G'day Darren,
On Fri, 19 Mar 2004, Darren Nickerson wrote:
I give people much more credit than you do, as does the author of that
essay. So do most experienced list-owners out there. Anyone who wants a
post to go to the list will use the 'reply all' feature of their mailer.
They'll
Carlos Chavez wrote:
I have been trying out Asterisk with the speex codec with X-lite as a
client. I applied the REG patch on my windows machine that is recommended in
Voip-info.org. Every time I make a call I get the following error:
codec_speex.c:167 speextolin_framein: Out of buffer
As I started this trend I take the right to end it.
I just want us to follow John Postel's rule for how to act on the Internet
(I think he defined it for TCP/IP software, but it can be applied here too.)
Be strict in what you send
Be generous in what you accept
Sending a reply to
mattf [EMAIL PROTECTED] wrote:
Every time we get close to having old works fall
into the public domain, the large hollywood lobby spreads it's cash
around and buys enough votes to extend copyrights yet again.
The U.S. Senate -- white male millionaires working for YOU!
--
_/ _/ _/_/_/_/
Justin Carlson wrote:
I am sorry if this is a silly question but I can not seem to locate the
festival binaries. does this come with asterisk or is it another project?
No question is silly. This is a good time to remind the list of the FAQ
Thomas Gallaway wrote:
Here is my problem. I have 2 phones (Grandstream Budge Tone-100)
loosing the sip registration
every 4 hours. I can not find out why.
It seems like the registration fails, then a few minutes after
registers sucessfull.
Mar 19 14:06:14 NOTICE[147466]: Registration from
Darren Nickerson [EMAIL PROTECTED] wrote:
I strongly support removing the current reply-to-list setting, and you
should too.
Like many new list admins, I once thought the reply-to was kewel. Requests
to remove it kept coming up, ... usually around the same time someone
embarrassed
Kevin Walsh wrote:
Darren Nickerson [EMAIL PROTECTED] wrote:
I strongly support removing the current reply-to-list setting, and you
should too.
Like many new list admins, I once thought the reply-to was kewel. Requests
to remove it kept coming up, ... usually around the same time someone
Greetings from downunder,
Does anybody know of any organization providing reasonably priced voip call
terminations in Australia and New Zealand ??
Does anybody know of any reasonably priced DID providers in Australia and
New Zealand ??.
Please feel free to contact me off list.
cr
WipeOut wrote:
Carlos Chavez wrote:
I have been trying out Asterisk with the speex codec with X-lite
as a
client. I applied the REG patch on my windows machine that is
recommended in
Voip-info.org. Every time I make a call I get the following error:
codec_speex.c:167
How can I configure * to store the caller and called Party IP Address in the
CDR file.
Thanks for support
Craeck
_
Fotos - MSN Fotos das virtuelle Fotoalbum. Allen Freunden zeigen oder
einfach ausdrucken:
Fritz Müller wrote:
How can I configure * to store the caller and called Party IP Address in
the CDR file.
Depends on the channel, not all channels are IP based.
Check the CDRuserfield - it's a free field in the CDR you set in the
dialplan or from a script.
Without knowing why you want this, I
Does anyone know if qualify=XXX should be used ONLY for user agents
behind NAT.
I tried to use it on a devices (ATA 186 and SPA 2000) on a public IP,
and * goes to segmentation fault every time it starts.
If it is meant to be used just behind NAT fine, but what and how does *
monitor user agent
On Fri, Mar 19, 2004 at 11:23:53PM -0500, Darren Nickerson wrote:
I strongly support removing the current reply-to-list setting, and you
should too.
I would agree with this too, when replying to a post, the reply should
be to the sender, if the receipient wants to reply to everyone, then
they
Hi Craig,
Someone mentioned packet8 to me earlier, having reasonable international
calls from australia, I'd assume they could terminate to it.
On Sat, Mar 20, 2004 at 10:14:44PM +1130, Craig wrote:
Greetings from downunder,
Does anybody know of any organization providing reasonably priced
Hi,
Senad Jordanovic wrote:
Does anyone know if qualify=XXX should be used ONLY for user agents
behind NAT.
