On Wed, Jul 06, 2005 at 11:32:11PM -0400, Carlos Alperin wrote:
> That sound like the Spanish TV Show. Is a similar of MRTG?
>
> If that is the case, the problem is the SNMP module for Asterisk.
Why use snmp? you don't weant to minitor asterisk's snmp. You want to
monitor Asterisk. Either poll as
Remco Barende ha scritto:
Does this version of chan_sccp replace the version at sourceforge or
is this Yet Another Fork(tm) :)
It's a fork.
Sergio
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listin
I use Nagios to monitor lines. I use the check_asterisk script that you'll find floating around the place. I connect via the mgmt interface. Added to
nagios is nagiosgraph. This keeps historical RRD graphs of my line usage.
==
Rod Bacon
Empowered Communica
hello austin
how to install perl module
i m following
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
and i did
sudo perl -MCPAN -e shell;
install Config::IniFiles
install Crypt::CBC
install Crypt::DES
install Authen::Radius
any other help full link i m new to perl
JD Austi
> That is not going to work. Asterisk shouldn't be behind a NAT to get
> registration of boxes behind NAT.
I've done it, and it works. It is not a great situation though because
of the provisioning problem. Specifically, an IAX device behind NAT
has no way of getting its provisioning "out of the
Is anyone having issues with audio being passed inbound via Teliax?
Trying to isolate an issue here.
Thx,
-Rob.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or u
I have an application where calls come into an *box from a DID
provider, and may be transferred to a meetme conference on another
*box (the call is released by the first *box after transfer).
These are ulaw IAX channel calls, and if the source is from a Verizon
or Nextel mobile phone to the
I m using oh323 and i m receiving incoming calls at windows NetMeeing and at SJPhone from SIP & IAX softphones but what should i do to be able to call from NetMeeting or any H323 softphone .when i dial any extension... it starts OH323/R4096 and then fails and plays demo-congrates from defa
Julio Cesar Ody wrote:
Thanks all of you. I forgot to set the stun server,
bye
Ronald
So you mean your SIP server is on the outside network and your peers
on the private one?
If so, sound behind NAT is always an issue, because it goes through a
different port than 5060 udp (which is t
hi !
i have the following in my extensions.conf
exten => 2000,1,Wait(60)
exten => 2000,2,Hangup
When i dial '2000' from my phone, I see 'Wait' being called. After 60 secs,
I also se 'Hangup' being called. If I hangup the phone line before 60 secs
are over ('Wait' command is probably interrupt
Ok, so I realised I have to have realm in [global] which got me a step
closer
I think:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 203.83.240.5:5060;branch=z9hG4bK6e81d139
From: "Jason" ;tag=as16b3c667
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: M2 X-Lite rel
I wouldn't mind such a single message. It is really a new breed of
product that is not known to most of us. Correct me if I am wrong.
While there have been some discussions on HOP-ON's wifi phone of $39
that never came true, this may be a sound alternative for all of us.
On 7/7/05, Terry H. Gilse
Is it just me that sees the post above as spam?
If we (tinw) even consider buying stuph from spammers, then we are
encouraging them in their sociopathic behavior, and as a consequence they
will do more spamming.
What is the consensus here?
It is a product announcement for a new product that is
So you mean your SIP server is on the outside network and your peers
on the private one?
If so, sound behind NAT is always an issue, because it goes through a
different port than 5060 udp (which is the reg. port). I had the same
problem a few days ago. Posted the question to the list twice, and
fo
I just set my asterisk to the NAT and it works magically.
On 7/6/05, Blake Krone <[EMAIL PROTECTED]> wrote:
> Ronald I've been struggling with the same problem. So far I have set
> my asterisk server to be in the DMZ and still nothing with sound, no
> music on hold or anything. From what I'm readi
Well it's now working somehow magically, heh.
Must have been the DMZ settings instead of the port forwarding.
