I just got it working with a 2.6 kernel without any problems,
although I'm having a little bit of trouble setting the callerid on it
Keith
On Jul 1, 2005, at 11:00 PM, Michael Jia wrote:
Hi,
In digium website.
http://store.yahoo.com/asteriskpbx/newitastdmde.html
It is said Dev Kit PCI
Eric Wieling aka ManxPower wrote:
Kevin P. Fleming wrote:
Tamas J wrote:
Does anybody know what will be the around price for the announced E3
card from Digium? When is it planned to be ready?
Pricing and release date have not been announced at this time.
The product has also been
On 07/02/05 02:15 Matthew Boehm said the following:
according to the wiki, I should be able to do this:
exten = _9./3003,1,Set(CALLERID(number)=281443)
exten = _9./3004,n,Set(CALLERID(number)=281444)
exten = _9./3005,n,Set(CALLERID(number)=281445)
exten =
On 2 Jul 2005, at 08:48, Dinesh Nair wrote:
On 07/02/05 02:15 Matthew Boehm said the following:
according to the wiki, I should be able to do this:
exten = _9./3003,1,Set(CALLERID(number)=281443)
exten = _9./3004,n,Set(CALLERID(number)=281444)
exten =
I have googled this to death, and all I get are reference to the MoH
needing a Zaptel timing source, and then people saying no they don't
any more.
-- Set Response Timeout to 2
-- Executing BackGround(SIP/211-57ba, my-greeting) in new stack
-- Playing 'my-greeting' (language
IPSwitchBoard Version 0.121 - 02 July 2005
* Extensions can be added to speed dial number. This can be used to dial
speed dial numbers from any phone connected to your asterisk system. This
requires that you configure your dial plan to take advantage of this
feature. See sample Dial Plan in the
On Friday 01 Jul 2005 16:43, Zoltan Szecsei wrote:
Bob Goddard wrote:
On Friday 01 Jul 2005 15:14, Zoltan Szecsei wrote:
Hi Bob,
Thanks - I'll run with the README idea of yours.
Your comment regarding re-boot however is not valid. I also thought that
was the case and (as I said on the first
Today we have been having some problems with the dchannel of out T1's.
I was wondering if there is a way for asterisk to send out an email or
page whenan error occurs. Not I know errors happen quite offen for many
reasons, but I would like an email sent when there is a TI problem, or
i dont know how to edit the time 3ms for ringing
in astcc when it says there is no body to answer.i
want to change this time to 4ms but i dont know
how.please help please.
__
Yahoo! Mail
Stay connected, organized, and protected. Take
hello
i m trying to use radius with asterisk
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
how to fix this patch
8. Make sure that your Asterisk includes all related
bug fixes and patches, namely:
- SIPGetHeaders for chan_sip (derived from chan_sip2 )
i m using
On Saturday 02 July 2005 02:04, Keith Caldwell wrote:
I just got it working with a 2.6 kernel without any problems,
although I'm having a little bit of trouble setting the callerid on it
You can't set outgoing callerID on analog PSTN lines. This is a telephone
network limitation, not a
On Saturday 02 July 2005 02:04, Keith Caldwell wrote:
I just got it working with a 2.6 kernel without any problems,
although I'm having a little bit of trouble setting the callerid on it
You can't set outgoing callerID on analog PSTN lines. This is a telephone
network limitation, not a
On Saturday 02 July 2005 07:57, Rich Adamson wrote:
Think he's trying to set it on a TDM-fxs module (not fxo). Or did I miss
something.
Nope I am probably the one who is missing it, it's early here. :-)
-A.
___
Asterisk-Users mailing list
Looking for some advice from vegastream users if possible?
I am having a nightmare trying to find the best settings for G729 and
G723.1 codecs. My users are using Snom phones. Any recommendations as to
the best codec settings would be very appreciated as trial and error is
proving long and
On Fri, Jul 01, 2005 at 09:53:33AM -0700, Wiley Siler wrote:
Anyone know a good distro for an Epia Mobo with the C3 chip?
Debian, as for any hardware :-p
I have been trying to get Debian and Gentoo installed (new to me) and so
far having little luck.
Debian i386 packages should
Bharat M. Sarvan wrote:
Hello everybody,
I have made a application of my own. (I.e. Def ( )). I am able to
compile the application successfully. And the .so file is created as
well. But when I load asterisk I get the following error.