No, you can use it if you want to monitor the agent.
I tried to use it on a devices (ATA 186 and SPA 2000) on a public IP,
and * goes to segmentation fault every time it starts.
If it
Fritz Müller wrote:
How can I configure * to store the caller and called Party IP Address
in the CDR file.
Smells like you need to re-think your billing process. There is
absolutely no reason to key on the IP address for billing purposes.
Jeremy McNamara
G'day Craig,
On Sat, 20 Mar 2004, Craig wrote:
Does anybody know of any organization providing reasonably priced voip call
terminations in Australia and New Zealand ??
Does anybody know of any reasonably priced DID providers in Australia and
New Zealand ??.
http://www.oztell.com
I just
Senad Jordanovic wrote:
Does anyone know if qualify=XXX should be used ONLY for user agents
behind NAT.
No, you can use it to qualify any address. Qualification means that
Asterisk regurlarly sends SIP messages with the OPTION method and
the UA answers. We clock the time and if the client takes
Daniel Bichara wrote:
WipeOut wrote:
Carlos Chavez wrote:
I have been trying out Asterisk with the speex codec with X-lite
as a
client. I applied the REG patch on my windows machine that is
recommended in
Voip-info.org. Every time I make a call I get the following error:
I tried to use it on a devices (ATA 186 and SPA 2000) on a public IP,
and * goes to segmentation fault every time it starts.
Does it crash even if you remove Qualify= from sip.conf?
No it does not...
Only when:
Host=dynamic OR host=$PUBLIC IP AND qualify=YES
TO help you we need to get
Hi Craig,
Packet8 doesn't allow asterisk terminations, you have to use their TA
though I haven't looked yet I sure someone must have worked out a way to
fake the info provided by TA.
Costs $50 a month for unlimited calls into the USA, Australia and about
6 asian countries.
BTW if you type source
Thanks a lot I might give it a try. Any specific instructions for running
it with asterisk?
AJ
___
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To UNSUBSCRIBE or update options visit:
EVERY other mailing list I use sets Reply-To to the list address. If the
asterisk lists change, then I'll be increasing the chance of so-called
embarrassing gaffes by not remembering how the list I'm posting to this
minute operates. Besides, the times I've seen such gaffes from mailing
You give too much credit to people, indeed. I cannot say about this list,
but most lists I use have high corporate populations, where the users
*have* to use mailers like Outlook or (cringe) Notes.
Outlook and Outlook express implement Reply, and Reply All, which works well
without needing
The AS5300 will NOT work as a gatekeeper. None of the IOS images support
gatekeeper or IP-IP gateway functionality. It will NOT do IP to IP... it
will do T1/E1 to IP or IP to T1/E1.
The AS5300 will accept traffic from the VIP-400's but will not be able to
forward them except to it's T1/E1 ports(
At 12:41 AM 3/20/2004 -0600, Carlos Chavez wrote:
I have been trying out Asterisk with the speex codec with X-lite as a
client. I applied the REG patch on my windows machine that is recommended in
Voip-info.org. Every time I make a call I get the following error:
codec_speex.c:167
How can I settup a way for Asterisk doesn´t make any use of DIGEST
AUTHENTICATION method?
I don t want ASTERISK to check out any username or password of my users.
Thank you
Joao Carlos Moura
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Try adding 'insecure=yes' in sip.conf.
Regards,
Gus
- Original Message -
From: Joao Carlos Moura [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, March 20, 2004 12:02 PM
Subject: [Asterisk-Users] Basic authentication
How can I settup a way for Asterisk doesn´t make any use of
OK, I think I have an idea as to why it get the private number error.
I do have externip=63.88.139.198 but its not being passed over to FWD.
