Hello,
Is it possible to use a database for provisionning sip clients?
CVS provides sip-friends.sql in order to create tables (not database)
what may i do with that tables?
Regards
Harry
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Dear Group,
At the moment I use SJPhone as my soft phone with Asterisk.
I prefer the look and feel of the x-ten lite. However, when
ever I use my x-ten lite I get a lot of breakup in my communication.
E.g. I will play some hold music, and every 5-6 seconds I
drop some packets. I
The problem could be to do with the silence surpression feature in X-ten
Lite. If you go into Advanced System Settings, Audio Settings, Silence
Settings, you should have Transmit Silence set to Yes as Asterisk has a
compatibility issue with the way it tries to surpress transmitting the
Hi,
I have a Snom 200 that has had admin mode switched off and I have no
idea when the admin password has been set to.. Does anyone know of a way
to reset the phone to factory defaults??
Later..
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* Brethren,
It's a sad day in our community. Please join me in a moment of silence for
the death of responsible journalism. Silence.good
enough.
This article goes on to tell about Pingtel's announcement of forming the
first open source community aimed at creating SIP based
Hi.
Did you buy the phone or get it second hand ? If second hand do you have
any paperwork from the person you bought it from and did they buy it
through official distribution?
If you got it through distribution I would am fairly sure your vendor
might be able to help ?
I have a rough idea of
I'm having a sound issue. I'm using BT100 (102). When I dial the echo
test ( or anything for that matter) outside of my LAN there's no sound
when it answers although I hear the ringing tones. Is this an RTP or
codec issue. When I dial through Zap everything is fine. Thanx.
Sounds more like a firewall issue. TL
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Vlok Stone
Sent: Saturday, April 17, 2004 6:03 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] no sound when connected
I'm having a sound issue. I'm using BT100
Hi!
I am having difficultly in having users of various SIP devices obtain the
correct behaviour when they call a busy number ie. only hearing the
Congestion/Busy tone. I assume this might be because the SIP device
itself generates the 'ring' tone?
With my current setup in the dialplan extract
Hi Vlok,
When a call connects is the audio one way ? Can the remote person hear
you but you can't hear them ?
Which way is the audio or is it silent in both directions ?
The echo test? Is this FWDs echo test or the one running on your
asterisk box (as that is not outside you LAN is it) ?
I'm
Same here.. aliases file looks botched. Not just the article
potentially broken perhaps ?
Chris
- The following addresses had permanent fatal errors -
[EMAIL PROTECTED]
(reason: 550 5.1.1 [EMAIL PROTECTED]... User unknown)
- Transcript of session follows -
... while
On Sat, 2004-04-17 at 14:01, Chris Orme wrote:
Hi Vlok,
When a call connects is the audio one way ? Can the remote person hear
you but you can't hear them ?
yes.
Which way is the audio or is it silent in both directions ?
The echo test? Is this FWDs echo test or the one running on your
There is a way.
Right after reboot, and when you see the first text, hit any key
and you will get a 'boot menu'. Give the phone an ip-address and define
a tftp-server.
The bootfile must be named snom200.bin ( e.g rename the latest snom sw ).
After you have succesfully got it to download the
Very much sounds like a firewall issue not allowing voice packets back in
to you (for the received audio) or them not finding you somehow.
Think about how do you connect to the internet. Perhaps 'it' (whatver
device it is doing firewalling/NAT) is configurable through its bios or a
web
On Sat, 2004-04-17 at 14:43, Chris Orme wrote:
Very much sounds like a firewall issue not allowing voice packets back in
to you (for the received audio) or them not finding you somehow.
Think about how do you connect to the internet. Perhaps 'it' (whatver
device it is doing
this is my BT100 phone dtmf. Is this correct.Send DTMF:
in-audio via RTP (RFC2833) xvia SIP INFO
i chose sip info
On Sat, 2004-04-17 at 14:43, Chris Orme wrote:
Very much sounds like a firewall issue not allowing voice packets back in
to you (for the received audio) or them not
My dialplan is for the outgoing SIP call is:
exten = _00.,1,AbsoluteTimeout(3600)
exten = _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
exten = _00.,3,Answer
exten = _00.,4,Hangup
exten = _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r)
exten = _00.,104,Answer
exten = _00.,105,Hangup
I can't help with
I hear some echo there :)
simply, you can define sip friends from a database.
just create the table, enable SIP_FRIENDS into channels
Makefile and read chan_sip.c how to set
db access (db access data must be into sip.conf)
but, firstofall, you must be familiar with sip.conf
and
Chris Orme wrote:
Hi.
Did you buy the phone or get it second hand ? If second hand do you have
any paperwork from the person you bought it from and did they buy it
through official distribution?
