czg/lca_index.php
I used to use this site DAILY when I owned an ISP. Now we have mostly
moved away from dialup, but I still use it once and a while.
Enjoy!
~kurth
___
Hi all,
I've been installing [EMAIL PROTECTED] and (of course) all the answering
machine (I don't sure that's the right word in english,
preatendedora in spanish) speech is in enlgish languaje.
Is there anyway to download all those .gsm files speaked in spanish?
Or may be another site which
Hi,
- Original Message -
Since we are all trading secrets, check this site out
http://members.dandy.net/~czg/lca_index.php
You can get this Perl scripts that extract NPA-NXX directly from
dandy.net...
http://www.voip-info.org/tiki-index.php?page=ScopServ%20LCA%20Map
--
Joel
Hi All,
I was wondering if there's a way to clear a SIP channel ('sip show
channels')
thru the asterisk CLI?
kind regards,
Bart
Bart van Daal
Network Operations
Van Landeghemstraat 20
9100 SINT-NIKLAAS
[EMAIL PROTECTED]
www.edpnet.be
T +32 (0)3 265 67 00
F +32 (0)3 265 67 01
On Sun, Sep 18, 2005 at 01:09:40PM -0600, Joseph wrote:
How to implement call limitation based on amount of time call per day.
Something that is not very accurate:
check once in a while based on the number of records in the CDR. There
should be ways to make that check cheap.
More accurate:
According to our tests, the Polycom 301 and 501 both
failover instantly using DNS SRV records. The grandstream BT101 and
GXP-2000 also failover using DNS SRV, but only upon re-registration. This
requires that you set the registration interval to a very small value. Has
anyone gotten the
On 9/19/05, Sebastian Milioto [EMAIL PROTECTED] wrote:
Hi all,
I've been installing [EMAIL PROTECTED] and (of course) all the answering
machine (I don't sure that's the right word in english,
preatendedora in spanish) speech is in enlgish languaje.
Is there anyway to download all those .gsm
- Original Message -
Since we are all trading secrets, check this site out
http://members.dandy.net/~czg/lca_index.php
You can get this Perl scripts that extract NPA-NXX directly from
dandy.net...
http://www.voip-info.org/tiki-index.php?page=ScopServ%20LCA%20Map
On a
On 9/19/05, Alexandre Leclerc [EMAIL PROTECTED] wrote:
Hi all,
I'm currently doing a french canadian (quebec) translation of the
prompts. Almost all the about 140 default prompts are done, but there is
one I can't find...
In the directory, when the user found the good persor it press '1'.
I second that...it actually would be a good place
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tad
Heckaman
Sent: Sunday, September 18, 2005 9:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AstriCon
What have it outside the US
j/k
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giovanni
Miano
Sent: Monday, September 19, 2005 4:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AstriCon 2006 Location
AHHH HA! Yes that answered all of my questions.
Ok.. everything is right with the world, except I still can't get the
'pager' e-mail address to not be
[EMAIL PROTECTED]
I have a serveremail= string and that gets set for e-mails.. but I
don't see anything about a pager string. I set the
I wanted to post a follow-up to my message about the dlink dph-140s
configuration I was having problems with. It's is now working fine on
my aah system and has been for a week or two. I believe the problem
was releated to silence suppression settings (NAS) on the phone.
It now sounds great - to
Hi, I configured an asterisk box with 1
Digium Wildcard TE110P T1/E1 Card 0
I setup the jumper in e1 position.
my zaptel.conf :
defaultzone=it
loadzone=it
#gestione PRI
span=1,0,0,ccs,hdb3,crc4,yellow
bchan=1-15 # set this to 1-15,17-31 for E1
dchan=16 # set this to 16 for E1
bchan=17-31 #
Great Idea! I suggest Sydney :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k.
Creasy
Sent: Tuesday, 20 September 2005 12:02 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users]
On a related note, I wanted our phones to display city, st for the
caller-ID name in the event that none was provided.
Interesting code. What sort of memory does * take up when you load up
all those CLID values?
Nathan
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Sydney or Cannes
Australia is the place to be.
W. Kevin Hunt
CCIE #11841
MCSE, Linux+ SME
www.huntbrothers.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Younger
Sent: Monday, September 19, 2005 9:09 AM
To: 'Asterisk Users
Hi,
do you know what it means the following:
Call failed to go through, reason 3
I received it when I try to send a FAX and no one answer it.