I think the problem is that I don't use the localnet and localmask
statements. On my network I have an old 38.349.233.0/24 series that we
used to use back
Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out
why the call doesn't go trough ...
sip.conf extract :
[gw001]
type=friend
host=dynamic
defaultip=192.168.0.12
nat=no
dtmfmode=rfc2833
canreinvite=yes
qualify=no
context=tlsgw
extensions.conf extract (from
A post in 2002 refered to Mike Sandman as a source for inexpensive (cheap)
message waiting indicators. I called Mike but he doesn't know what
Asterisk is (!) and wants to know what type of phone system I have or what
protocol it uses so that he can send me a compatible indicator. I tried
[EMAIL PROTECTED] wrote on 03/20/2004 02:58:21 AM:
You give too much credit to people, indeed. I cannot say about this
list,
but most lists I use have high corporate populations, where the users
*have* to use mailers like Outlook or (cringe) Notes. For mailing list
admins to expect
Does someone please
have a sample that shows how to use the directory command in
extensions.conf?
Thanks!
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
Hello,
Did anybody make Adtran TSU600 work with T100P?
I cannot find anything in archives.
I want to buy AdtranTSU600 and T100P but I am
not sure if this is going to work.
Bart
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Go on www.voip-info.org an search for IVR examples ...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Mahler
Sent: Saturday, March 20, 2004 5:35 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Need an example of using the directory command
Does
Hi all,
Does anyone know the procedure for adding a serial output to a cheap caller
display unit. If I can find a way of doing this then I'm sure there will be
away for linux to take the CallerID info, write it to a file, * to open that
file an read the number from it.
TIA
Jon
John Lawrence wrote
Hi all,
Does anyone know the procedure for adding a serial output to a cheap caller
display unit. If I can find a way of doing this then I'm sure there will be
away for linux to take the CallerID info, write it to a file, * to open
that
file an read the number from it.
TIA
Come on, man! Take a look at all of the wonderful resources available
before asking questions. http://www.voip-info.org is your friend.
Start there, and take a few days to read over everything. Then you will
find this: http://www.voip-info.org/wiki-Asterisk+Hardware. The mailing
list is a
Does anyone know the procedure for adding a serial output to a cheap
caller
display unit. If I can find a way of doing this then I'm sure there
will be
away for linux to take the CallerID info, write it to a file, * to
open that
file an read the number from it.
Sorry I never got round to
Everyone--
I filed a bug with ximian against Evolution, in the form of an
enhancement request for integration with Asterisk.
Have a look: http://bugzilla.ximian.com/show_bug.cgi?id=55854
It wouldn't hurt to pile on! Please, add your own comments,
suggestions, disagreements and clarifications.
Useful Asterisk Docs (BOOKMARK THEM!):
http://www.digium.com/index.php?menu=documentation (look at the
Unofficial Links) and http://www.voip-info.org/wiki-Asterisk and
http://www.fnords.org/~eric/asterisk/ (my site) and
http://asteriskdocs.org/
___
On Sat, 2004-03-20 at 13:12, Steve Murphy wrote:
I filed a bug with ximian against Evolution, in the form of an
enhancement request for integration with Asterisk.
Have a look: http://bugzilla.ximian.com/show_bug.cgi?id=55854
Clever. However rather than needing to actually know what the
On Saturday 20 March 2004 18:51, Patrick Lidstone (Personal E-mail) wrote:
In the meantime, there's some good info on hacking CID boxes here:
http://www.automatedhome.co.uk/modules.php?name=Newsfile=printsid=1207
Cheers. That'll do the job.
No to rip apart a few Caller ID units I've got lying
quote who=Steve Murphy
Have a look: http://bugzilla.ximian.com/show_bug.cgi?id=55854
Since you don't want Jane magicaly making John dial Claire, there would need
to be individule login authentication that would only allow Jane to dial and
connect her channel.
So, this is not just Evolution
Joao Carlos Moura wrote:
How can I settup a way for Asterisk doesn´t make any use of DIGEST
AUTHENTICATION method?
I don t want ASTERISK to check out any username or password of my users.