[Def.so]Jul 1 19:20:06 WARNING[15664]: loader.c:295
I have experienced a * problem with all forwarded calls where the inbound
caller cannot hear any audio for 2-4 seconds after the forwarded call is
answered, causing the caller--who cannot hear anything--to think there is no
connection and thus hangs up. If the caller waits a couple of seconds,
Problem resolved with Astcc, certified fully working.Juan Luis Moyano [EMAIL PROTECTED] wrote:
Ade Agbero wrote: Finally, We have lift off, a shaky one though. I deleted my Astcc.gi and replaced it with Darren's copy posted on his website and I have finally been able to get something recorded as
Hi People!
Having interesting issues with app_addon_sql_mysql.c:
[EMAIL PROTECTED]:/usr/src/asterisk-addons# make
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include
-I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c
app_addon_sql_mysql.c:164:64: macro
List Receiver wrote:
Having used Broadvoice for a while with marginal service, I want to
move on to another provider. So my question to the List is who is
good? I know now one service is perfect but somebody out there has to
be decent. Who have you guys had the best luck with?
I suggest,
The system startup script /etc/init.d/asterisk calls the script
/usr/sbin/safe_asterisk
In safe_asterisk, the program is started with -c by default (console on
TTY9).
That explains why it is starting with a console, but not why it's
running so many times! Here is what my system (FC3) shows:
Woah woah woah.. why not just disable SBC voicemail and have asterisk
handle it? I don't understand why you would go to such great lengths
when you can just have Asterisk deal with it.
On 7/1/05, andrew matthews [EMAIL PROTECTED] wrote:
Is there alternative access to voicemail? Like web access?
Hello All ,
On Sat, 2 Jul 2005, Michael Stahl wrote:
The system startup script /etc/init.d/asterisk calls the script
/usr/sbin/safe_asterisk
In safe_asterisk, the program is started with -c by default (console on
TTY9).
That explains why it is starting with a console, but not why it's
Just get one of these.
The PCI 921-CDS is a low-cost channelized DS3 WAN adapter that can be
used in ImageStream's Industrial Series routers or OEM products
running Linux. The PCI 921-CDS can individually address all of the
DS0s and T1s in a DS3, and it can be used in a wide range of
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jon Radon
Sent: Saturday, July 02, 2005 10:49 AM
To: andrew matthews; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] passing through MWI info
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tom Rymes
Sent: Friday, July 01, 2005 11:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AMP/[EMAIL PROTECTED] (asterisk at home) custom
I m new to asterisk n i've got an IP phone that supports h323 protocol but i dont know how to configure asterisk to use it... i m comfortable in using sip iax softphones butthere is no h323.conf in /etc/asterisk/ i read that i've to compile some files but i m confused regarding h323
IAX doesn't use INBAND DTMF.
Denis Galvão.
On 01 de jul de 2005, at 03:23, Mark Edwards wrote:
Hi.
Probably been asked before, but my IAX provider assures me its not
their problem
I have a IAX connection to a peer providing a DID. I am dialing up
my number, seeing the DTMF tones
Sorry I was not available yesterday. It was Canada Day and we got to
celebrate Alberta's centenial. What did you wind up doing to get it
working?
Darren Wiebe
[EMAIL PROTECTED]
Ade Agbero wrote:
Problem resolved with Astcc, certified fully working.
*/Juan Luis Moyano [EMAIL PROTECTED]/*
Manjit Riat wrote:
I did a make webvmail and I get the following error on redhat 9.0
No HTTP directory
make : *** [webvmail] Error 1
I have the perl-suidperl rpm installed and apache installed
Thanx .
The webvmail make script isn't terribly intelligent about
I've been doing some reading on ENUM and am almost ready to start
testing with it. However, I have a question.
As I understand things the following ENUM entry would return info for
all telephone numbers of any length beginning with 00393. The Asterisk
pattern would be _00393. (notice the
Hello there,
I'm a new Asterisk user and I wonder if it is possible to associate a
voicemail box with a group of users, i.e., a single recorded message is
sent to everyone in that group. If so, where can I find more
information about that?
Thanks in advance,
Leo Burd
Hi Leo,
here's a suggestion:
in your dialplan (extensions.conf) send multiple users to the same
mailbox (e.g. 999) if they do not pick up within 30 seconds:
; SIP Phone 100, Tom
exten = 100,1,Dial(SIP/100,30)
exten = 100,2,VoiceMail(999)
; SIP Phone 200, Eric
exten = 200,1,Dial(SIP/200,30)
Hello,
i try to do the following:
1) call comes in on extension 999
2) caller should hear music (NOT MoH!!!)