Thank you
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A private call is a call that someone has specifically blocked. An
out of area or unknown call is simply a call that the caller-id
did not come through on correctly, for some reason.
On 9/18/05, Goran Dj. [EMAIL PROTECTED] wrote:
Is there any method to make difference between Hidden (Private)
Im thinking Tampa or Orlando Florida! Nice warm.. Granted you may
have to
dodge hurricanes.. But hay its worth it!
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-Original
Does it have good international connections ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k.
Creasy
Sent: 19 September 2005 15:01
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users]
I want to visit oz but can't find any $79 tickets. Only way to do this
is get enough people to fill a chartered widebody from western US.
W. Kevin Hunt wrote:
Sydney or Cannes
Australia is the place to be.
W. Kevin Hunt
CCIE #11841
MCSE, Linux+ SME
www.huntbrothers.com
-Original
this will be usefull for you
http://www.voip-info.org/tiki-index.php?page=Asterisk+multi-language
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international
if you have troubles downloading the files, email me, i have a backup.
Other option is to use a text to speech
Hi Bob,
Found the DHCP options but the phone won't use it :(
Thanks for the help,
Francois
On Sunday 18 Sep 2005 15:15, Francois Meehan wrote:
Hi all,
I have bought an Aastra 480i phone.
In order to configure the phone for using a TFTP server, I had to enter
the TFTP ip address
Thanks!
In my dialplan there was no rule for 6092991xxx
Michael
On 9/18/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Sun, 18 Sep 2005, Michael Stearne wrote:
Looking for 6092991xxx in from-broadvoice
Reliably Transmitting (no NAT) to 147.135.20.128:5060:
SIP/2.0 404 Not Found
Hay is very much not worth it
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian C.
Fertig
Sent: Monday, September 19, 2005 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] AstriCon 2006 Location
On Mon, 19 Sep 2005, Michael Stearne wrote:
Thanks!
In my dialplan there was no rule for 6092991xxx
Michael
That's what I thought. Glad you got it sorted out.
Steve
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So, you admit that you can do what you want using RealTime Static, but you
are just unwilling to do so. So, how is that a limitation if you 'can' do
it?
-Matthew
From: Urban [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
The airport here in Louisville is an International Airport but I think
there are very few actual international flights coming in/out here.
That probably doesn't really answer your question... :)
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I would imagine that there would be enough people wanting to go that
with some coordination that could be done. I would probably be
interested it just might make the travel cheap enough that I could go.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Try changing the Gecos field in /etc/passwd for the user Asterisk is running
as. This is normally the 5th field, and is used for user information. I
made mine just say PBX and now it says [EMAIL PROTECTED] on text
messages/pages. This is from a default CentOS install with sendmail
I thought this might interest a few
asterisk users. I dont use them so I have no idea about Adtrans
but I know a heap of people on this list swear by them.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: Monday, 19 September 2005
9:59 AM
Subject:
I think I configured the MeetMe right. Since I am using SIP for
inbound calls I followed the
instruction, for 2.6 kernel, from this web page:
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
When I call the MeetMe number I get the greeting to enter in your
conference room. I do and get
If anyone else is trying to use asterisk with the sound port AND use
something
else like mplayer my experience was asterisk BLOCKS the port.
I added a bug item this morning to suggest a parameter control in alsa.conf
and 1 line program change to chan_alsa.c of:
snd_pcm_nonblock(handle, 1);
Shaun Ewing a écrit :
On 9/19/05, Alexandre Leclerc [EMAIL PROTECTED] wrote:
Hi all,
I'm currently doing a french canadian (quebec) translation of the
prompts. Almost all the about 140 default prompts are done, but there is
one I can't find...
In the directory, when the user found the good
Simple test extension
exten = 14,1,Wait(1)
exten = 14,2,SayPhonetic(${CALLERIDNAME})
exten = 14,3,Wait(1)
exten = 14,4,SayDigits(${CALLERIDNUM})
exten = 14,5,Hangup
Works fine from spa2k extension on lan
Works fine calling broadvoice sip did
When I call voicepulse sip did I get the
I receive an invite from a vendor device..
U 2005/09/19 09:00:18.991139 66.185.96.32:5060 - 66.185.96.23:5060
INVITE sip:[EMAIL PROTECTED];userphone SIP/2.0..Via:
SIP/2.0/UDP voip.livewirenet.com:5060;branch=z9hG4bK6ee64f86
Max-Forwards: 70
To: 3036284320 sip:[EMAIL PROTECTED];tag=as60bbddbc
On a related note, I wanted our phones to display city, st for the
caller-ID name in the event that none was provided.