Set no secret in sip.conf our use autocreatepeer
/Olle
___
CW_ASN wrote:
Try adding 'insecure=yes' in sip.conf.
insecure=yes doesn't help in regards to authentication, or?
Please explain more.
/O
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Asterisk-Users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
To
Michael Devenijn wrote:
Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ...
sip.conf extract :
[gw001]
type=friend
host=dynamic
defaultip=192.168.0.12
nat=no
dtmfmode=rfc2833
canreinvite=yes
qualify=no
context=tlsgw
Hey all!!
I know this issue has been address before, but I can not find someone who
has the answer.
I am trying to get my * server to authenticate directly to packet8.
I was very close to them actually giving me the information and possibly
using them for my SIP - PSTN termination, but that fell
I have a network of IAX servers connecting to each other. I just realized that IAX
does some
clever magic by itself. Let me explain:
---
Let's say you have three servers: A, B and Q
A calls B with IAX2
B connects the call to Q with IAX2
B realizes that
Hi
folks,
I need some help
from php/agi experts out there;
I am having
difficulties in extracting the callerid number from php. My script is given
below;
#!/usr/local/bin/php -q
?php//environment dump
ob_implicit_flush(true);set_time_limit(0);
$err=fopen("php://stderr","w");$in =
Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around
with distinctive ringing, trying to make it work. Extensions.conf looks like:
exten = 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer
exten = 3010,2,Dial(SIP/3010,15)
exten =
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk-a-users-list [EMAIL PROTECTED]
Subject: [Asterisk-Users] Use of Alert_Info with C7960?
Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around
with distinctive ringing, trying to make it work.
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk-a-users-list [EMAIL PROTECTED]
Subject: [Asterisk-Users] Use of Alert_Info with C7960?
Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around
with distinctive ringing, trying to make it
And yes, there's a config in iax.conf so you can turn it off if you
for some reason want to bother B with staying in the middle of the
call.
Yap. Great stuff :)
Just so everyone knows the config is: notransfer=yes
It would be good to know what happens with cdr records and call control?
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, March 20, 2004 8:55 PM
Subject: Re: [Asterisk-Users] Use of Alert_Info with C7960?
On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old
Style
and Synth Low. The first three
Hello all,
I'm having a problem with a T1 connection to a Avaya PBX (asterisk is
an IVR).
I could not get pri working and now I'm simply trying to get asterisk
working with fxs_ks.
Questions:
1. Is there anyway to troubleshoot or see what is being sent on the T1.
zttool shows no errors, and
Check out : http://www.voip-info.org/wiki-Asterisk+Avaya
Depending upon the card you are using in the Avaya, you should set it up as
a tie trunk on the Avaya side.
James-
- Original Message -
From: Jeb Campbell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, March 20, 2004
Unfortunately even though it would seem the phone should support the ability
to play custom ring tones, at present it only supports the internal tones
which are:-
Bellcore-BusyVerify
Bellcore-Stutter
Bellcore-MsgWaiting
Bellcore-dr1
Bellcore-dr2
Bellcore-dr3
Bellcore-dr4
Bellcore-dr5
---
Then what's the point of being able to upload custom ring tones?
(as shown in
http://www.loligo.com/asterisk/Cisco/79xx/current/RINGLIST.DAT )
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Christopher Lee
Sent: Saturday, March 20, 2004 8:50 PM
The custom ring tones are selectable through the Ring Type option in the
Settings menu. When the phone rings, it will play that custom ring tone.
Perhaps it's a memory limitation or an issue with the way Cisco are
implementing the SIP firmware as to why you can't select a custom ring tone,
you'd
That's right, you can still select a different ring tone for ALL lines.
Ok.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Christopher Lee
Sent: Saturday, March 20, 2004 9:04 PM
To: Asterisk Users
Subject: RE: [Asterisk-Users] Use of Alert_Info
Greetings All
I'm busy trying out my new snom 200(s)
I have it connected and * CLI tells me registered
1) I pick up the handset and hear the dial tone
2) Dial and Ext, that says Date Time (13)
3) * CLI scrolls that the call is connected and time is being spoken
YET the handset is quite and
From: Olle E. Johansson [EMAIL PROTECTED]
snip
Check the CDRuserfield - it's a free field in the CDR you set in the
dialplan or from a script.