3) a call should be initiated to SIP Phone 100
4) when SIP Phone 100 is answered, a sound file should be played to the
user at SIP Phone 100
5) the incoming call (at extension 999) should
Hi Adeel
http://www.inaccessnetworks.com/projects/asterisk-oh323
Please visit there, you will find
your way.
Bashir
- Original Message -
From:
Adeel -31
To: asterisk-users@lists.digium.com
Sent: Saturday, July 02, 2005 9:13
AM
Subject: [Asterisk-Users] What
exten = 1234,5,Voicemail(u,1234234534564567)
As you can see the same voicemail will go to all the users.
On 7/2/05, Leo Burd [EMAIL PROTECTED] wrote:
Hello there,
I'm a new Asterisk user and I wonder if it is possible to associate a
voicemail box with a group of users, i.e., a single
On Jul 2, 2005, at 10:23 AM, Roland Zagler wrote:
Hello,
i try to do the following:
1) call comes in on extension 999
2) caller should hear music (NOT MoH!!!)
3) a call should be initiated to SIP Phone 100
4) when SIP Phone 100 is answered, a sound file should be played to the
user at SIP
try this one
exten = 999,1,Answer()
exten = 999,2,playback(~.mp3)
exten = 999,3,dial (sip/100)
exten = 999,4,playbackground(~.mp3)
exten = 999,h,Hangup()
not sure abt playbackground should be before the dial command or after
From: [EMAIL PROTECTED] on
hello all,
i need some help and after trying the wiki i'm even more confused than i was.
i'm trying to set up call forwarding and running into problems...
i want the most basic call forwarding imaginable.
1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's
On Jul 2, 2005, at 12:55 PM, Mahmoud Badran wrote:
try this one
exten = 999,1,Answer()
exten = 999,2,playback(~.mp3)
exten = 999,3,dial (sip/100)
exten = 999,4,playbackground(~.mp3)
exten = 999,h,Hangup()
not sure abt playbackground should be before the dial command or after
Mahmoud:
Thank you, Robert!
The announcementfile plays well, but at Dial-option m i have to
specify a MoH class,
that is something i cannot use (as i wrote in my post).
Background command waits for a user input, but the caller should be
connected to
SIP Phone 100 after it has answered and the
You can also set anything you wish into the CDR variables. We came
up with the whole CDR variable thing for this exact purpose. Check
cdr_custom to log it like you want.
ie Set(CDR(GROUP)=${GROUPCALL})
/b
PS don't for get to come to cluecon!
On Jun 30, 2005, at 4:15 AM, Chris
On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote:
Thank you, Robert!
The announcementfile plays well, but at Dial-option m i have to
specify a MoH class,
that is something i cannot use (as i wrote in my post).
Noted, which is why I offered option two.
Background command waits for a user
On Fri, Jul 01, 2005 at 12:15:06PM -0400, Michael Stahl wrote:
You should be able to do a good job with IPTABLES which is included in
FC3. You can limit source destp IP and protocol, etc.
Type man iptables | more for more details...
Which will not get you anywhere. There are a number of
Denis Galvão - iSolve wrote:
IAX doesn't use INBAND DTMF.
Denis Galvão.
Denis,
A clarification, I hope, just to make Mark aware of the small difference.
IAX sends DTMF in the signaling 'stream', that happens to follow the
same path as the media.
But, in IAX DTMF is not sent as voice payload
On Sat, Jul 02, 2005 at 09:13:47AM -0700, Adeel -31 wrote:
I m new to asterisk n i've got an IP phone that supports h323 protocol
but i dont know how to configure asterisk to use it...
There are currently a number of options for h323 support:
- The original chan_h323 . Works only with
sorry for the misunderstanding, robert!
the point is: during the caller is listening to the soundfile played to
him
the dialplan should continue to dial the sip phone 100 and after sip
phone
100 is answered and the announcement file is played the caller should be
connected
to the sip phone 100.
In the subdivision where I live, we have a well that time to time has
problems. Currently, our trustees call me, I take a message, and
then call the people on our phone calling tree and give them the
message. They, in turn, pass the message on to the rest of the
residents.
We have a
Hi Scott, everything thing you are looking to do is possible.