On 7/6/05, Blake Krone <[EMAIL PROTECTED]> wrote:
> forgot to include the list
>
> -- Forwarded message --
> From: Blake Krone <[EMAIL PROTECTED]>
> Date: Jul 6, 2005 9:0
forgot to include the list
-- Forwarded message --
From: Blake Krone <[EMAIL PROTECTED]>
Date: Jul 6, 2005 9:07 PM
Subject: Re: [Asterisk-Users] Re: Remote SIP Connections
To: dbruce <[EMAIL PROTECTED]>
Just had my brother connect from his time warner cable in minnesota to
my ade
"outgoing", "monitor" and "voicemail/default" are good enough.
Here you would loose all the voicemail stuff. So you need to re-record
busy and unavialble messages for each
mailbox user (if at all you had them before).
~Vamsi
On 7/7/05, Jeffrey Starin <[EMAIL PROTECTED]> wrote:
> 911 Help!
>
> I
That sound like the Spanish TV Show. Is a similar of MRTG?
If that is the case, the problem is the SNMP module for Asterisk.
It was made for use with UCD-SNMP, not for NET-SNMP. And my platforms are
all RH-9. That is why I never was able to made it work.
Then, tired to look for a miracle I decid
Hi. I am trying to connect to a SIP provider as a client but
cannot get asterisk to work the same way as the UA.
My Configs:
-
sip.conf
[70501956]
type=friend
host=taraba.net
username=70501956
password=
insecure=very
fromuser=70501956
fromdomain=taraba.ne
Ronald I've been struggling with the same problem. So far I have set
my asterisk server to be in the DMZ and still nothing with sound, no
music on hold or anything. From what I'm reading if your SIP clients
connect from behind a NAT/Firewall/Private LAN there are problems and
it doesn't work well o
Hello all, I HAD video working before I upgraded to 1.08 (latest
stable with Gentoo) and now it won't work. I just see noise bars and
not the video. I know the camera works as I can use it in other
programs such as AIM & Yahoo.
I have the following setup:
sip.conf
[general]
videosupport = yes
po
Ahh...sorry. Got this bit sorted now.
Kevin P. Fleming wrote:
Jason Frisch wrote:
sip.conf
[70501956]
type=peer
useragent="M2 X-Lite"
host=p10.taraba.net
username=12345
password=secret
qualify=yes
fromuser=12345
fromdomain=taraba.net
realm=taraba.net
Read the example sip.conf more closely
Ronald:
If you're using SIP and I'm reading your email correctly, you most
likely have some sort of NAT issue.
There have been tons of NAT etc. emails on the list; those may help.
-Andy
On 7/6/05, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> I have an asterisk box installed, but all connection
Might be simpler to do a plugin for BigBrother/BigSister
Carlos Alperin wrote:
Ok,
I got it. None use MRTG to track status & history on Asterisk.
Someone uses ARGUS?
Any other tool?
Someone track their lines?
HEL
_
On 7/6/05, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> I have an asterisk box installed, but all connections to outside of the
> private network do not have a sound.
>
> Can you give me a hint what it could it be?
>
> bye
>
> Ronald
Ronald:
If you're using SIP and I'm reading your email cor
Jeffrey:
Don't you hate it when that happens?? :)
Anyhow, here's what I've got in that directory:
[EMAIL PROTECTED] asterisk]# pwd
/var/spool/asterisk
[EMAIL PROTECTED] asterisk]# du -h
4.0K./voicemail/default/512/INBOX
8.0K./voicemail/default/512
4.0K./voicemail/default/500/INBOX
8.0
Hi,
Is there a way to restart the DISA to the enter phone number? For
instance, Bell Calling Cards let you hit # at any point which lets you
enter another number to call. This is useful to reduce the number of
digits dialed and to utilize per-minute calls.
I was not able to find anything on the web
> Okay, so I believe I've gotten rid of the IRQ conflict, and it's still
> not working.