Interesting code. What sort of memory does * take up when you load up
all those CLID values?
I would think they'd be stored in the database, not in memory. However,
it
This is a wild guess, but maybe it means
no one answered!
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Il Neofita
Sent: Monday, September 19, 2005
8:22 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] TXFAX
Hi,
do
List,
Okay, here's one that has me stumped, and it might just be something simple.
Currently, we are setup so that when someone calls in and tries to reach
the operator / front desk, it rings several different phones in
sequence. (i.e. it rings the front desk for 15 seconds, then a guy down
Yes, I know that, but, how to distinguish between them at incoming call?
- Original Message -
From: Matt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: pon 19. sep 2005 16:22
Subject: Re: [Asterisk-Users] Differ
I think I configured the MeetMe right. Since I am using SIP for
inbound calls I followed the
instruction, for 2.6 kernel, from this web page:
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
When I call the MeetMe number I get the greeting to enter in your
conference room. I do
I think most people would be more interested if it was for a few extra
days and included a deal on lodging. There are lots of travel agencies
that are good at arranging such things. We don't need astricon. We can
start an international * users group and go just about anywhere enough
members
I know this is OT but. J
Not sure if you guys have ever come across www.xoops.org but its this amazing open
source cms platform.
Its extremely well developed with a heap of
innovations and has just undergone the latest revision to 2.2.3 with even
easier access to module development.
this happened to me on a cvs update, rebuilt a clean chan
capi cm and all is well.
Greg
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Voicomm
UserSent: Monday, September 19, 2005 3:29 AMTo: Armin
SchindlerCc: asterisk-users@lists.digium.comSubject: Re:
[Asterisk-Users]
I've been trying to diagnose why my server has a constant idle time of 90%
even when nothing is running.
After finally discovering what hi means in 'top' (it means hardware
interrupts) I find that this percentage always averages around 7-10%.
How can I find out what is causing this constant load
You could setup a dial plan to save the sould file directly to your
main menu. Maybe something like this:
exten = *11,1,Play(record_greeting)
exten = *11,2,Record(/path-to-asterisk-sound-files/your-main-menu:gsm)
exten = *11,3,Play(recording_finished)
Now I'm assuming you have recorded the
kurt x wrote:
exten = _15551232432,2,Meetme
exten = _15551232432,3,Hangup
Try exten = _155512324232,2,MeetMe(|Msicp)
Doug
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For those that have experienced low VM recording volumes when using
a Digium TDM04b (or similar analog pstn card), a work around has been
committed to cvs-head. Need some folks to test it; it doesn't seem
to work for me, but need some feedback from others to ensure the
work around is actually
i've tried it on both snom190 and eyeBeam none of them work.
nothing is changed in configs.
is there any success in making snom LEDs work on CVS HEAD?
thanks,
paradise dove
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Asterisk-Users
I'm looking to
upgrade my unit, and would like to not have to wait on our company's suppliers
to get back to me on it.
Thanks in advance
for any help
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Louisville would be great for me. I'd get home to see the folks. Lots
to do in Louisville, very reasonably priced.
Dallas REALLY works for me, I'm already here. Dallas is not the cheapest
to fly in and out of but there are LOTS of flights.
Vegas, been there, done that. Had a good time, but
On 9/19/05, Frank Tarczynski [EMAIL PROTECTED] wrote:
I'm new to asterisk and need some help with ideas to handle thisconfiguration question.I am trying to establish a termination point/DID number in another
country.I am currently running Asterisk CVS-HEAD.My foreign provideruses SIP and
The limitation is that it doesn't work on freebsd, probably due to
libiodbc...
That's a limitation, isn't it?
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Matthew Boehm
Envoyé : lundi 19 septembre 2005 16:42
À : Asterisk Users
Objet : Re:
Does your phone company not pass on to you UNKNOWN or PRIVATE in
the caller-ID field?
On 9/19/05, Goran Dj [EMAIL PROTECTED] wrote:
Yes, I know that, but, how to distinguish between them at incoming call?
- Original Message -
From: Matt [EMAIL PROTECTED]
To: Asterisk Users Mailing
Hi Sebastian,
Do you know Asterisk-ES.org?