You might need to offer a bounty for someone if you don't feel
comfortable to do it yourself but checkout the call agi scripts.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network
adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP:
___ HOME ___ OFFICE
SPA2000 --- Linux Box -- Asterisk Box
192.168.0.253192.168.0.1 eth1 200.93.xxx.a
If you do add a bounty, I'll add a little bit $50US or so to it. I know
there are others that have written or would like a script like this as
it was discussed a few months ago on this list I believe.
Darren Wiebe
[EMAIL PROTECTED]
Dean Collins wrote:
Hi Scott, everything thing you are
Scott Nelson wrote:
In the subdivision where I live, we have a well that time to time has
problems.
How about just fix the well :-)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I used the astcc.agi you posted on your website,and changed the"BRANDS" INCvalue from 6 to 60.
So, I guess all you need to do now is incorporate your astcc.agi file into the CVS version.
Thanks goes to you and those who contributed and responded to my HELP HELP HELP plea.
From a satisfied ASTCC
Hi folks,
when I start asterisk directly, I get a colored CLI. When connect to a
already running asterisk with asterisk -R, it's never colored, despite
I'm running both from the same console (tty). Is there a way to force
asterisk -R into color mode?
Regards,
Stefan
--
(o_ Stefan
I've been reading a bit about Enum and DUNDi and still have something
not very clear to me.
This is a HYPOTHETICAL scenario:
I have 4 asterisk servers. All of them are handling registrations of
both SIP and IAX2 UAs. SIP agents are being load balanced by
something like SER. I have another
Try this
http://www.voip-info.org/tiki-index.php?page=Asterisk+Tips+follow+me
I used and it works well. Rather than segregate calls based on caller ID, it
carries the caller's ID through to the forwarded phone (cell phone, or
other?), but inserts a 0 before the number, that way you know it is an
Asterisk -gc
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Stefan
Gofferje
Envoyé : samedi 2 juillet 2005 22:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [Asterisk-Users]
Carlos,
Thank you for your fast response :) , this is the output of iptables -nL
on my linux box:
[EMAIL PROTECTED]:/home/guillermo # iptables -nL
Chain INPUT (policy ACCEPT)
target prot opt source destination
Chain FORWARD (policy ACCEPT)
target prot opt source
Guillermo,
I'm not very expert with iptables, but this is the issue:
I don't see the forwarding from the ip of the sipura box ( that should be
the only one to receive both UDP RTP traffic on the 5060 16384 to 32767
ports. On the other hand, the Asterisk box is also in an fix ip, so the
traffic
Jon Radon wrote:
Woah woah woah.. why not just disable SBC voicemail
and have asterisk
handle it? I don't understand why you would go to
such great lengths
when you can just have Asterisk deal with it.
Jon, etc..., the issue here is her family all uses
special features of SBC voicemail. E.g.
[EMAIL PROTECTED] schrieb:
Asterisk -gc
I don't see a -R in that...
Regards,
Stefan
--
(o_ Stefan Gofferje | Linux Systems Specialist
//\ Reg'd Linux User #247167 | Network Security Specialist
V_/_ Heckler Koch - the original point and click interface
I'm running both from the same console (tty). Is there a way to force
asterisk -R into color mode?
It works fine for me to run astersik -rc
Hatton
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
You can send both paties to a meetme conference with Manager Redirect. Or if
you are feeling more adventurous you could load the Manager Bridge patch
that I posted to the bugtracker two months ago. It allows bridging of any
two existing channels together through a manager action:
Hello
This iptables setup won't work
You need specific rules for the incoming UDP packets with status ESTABLISHED
and RELATED like these simple ones
Remember it's a statefull firewall.
In the nat section
-A POSTROUTING -p udp -m udp -m state --state RELATED -j MASQUERADE
-A POSTROUTING -p udp
By user do you mean the caller (initiator of the call) or the recipient? If
you mean that user is the call recipient, it is very easy. The caller's call
comes to you with its Caller ID--if you want the call to go to VM, then
don't answer the call. I use this for forwarding to other PSTN lines
Yair,
One option is like this:
1) User dials ext. 154 to activate call forward (to voicemail)
2) User dials ext. 155 to de-activate call forward
3) Macro to check incoming calls for database entry's
4) The local extention must use that macro (or other way of screening)
1)
exten = 154,1,Answer
Right... that's the one. My mistake.
On 7/1/05, Rob Scott [EMAIL PROTECTED] wrote:
I don't find this option in the Makefile.