>
> What concerns me is that even channels with absolutely nothing connected
> to the slots will say "answered."
>
> The log output looks like this:
> -- Executing Dial("SIP/2000-401a", "Zap/2/w9w155540148
I forgot to tell you one thing, you lost all your Voice Mail messages,
but not the configs for the VM, since thats kept in
/etc/asterisk/voicemail.conf
On 7/6/05, C F <[EMAIL PROTECTED]> wrote:
> Just make a backup of what you have in /etc/asterisk and run
> /usr/src/asterisk/make install
>
> On
Jason Frisch wrote:
sip.conf
[70501956]
type=peer
useragent="M2 X-Lite"
host=p10.taraba.net
username=12345
password=secret
qualify=yes
fromuser=12345
fromdomain=taraba.net
realm=taraba.net
Read the example sip.conf more closely; some of these fields cannot be
set on a per-peer basis, only in
Just make a backup of what you have in /etc/asterisk and run
/usr/src/asterisk/make install
On 7/6/05, Jeffrey Starin <[EMAIL PROTECTED]> wrote:
> 911 Help!
>
> I accidentially deleted all directories under /var/spool/asterisk
>
> I did use the backup facility not too long ago but cannot find t
Hi,
That would probably be a problem with nat.
Just read this on the wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+NAT+solutions
Best regards,
Stevanus
Ronald Wiplinger wrote:
I have an asterisk box installed, but all connections to outside of
the private network do not hav
How exactly are you thinking. So that a certain aah extension points to
it or so that once you have been verified you can call aah extensions?
Darren
Erick Weber V. wrote:
Hello:
Does anyone know how to incorporate astcc to aah so it will use amah
extensions.
Any help will be appreciate
I have an installation where one of the users claims that they have had
calls where they could hear audio and the other party could not.
I went to /var/log/asterisk and looked at messages and full, but they didn't
have much info in them.
I have:
full => notice,warning,error,debug,verbose
in logg
I have a single Vonage line for my primary outbound voice calls and fax
calls. Inbound it is only used as a fax line. I have not had any losses in
transmission or broken fax jobs yet. (3 months and counting)
This is on the primary ATA channel that I pay 24.95 per month for. So I
don't know abou
Yes.. I've checked that already. and that's why I'm asking the
list what would make an ATA and Softphones not send packets?!?!
It's not like they are being blocked.. the ATA just has decided not to
send anything.
On 7/6/05, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> Matt wrote:
>
>
Hi.
I am having trouble changing the User-Agent field in *.
sip.conf
[70501956]
type=peer
useragent="M2 X-Lite"
host=p10.taraba.net
username=12345
password=secret
qualify=yes
fromuser=12345
fromdomain=taraba.net
realm=taraba.net
Sip Debug log:
set_destination: Parsing for
address/port to sen
I choose not to acknowledge Sangoma's existence whenever possible.
I've had some very poor experiences with the quality of their cards,
firmware, and drivers, so I tend to /not/ recommend them.
-Bryce
On Jul 6, 2005, at 16:24, TC wrote:
this would be with a couple channel banks and a quad
On Wed, 2005-07-06 at 17:57 -0500, Jay Milk wrote:
> What does Teliax have to say about this? They have your money, make
> them earn it.
I called them, they have to return my call yet.
On my previous conversation with them they advised to play with the
iax.conf setting according to their tech-su
[EMAIL PROTECTED] dkwiebe]# cd /var/spool/asterisk
[EMAIL PROTECTED] asterisk]# ls -R
.:
CVS fax monitor outgoing qcall tmp vm voicemail
./CVS:
Entries Repository Root
./fax:
CVS fax.call
./fax/CVS:
Entries Repository Root
./monitor:
CVS
./monitor/CVS:
Entries Repository Root
.
Jeffrey Starin wrote:
> 911 Help!
>
> I accidentially deleted all directories under /var/spool/asterisk
>
> I did use the backup facility not too long ago but cannot find the
> process for restore.