If you got it through distribution I would am fairly sure your vendor
might be able to help ?
I
I assume you've got masquerading working so other hosts inside this
network are ok ?
something like...
/usr/local/sbin/iptables -v -t nat -A POSTROUTING -i eth0 -o eth1 -j
MASQUERADE ?
Definitely sounds more network than asterisk.
How about trying to connect to IAXTEL (which uses IAX2 rather
Pertti Pikkarainen wrote:
There is a way.
Right after reboot, and when you see the first text, hit any key
and you will get a 'boot menu'. Give the phone an ip-address and
define a tftp-server.
The bootfile must be named snom200.bin ( e.g rename the latest snom sw ).
After you have
Hi Linus,
Thanks for pointing that out.
Luckily Asterisk has a 'billed seconds' field in the cdr which is 0 when a
number is unavailable or busy despite showing the call as 'answered'.
A view could be taken that 0 length billed seconds calls need not be
billed with a minimum connection
Wipeout,
I bought the phone new about a year ago so its not provider locked..
I set the password to be nothing (I think) and then I set admin mode
off, then when I tried to get into the admin area I couldn't, it would
seem that either there is a bug that doesn't allow a blank password or
The tftp suggestion you received is well worth trying :-) I didn't know
that was possible.
Although I wonder if it works as it might mean that carriers deploying the
snom may not be able to properly lock their phones perhaps??
Chris
On Sat, 17 Apr 2004, WipeOut wrote:
Chris Orme wrote:
The [EMAIL PROTECTED] appears to be broken. I dug around the magazine
contacts and found Doug Allen, senior editor, you can send comments to
[EMAIL PROTECTED] . I didn't get a bounce back from that e-mail so I assume
it made it to the editor.
JR
If there is one I'd love to know about it feel free to let me know as I
use snoms a lot.
The only other ways I know that might be possible are more complicated
than the tftp method.
Good luck!
On Sat, 17 Apr 2004, WipeOut wrote:
Pertti Pikkarainen wrote:
There is a way.
Right after
exten = _00.,1,AbsoluteTimeout(3600)
exten = _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
exten = _00.,3,Answer
exten = _00.,4,Hangup
I can't help with presenting busy to the SIP devices, but if you have
the
above on any sort of PSTN gateway you are going to annoy the PSTN
users - as
Duane wrote:
Tom Green wrote:
Brian,
Encrypted SIP messages can be sent using TLS. However,
I don't think it is realistic to expect everyone
calling you to have a public/private key pair.
I don't quite agree.
SMTP servers that support SMTP-TLS and have valid certs + config do
exactly that
Sorry for echo I just wait for a reply :)
I looked at Voip-info but does a GUI is provided to insert datas in
tables ???
Le sam 17/04/2004 à 17:36, Brancaleoni Matteo a écrit :
I hear some echo there :)
simply, you can define sip friends from a database.
just create the table, enable
Nicholas Bachmann wrote:
1. It's a chain of trust: it's hard for Bob to verify Alice's signature
directly
-Not impossible to fix
CAcert.org's whole purpose is cheap, easily obtainable security... It
employs a web of trust in the website frame work to build up and
distribute face
I *think* the default password is (all zeros).
HTH,
Ryan
On Apr 17, 2004, at 10:38 AM, WipeOut wrote:
Pertti Pikkarainen wrote:
There is a way.
Right after reboot, and when you see the first text, hit any key
and you will get a 'boot menu'. Give the phone an ip-address and
define a
Pertti Pikkarainen wrote:
There is a way.
Right after reboot, and when you see the first text, hit any key
and you will get a 'boot menu'. Give the phone an ip-address and
define a tftp-server.
The bootfile must be named snom200.bin ( e.g rename the latest snom sw ).
After you have
Chris Orme wrote:
exten = _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
Isn't the 'r' forcing a 'ringing' signal from start, regardless
of what the device you are calling are signalling. If you are calling
a SIP device, that device might return 'busy' and that's propably
why you first hear 'ringing' and
Duane wrote:
Nicholas Bachmann wrote:
1. It's a chain of trust: it's hard for Bob to verify Alice's
signature directly
-Not impossible to fix
CAcert.org's whole purpose is cheap, easily obtainable security... It
employs a web of trust in the website frame work to build up and
Thanks Olle !
I'm getting on better without the ,r now when making outgoing SIP calls
though not confident I've got rid of any ringing with all devices on test
yet.