I find RADIO_RELAX which is something to do with radios and DTMF.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohit
From what I understand, that is one of the reasons with SIP inband
doesn't mix well with any codec other than G.711.
I believe it's just the ulaw/alaw PCM codecs that allow inband DTMF
for SIP. Anything else will just chew it up.
Mohit.
___
It's a little bit hard to compile but Try oh323 first.
Although, There will be some few situations that H323 will work better
than oh323. So, have both.
Isamar
On Sat, 2 Jul 2005, Adeel -31 wrote:
I m new to asterisk n i've got an IP phone that supports h323 protocol
but i dont know
I'm sending an Asterisk module I've written to see if anybody finds it useful
or wants to provide some feedback. The name of the module is app_ldap and the
application it provides is named LDAPSearch.
LDAPSearch allows any kind of searches on an LDAP directory from the Asterisk
dialplan. It
Thanks guys - appreciate the comments. I understand that IAX does not support inband dtmf, but I still can't fathom why 9 times out of 10 my * box is ignoring DTMF's even though they are showing up in the IAX2 protocol debug output. The really annoying thing is that I can't consistently reproduce
Hello all,
I'm evaluating a VRU project which has huge requirements. I'm looking
for metrics but I haven't found anything that cover my requirements
Initial estimation:
Erlang 61.450
BTH 25.980
T1 req. 88
Digium HW support 4 T1 per card, assuming
Do you think this might have an impact on http://bugs.digium.com/view.php?id=4631?
Mark
On 7/3/05, Mohit Muthanna [EMAIL PROTECTED] wrote:
Right... that's the one. My mistake.On 7/1/05, Rob Scott [EMAIL PROTECTED]
wrote: I don't find this option in the Makefile. I find RADIO_RELAX which is
Does anyone have configs on the MP-108 FXO to asterisk setup? I'm doing
my best with the limited docs, but having very little success.
Thanks,
-Darren
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi List,
This is somewhat off-topic since the problem itself isnt asterisk but the
Ericsson BP250 I want to connect to.
But since there have been a couple of posts relating in part to that system
I am hoping someone can help me out.
What we want to do:
PRI --- BP250 --- Asterisk
Currently
Ok, after hours of research I finally found the problem. I found a
document from digium at
http://www.digium.com/asterisk_handbook/zapata.conf.html
which states that everything above the channel=x statement applies
to that interface which seems a little backwards to me. After
On Saturday 02 July 2005 19:56, Mark Edwards wrote:
Thanks guys - appreciate the comments. I understand that IAX does not
support inband dtmf, but I still can't fathom why 9 times out of 10 my *
box is ignoring DTMF's even though they are showing up in the IAX2 protocol
debug output. The
chawki hammoud wrote:
In the ISDN case, setcallerid or hidecallerid can be
configured and I am aware that Asterisk doesn't
support that on analog line. My question is whethere
there is something like add-on script or hardware that
will do the job. The teleco company provide the
callerid
I have installed festival and it works, but it takes a long time, till
it starts. People hang up before they can listen the info.
Q:
1. How can I shorten this time?
2. If two people call this extension, the second one must wait till the
first one is finished. Is there a way to serve multiple
Robert Goodyear wrote:
On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote:
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I try
to dial an extension (from X-Lite), next time it is done.
X-Lite does not have a tone, nothing
I hear you. background is in definitely in use in my extensions.conf here.
Hopefully this partially accounts for the 10% of times when it _does_ work! ;-)Mark
On 7/3/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Saturday 02 July 2005 19:56, Mark Edwards wrote: Thanks guys - appreciate the
Here is my configuration everything would seems be straight forward, but
I can not register both asterisk with each other.
Both asterisks have Static IP but they are behind firewall/router, so
it means I have to use Register statement.
I'm a bit confused about the register statement.
How can
hello Mike,
we are talking about very different things here. please look at my
original mail again. I want the call recipient to be able to toggle on
and off do not disturb. I don't want the phone to ring at all.
thanks,
yair
On 7/3/05, Mike Hillerbrand [EMAIL PROTECTED] wrote:
By user do
That's all doable. How many residents are you talking about? -- could
take quite a while to call them all. Considering you have outlay in
hardware, phone-cost, utilities (a 100W computer draws $5-$10/month),
consider fixing that well as someone suggested.
-Original Message-
From:
asterisk -nr
n - no colors
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 02, 2005 3:51 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE : [Asterisk-Users] Colored asterisk -R?
Asterisk -gc
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