>
> However, I don't believe a full restore is needed -- I just need to know
> the names of the di
Hello:
Does anyone know how to incorporate astcc to aah so it will use amah
extensions.
Any help will be appreciate
Thanks
Erick W.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asteris
Jeffrey Starin wrote:
However, I don't believe a full restore is needed -- I just need to know
the names of the directories under /var/spool/asterisk and re-create
them (I hope!). Can some kind soul give me some direction or tell me
the directory structure under /var/spool/asterisk?
Do a 'mak
My /var/spool/asterisk has the following directories
drwxr-xr-x2 asterisk asterisk 4096 Jul 5 10:55 fax
drwxr-x---2 asterisk asterisk 4096 Jul 4 18:53 monitor
drwx--2 asterisk asterisk 4096 May 11 17:10 outgoing
drwxrw2 asterisk asterisk 4096 Mar 31 05:26
I will be installing an Asterisk system in Honduras, and I need to
convert many 2-wire analog circuits to E1 PRI. I have no idea if there
is answer or disconnect supervision on the POTS circuits.
An E1 from Hondutel seems to be out of the question because they will
only provide SS7 signaling
Bryce Chidester wrote:
Assuming you mean you have 30 analog POTS lines, the way to go about
this would be with a couple channel banks and a quad-T1 (I haven't seen
a two-port around, but that's all that is needed) card.
For the record, 30 individual analog lines is generally inefficient and
I found an unanswered mail in the archives that implied that perhaps
there is no direct way to delete a DB entry with the new "Set" syntax.
I have been playing, and so far if there is a way to do it I haven't
been able to suss it out.
Set(DB(family/key)=) sets the value for the key to null, b
Come on now children. Is this not a place to share knowledge?
Jimmy Smith wrote:
coudnt agree more.. thats exact thing i was saying the other day..
please hold my di..k while i take a leak i don't want to wet my hands.
RTFM, google and test. || Pay
On 7/6/05, Brian West <[EMAIL PROTECTED]>
T1/PRI and/or a channel bank -- you'll find more about this if you
google this list.
> -Original Message-
> From: Angel Diaz [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, July 06, 2005 2:28 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Connect 30 phone lines to aster
When I call from a Pulver communicator to my asterisk box, the caller
should hear the announcement. However, the announcement is only played
for a few seconds and the call is terminated.
bye
Ronald
___
Asterisk-Users mailing list
Asterisk-Users@lis
I would like to make some tests with video phones. If anybody on the
list has time for some tests. I would appreciate if you could send me a
private email to ronald_wiplinger (at) yahoo . com
I would like to test with different phones.
bye
Ronald
__
Matt wrote:
Hi,
I've just setup a second asterisk system and am having a wierd
problem. If I am on the same network as the system it works fine...
if I am outside coming in from the internet through a NAT I get the
following:
I can place calls I can hear the asterisk system. The asterisk
Hurray! Thanks for this new release!
I've been plagued by segfaults for a long time, hope this is now solved.
Does this version of chan_sccp replace the version at sourceforge or is
this Yet Another Fork(tm) :)
Thanks!!
On Wed, 6 Jul 2005, Sergio Chersovani wrote:
http://chan-sccp.berlios
Ok,
I got it. None use MRTG to track status
& history on Asterisk.
Someone uses ARGUS?
Any other tool?
Someone track their lines?
HEL
Mensaje analizado y p
Angel,
The neatest way is to use an E1 to a channel bank. This splits the E1
into 30 analogue lines.
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
Angel Diaz wrote:
Hi,
I have to connect 30 phone lines to my asterisk server, can somebody
help on how I hav
775.219.9924 but none answer. Only a Voice Mail, but they have numeric
paging.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Wednesday, July 06, 2005 5:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Use
> this would be with a couple channel banks and a quad-T1 (I haven't
> seen a two-port around, but that's all that is needed) card.
not sure how hard you look :)
http://www.sangoma.com/products/p_aft-et1-specs.htm
2 T1 spans with daughter board, and an adit 600 with 4 8 port fxo cards
would fit
I have an asterisk box installed, but all connections to outside of the
private network do not have a sound.