I put the ,r in irrationally when the iaxy wasn't ringing out not that I
think it necessarily helped anyway - taking it out
Linus, I assuming that for incoming service something like .
exten = incomingnumber,1,AbsoluteTimeout(3600)
exten = incomingnumber,2,Dial(SIP/sipdevice,120) (maybe with ,r)
exten = incomingnumber,3,Congestion
exten = incomingnumber,103,Busy
[where incomingnumber is whatever the
Hi,
I have a query from a client that can he use the E100P card to terminate
the 2Mbps bandwidth in a linux box, thus reducing the cost of cisco
router ??
The other end is a cisco 2620 router with E1 VWIC-1MFC.
Can anyone explain if its possible with Asterisk and further any
configuration
I now have a virtual number in addition to my main number from
broadvoice. Enabling distinctive ring in their portal results in the
following SIP header being received:
Alert-Info: http://127.0.0.1/Bellcore-dr3
It looks like Asterisk has the ability to generate an outgoing
Alert-Info header
I fired off a note to them at both addresses as well, and asked them check
out astricon as well..links included. (clues to the clueless mode...)
I read their article on Pingtel going OS while sitting on the porcelain
furniture the other morning, and thought they might be about to discover
fire
Here's a bit of a twist to the common UK Caller ID question... (Which I've got
working nicely thanks to some slight changes in Jonathan McHarg's scripts off
the asterisk-dev mailing list, and a Pace modem from ebay!)
Can a standard BT phone that supports CID (Such as a BT Decor 310) pick up the
Can a standard BT phone that supports CID (Such as a BT Decor 310) pick up
the
CID information that asterisk passes out to analog lines or would I have
to
get an analog phone with CID from the states?
Most of the BT brand caller display phones support both the BT/UK standard
means of
Nicholas Bachmann wrote:
A web of trust is different from the chain of trust I'm talking about.
In a web of trust, a key is signed by lots of different people; ideally,
everybody can trust everybody. In a chain of trust, each member only
knows and trusts the adjacent members.
CAcert doesn't
We did not see so far that a provider would pay the phone so it's only fair
that the user has the ability for example to change the provider. The user
owns the phone!
Btw take a look at http://www.snom.com/faq_en.php,
http://www.snom.com/faq/FAQ-04-03-24-sf.pdf.
CS
-Original Message-
Kudos to the CAPI developers.
I would like to have multiple MSNs on my ISDN Bri
lines.
I see all the cool features but cannot find any examples or
guides to build from.
Currently running Diva Eicon Cards with CAPI from http://www.junghanns.net
I would like to route calls to sip
On Sun, Apr 18, 2004 at 09:31:48AM +1000, Duane spake thusly:
be sure more are issued on a correct basis. PGP model if you lived in
say Africa and wanted to communicate with someone in South America with
little or no prior relationship and you wanted to be sure the
communication wouldn't be
Tracy R Reed wrote:
I prefer the PGP model because it includes the CA model. That is to say
that you can still have a CA within the PGP model. Both myself and my
colleague from Africa could pay a central CA we both trust (Verisign,
Thawte, whoever) to sign our keys and connect us in the web of
On Sun, Apr 18, 2004 at 10:22:08AM +1000, Duane spake thusly:
Just a little matter of key distribution, how do you know the CA key
given to you is actually the CA? Especially since Thawte no longer does
PGP key signing and verisign is making too much money from PKI...
Same way I know
Tracy R Reed wrote:
Same way I know someones key is theirs by the pgp fingerprint. It's well
publicized and they use it quite a bit.
But have you ever met face to face with an employee from a CA and
verified they were an employee or just grabbed the info from their
website and assumed there was
I fired off a note to them at both addresses as well, and asked them check
out astricon as well..links included. (clues to the clueless mode...)
I read their article on Pingtel going OS while sitting on the porcelain
furniture the other morning, and thought they might be about to discover
fire
On Sun, Apr 18, 2004 at 11:13:27AM +1000, Duane spake thusly:
But have you ever met face to face with an employee from a CA and
verified they were an employee or just grabbed the info from their
website and assumed there was no man in the middle attack sending you an
alternate
Tracy R Reed wrote:
No, I haven't. And you are right it is highly unlikely. Knowing that
someone was going to want to get a key signed, putting the bogus info
where they would find it, tricking someone into calling you and giving
them a bogus key, etc. is all very difficult. I think we are going
Title: Message
Try
upgrading to SIP 6.3. I heard from someone on the IRC Channel about this
problem and 6.3 resolved it
-gcc
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig
WaddingtonPosted At: Friday, April 16, 2004 1:04 PMPosted
To:
Here is my sip.conf and extensions.conf which allow me inbound and outbound
calling between * and Freeworld Dialup, with * behind a NAT.
;
; SIP Configuration for Asterisk
;
[general]
disallow=all
allow=ulaw
port=5060 ; Port to bind to
bindaddr=0.0.0.0;
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