How would you set the CDRuserfield from the dialplan
exten = ?
Thanks in advance
B
___
Barry,
I also just got a new snom 200.
Still discovering features, but as a whole it is working
fine.
Please include the sip.conf entry for the phone you have ..
Also, from your comments I assume that the snom 200 is on
the same LAN as the [*] box?
On the snom web interface, does it show that line
Thanks! I was searching for the wrong thing.
Paul
Paul Mahler
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Devenijn
Sent: Saturday, March 20, 2004 9:30 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Need an
The 7960 will absolutely play custom ringtones.
Paul Mahler
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee
Sent: Saturday, March 20, 2004 5:50 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Use of
From: [EMAIL PROTECTED]
Please include the sip.conf entry for the phone you have ..
SIP Configuration for Asterisk
;
[general]
port = 5060
bindaddr = 192.168.0.15
externip = 24.73.215.62
localnet = 192.168.0.0
localmask = 255.255.255.0
tos = lowdelay
disallow = all
allow =
Yes, I know it will definitely play custom ringtones, I was even using the
24ctu.raw ring tone for a while (I've gone back to Chirp 1 for now). But all
incoming calls get the currently selected ring tone.
I should have clarified on an earlier statement I made:-
Perhaps it's a memory limitation
You need to update the registry and take out the profile you created. This
will clear up the problem. In my opinion just dump firefly and use something
that works. I did.
Message: 3
Date: Sat, 20 Mar 2004 13:07:15 +1100
From: Adam Hart [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re:
Okay, give me a clue how to do it. The phone (sip v6.2) currently has
a ringer called Old Style that plays from the front panel. How do I
code * to play that ringer? (I can't seem to make anything other then
the Belcore-* stuff work.)
Rich
The 7960 will absolutely play
Yes, I followed those directions when I attempted to use pri, but I had
no luck.
(Note, I could not verify that those instructions were followed -- I
have no experience
with Avaya's). If you know the commands to verify those instructions
on the Avaya side,
I would appreciate the tips (Google
Here's another funny
* CLI puts put
-- Registered SIP '4405' at IP.address Port 5060 Expires 3600
and within seconds the snomm 200 beeps the MWI goes on the LCD and the
light flashes a call from asterisk Not Found
Willy if you could let me see you sip and config files, if you have yours
working?
The phone selects the ringtone, not asterisk. I don't know if SIP supports
asterisk selection of a different rington, I'll check it out.
Paul
Paul Mahler
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent:
Here's another funny
* CLI puts put
-- Registered SIP '4405' at IP.address Port 5060 Expires 3600
and within seconds the snomm 200 beeps the MWI goes on the LCD and the
light flashes a call from asterisk Not Found
Willy if you could let me see you sip and config files, if you have yours
Jeb,
What Avaya card are you using? What model of system? Definity, Merlin,
etc? With this I should be able to send you the base commands to review the
card slot settings for the PXB
James-
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Rich Adamson wrote:
The wiki indicates Alert_Info can be set to a number, and implies that
number is the ringer type listed on the phone. Is there a way to select
one of the internal ringer types via Alert_Info?
My understanding is that:
1. 7940/7960 pre version 6 may support numeric values 1-5
Your script is receiving the data correctly, as you will see if you
actually dump that data to a file rather than back to the asterisk console.
The problem is actually in your VERBOSE statement. You are passing back
this string:
VERBOSE Sathya Weerasooriya 1001
Naturally asterisk is confused
sorry, the sip extract is from a previous test now i get the same problem but with
looking for 57228047 in tlsgw and it's the same error, it searching in this direction
:
why are the 2 ast values 0 ??
Non-codec capabilities: us - 1, them - 0, combined - 0
-Original Message-
From:
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