Can you give me a hint what it could it be?
bye
Ronald
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.
Ramin Nikaeen wrote:
When I restart asterisk, I believe I should be able to see the asterisk
listening
on port 5038.
What could be the problem?
My manager configuration is:
---
What does Teliax have to say about this? They have your money, make
them earn it.
> -Original Message-
> From: Joseph [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, July 06, 2005 2:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Incoming 800-
Obelix,
Either is fine. Examples:
AGI(subdir/yourscript)
AGI(/path/to/yourscript)
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
Obelix wrote:
I want to organize my agi scripts by putting them in separate subdirectories.
Is this permissible, or it necessary fo
911 Help!
I accidentially deleted all directories under /var/spool/asterisk
I did use the backup facility not too long ago but cannot find the
process for restore.
However, I don't believe a full restore is needed -- I just need to know
the names of the directories under /var/spool/asterisk an
Sometimes for me unknown reasons a wakeup call cannot delivered to a
phone and ends up in the voice mail box (and consequently sent via email
to the phone user).
It would be nice to find the reason why the phone was not reachable, but
for sure it is useless to send a wakeup call to the mailbox
Angel Diaz wrote:
Hi,
I have to connect 30 phone lines to my asterisk server, can somebody
help on how I have to do it ?
I have a TDM405P and one TDM400P with 4 FXO ports.
Do I have to use 8 TDM400P ? Or, is there another way to do it ?
You buy a channel bank with FXO ports and plug it into
Running latest cvs here's pri debug's output
[Span 3 D-Channel 0]< Protocol Discriminator: Q.931 (8) len=47
[Span 3 D-Channel 0]< Call Ref: len= 2 (reference 135/0x87) (Originator)
[Span 3 D-Channel 0]< Message type: SETUP (5)
< [04 03 80 90 a2]
[Span 3 D-Channel 0]< Bearer Capability (len= 5) [
Valued Colleagues,
I am trying to configure and use asterisk manager API.
The /etc/asterisk/manager.conf and the output of “netstat
–nl” are appended below.
When I restart asterisk, I believe I should be able to see
the asterisk listening on
port 5038 using netstat. But when I
Andrew Sayman wrote:
Okay, so I believe I've gotten rid of the IRQ conflict, and it's still
not working.
What concerns me is that even channels with absolutely nothing connected
to the slots will say "answered."
The log output looks like this:
-- Executing Dial("SIP/2000-401a", "Zap/2/w9w15
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> IM.Nobody
> Sent: Wednesday, 6 July 2005 11:51 PM
> To: Asterisk-Users@lists.digium.com
> Subject: [Asterisk-Users] DECT VoIP Gateway
>
> Hi all,
>
> Just want to share with all of you a new hot
coudnt agree more.. thats exact thing i was saying the other day..
please hold my di..k while i take a leak i don't want to wet my hands.
RTFM, google and test. || Pay
On 7/6/05, Brian West <[EMAIL PROTECTED]> wrote:
> Why not do your research instead of asking the list to do it for
> you l
Andrew Sayman wrote:
Okay, so I believe I've gotten rid of the IRQ conflict, and it's still
not working.
All analog FXO ports are considered "answered" when the dialing is finished.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
___
Asterisk
Hi
When I make a call from the outside to asterisk and the call is asnwered,
all is OK, but when I make a call from the outside to asterisk and hangup
before the call is answered by a astesrisk extension, you can see in the
console this message
Jul 6 19:33:08 WARNING[10037]: channel.c:1521
Andrew Sayman wrote:
Elwin Andriol wrote:
Don't know if this will help you any further, but. After some trouble
with IRQ sharing mayhem we solved our little problem by tinkering the
linux kernel. I forgot the names of the actual modules, but after
disabling modules for APIC suppor
Assuming you mean you have 30 analog POTS lines, the way to go about
this would be with a couple channel banks and a quad-T1 (I haven't
seen a two-port around, but that's all that is needed) card.