It's an Asterisk Spanish Community forum and there you will find all
Asterisk voices translated on Spanish.
Speech is translated on spanish like 'discurso', 'charla',
'conversación' or 'locución'. :)
Sebastian Milioto wrote:
Hi all,
I've
Hi,
I know that SIP is using port 5060 for session initiation, but which port
does it use for audio ? is it dynamically assigned ?
Thanks,
Adrien
--
Adrien Laurent - CIO
www.modulis.ca
514-284-2020 ext 202
[EMAIL PROTECTED]
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I can't find solution anywhere. I googled and find people with the same
problem but there was no answers on how to fix this.
I have W6692 based PCI cards that uses hisax driver (card type=36).
Card is working fine under asterisk with i4l modem driver for incoming
calls. If I want to dial out
Matt Riddell [EMAIL PROTECTED] writes:
Is it possible that JACK creates an emulated alsa/oss layer for non
JACK connections?
I can now confirm that jackifying iaxcomm works. The kfusd and
oss2jack now works on linux-2.6.13.
http://fort.xdas.com/~kor/oss2jack/
I still have latency problems as
That is not a limitation of the Asterisk RealTime Architecture. That is a
limitation of libiodbc.
-Matthew
From: Olivier Taylor [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Mon, 19 Sep 2005 18:09:38 +0200
To:
Upon setting up and configuring the my extension.conf, meetme.conf and
following the instruction outlined at this web page:
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
I get the following errors when calling the meetme number.
Executing Wait(SIP/216.53.118.2-f41196e0, 1) in new stack
HooDaHek 0.5 has been released.
NEW AS OF VERSION 0.5
- Changed the format of the incoming call notification to be on one line, not
two.
- Changed how the script sleeps between call notifications -- it now goes and
outputs the line to everyone in the list, pauses, and then looks for a second
We have a few Asterisk boxes running under Fedora C3 with no issues.
Before we roll into full production mode, we're wondering if the gurus
prefer any particular Linux distribution?
There are pre-packaged Asterisk solutions out there based on WBEL,
CentOS and a few others, so there doesn't
On Mon, Sep 19, 2005 at 12:22:35PM -0400, Adrien Laurent wrote:
I know that SIP is using port 5060 for session initiation, but which port
does it use for audio ? is it dynamically assigned ?
RTP protocol is used for audio. Port range is defined in /etc/asterisk/rtp.conf
--
Stefan Tichy
I have been looking
for the firmware for the 12sp+ and 30VIP and have been unable to find it. Any
help in locating would be much appreciated.
Thanks===This email and any files transmitted with it are confidential and intended solely for the
On Mon, Sep 19, 2005 at 01:53:16PM -0400, kurt x wrote:
Upon setting up and configuring the my extension.conf, meetme.conf and
following the instruction outlined at this web page:
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
I get the following errors when calling the meetme number.
Hello, I'm a newbie to the asterisk system.
I'm trying to configure a dialplan so that when I use my IAXy it will
prompt me with an IVR and then send me off to different things like
dial and voicemail from that.
I've tried various combinations but I can't seem to get it to work properly. Here is
Any means of killing a .call file that is in progress?
James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx 75503
903-793-1956
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Asterisk-Users mailing list
I know that SIP is using port 5060 for session initiation, but which port
does it use for audio ? is it dynamically assigned ?
Its dynamically assigned on a per-call basis.
Asterisk assigns the port based on contents of rtp.conf.
Remote sip phones assign port numbers based on whatever the
out-of-area is displayed for calls that originate from LECs that have not
implemented caller id.
James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx 75503
903-793-1956
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent:
I'm attempting to implement an acceptable asterisk monitoring program.
One method is with a simple program i've written (Adapted from the perl
one in the wiki).
One problem with this solution is the timeout I expect a packet
response. Currently it's set to 1 second, and I still sometimes miss an
On Mon, 2005-09-19 at 13:43 -0500, jltaylor wrote:
Any means of killing a .call file that is in progress?
You mean once the call has begun? You prolly want to hangup the
call ...
asterisk -rx soft hangup callid
Or is there something else that you wanted?
--
Trixter
You are doing correct. But you have to explain what you want to do?
As per your second configuration, if you dial 1 then it will
ringPost exactly what you are trying to accomplish?
-Thameem
On 9/19/05, John Crowhurst [EMAIL PROTECTED] wrote:
Hello, I'm a newbie to the asterisk system.
I'm
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