For the record, 30 individual analog lines is generally inefficient
and would be done more clean
Hi,
I've just setup a second asterisk system and am having a wierd
problem. If I am on the same network as the system it works fine...
if I am outside coming in from the internet through a NAT I get the
following:
I can place calls I can hear the asterisk system. The asterisk
system does no
Well without valgrind
Running asterisk from gdb (sorry a bit of a newbie with linux debugging)
When doing a backtrace on the crash I get the following:
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 213005 (LWP 9886)]
0x08074ef0 in ast_extension_match ()
(gdb) bt
#0 0x0
hallo,
i just experienced that all meetme rooms share the same voice data, if i
connect to 499, it could be heard in all other rooms (498,500, 501)
could sombody help me, why does asterisk send the voice out of all rooms if
i only connect to one?
thanks for help,
alex
meetme.conf
conf =>
i can suggest using wavepad.
its on the voipinfo site
On 7/6/05, mohammad <[EMAIL PROTECTED]> wrote:
>
> HI ALL;
>
>
> I have problem converting a windows .wav file to .gsm format by Sox.
> Could anyone help.
>
>
> Cheers,
> Mohammad
>
>
Joseph wrote:
On Wed, 2005-07-06 at 14:22 -0500, Matthew Boehm wrote:
Chris Gamble wrote:
How adventurous would a person have to be to try to use the 1.1 from cvs? I
want to implement our phone system with the database connections built in,
which as I understand is being made very easy in t
E1 or T1 card???
Regards,
Marc
Angel Diaz wrote:
Hi,
I have to connect 30 phone lines to my asterisk server, can somebody
help on how I have to do it ?
I have a TDM405P and one TDM400P with 4 FXO ports.
Do I have to use 8 TDM400P ? Or, is there another way to do it ?
Thanks,
Angel.
I am getting:
NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on
Primary D-channel of span 1
on my asterisk box and it seems to be causing a poping sound in the
phones, I am wondering if anyone can shed some light on this. I have
scanned the archives and get possibilities ranging f
Hi,
I have to connect 30 phone lines to my asterisk server, can somebody
help on how I have to do it ?
I have a TDM405P and one TDM400P with 4 FXO ports.
Do I have to use 8 TDM400P ? Or, is there another way to do it ?
Thanks,
Angel.
___
Asterisk-U
I want to organize my agi scripts by putting them in separate subdirectories.
Is this permissible, or it necessary for at list the initial file to be in the
agi-bin directory?
In case I prefer to move them outside the main folder what syntax should I use
for the folder? will it be worked off the
does cdrtool handle 800 termination from different src ?
from the page they say
"Combined rating based on traffic, duration, application type and destination"
so not from src it seems..
anyone got that working ?
Example : billing depending on src number + destination #
JV
___
does cdrtool handle 800 termination from different src ?
from the page they say
"Combined rating based on traffic, duration, application type and destination"
so not from src it seems..
anyone got that working ?
Example : billing depending on src number + destination #
JV
___
It looks like this page has a phone number to someone that works for them:
http://www.prweb.com/releases/2005/3/prweb213673.htm
When I called it sprintpcs VM came up, but after ringing for a while
which makes me think that the phone is on.
On 7/6/05, Storm D. J. Petersen <[EMAIL PROTECTED]> wrote:
It's complaining that you don't have the perl module installed or it is
not in your path.
Kamran Ahmad wrote:
hello
how to solve these errors
/var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10
use Asterisk::AGI;
vi /etc/asterisk/extensions.conf
exten =>
_X.,1,agi,agi-rad-auth.pl|Routing=SIP
Okay, so I believe I've gotten rid of the IRQ conflict, and it's still
not working.
What concerns me is that even channels with absolutely nothing connected
to the slots will say "answered."
The log output looks like this:
-- Executing Dial("SIP/2000-401a", "Zap/2/w9w15554014809") in new stac
this happens to me too.
On 7/6/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Wed, Jul 06, 2005 at 11:48:27AM -0500, Brian West wrote:
> >
> > /b
> > ---
> > Anakin: "You're either with me, or you're my enemy."
> > Obi-Wan: "Only a Sith could be an absolutist."
> >
> > On Jul 5, 2005, at 12:02
Hi,
if I understand right, the best way to indicate a PSTN
line "busy", is something like that in extension.conf:
... background(busy-tone)
... busy
So the caller will first hear "my" busy-tone, and
after some seconds, when PSTN honours the busy
indication (cmd busy), he hears the busy sound by
Bartosz Jozwiak wrote:
Is anybody there using quadBRI form Junghanns.net with Asterisk ?
I would like to order that card but first would like to hear some
opinions.
I have been testing out with chan_capi and some cheap PCI cards in the
past, but since we've used the quadBRI cards, ... I don't
On Wed, 2005-07-06 at 14:22 -0500, Matthew Boehm wrote:
> Chris Gamble wrote:
> > How adventurous would a person have to be to try to use the 1.1 from cvs? I
> > want to implement our phone system with the database connections built in,
> > which as I understand is being made very easy in the 1.1
[root at pbx sbin]# ps ax | grep asterisk
3371 ?S 0:00 /bin/sh /usr/sbin/safe_asterisk
3417 ?S 0:00 asterisk -vvvg -c
6846 ?S 0:00 asterisk -vvvg -c
6848 ?S 0:00 asterisk -vvvg -c
6849 ?S 0:00 asterisk -vvvg -c
6850 ?S
On Wed, Jul 06, 2005 at 12:42:52PM -0500, Andy Brezinsky wrote:
> Console Output:
>-- Accepting call from '414944' to '80094042XX' on channel 2/24,
> span 4
>-- Executing Wait("Zap/48-1", "3") in new stack
>-- Executing Answer("Zap/48-1", "") in new stack
>-- Executing Playback
I have been unable to get DISA to work when dialing from
pstn into a zap interface.
I have the same context configured for an incoming IAX interface,
and it works fine.
I’m using a tdm400p. Others on IRC have gotten this
to work on the x100p.
Will it work on the tdm400p? Anyone
hello
how to solve these errors
/var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10
use Asterisk::AGI;
vi /etc/asterisk/extensions.conf
exten =>
_X.,1,agi,agi-rad-auth.pl|Routing=SIP&AuthorizeBy=SIP
vi /etc/asterisk/modules.conf
load => res_agi.so
<---errors>
On Wed, Jul 06, 2005 at 10:40:22PM +0430, mohammad wrote:
> HI ALL;
>
>
> I have problem converting a windows .wav file to .gsm format by Sox.
> Could anyone help.
We could if we knew what your problem were.
The command I use in one of my scripts:
sox $wav_file -r 8000 -c 1 -s -w $gsm_file
My apologies for any bounced mail from me today. My mail server was
having a bit of a fit.
MARK.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options v
Did you try tailing the /var/log/dmesg to see what happened when you
loaded zaptel and wctdm with modprobe?
Check that /etc/modprobe.conf still contains the correct module entries.
Does /lib/modules/2.6.11-1.35_FC3smp/misc still contain and correct
wctdm.ko files?
-Original Message-
F
Elwin Andriol wrote:
> Don't know if this will help you any further, but. After some trouble
> with IRQ sharing mayhem we solved our little problem by tinkering the
> linux kernel. I forgot the names of the actual modules, but after
> disabling modules for APIC support and something about IRQ shar
1 - 100 of 209 matches
Mail list logo
