You should use DHCP to enforce IP address to MAC binding when the phones
boot.
And then let the phones register and use host access (deny/permit)
permissions in peer section to restrict by IP address/mask.
alan wrote:
Gavin Hamill [EMAIL PROTECTED] wrote:
On Monday 13 June 2005
The snomSoft-SIP 3.60a is still beta quality and known to have some
audio problems on some systems,
I believe that this will improve soon.
If you have multiple audio devices on the PC you might want to try
different combinations.
Ronald Wiplinger wrote:
I have two snom phones, one is a Snom
That should be working ok.
Check the internal web page of the phone and look at the sip trace to
see if the phone is getting the NOTIFY messages.
dialplan should have:
exten = 301,hint,SIP/360
Henry Devito wrote:
Does anyone have station monitoring working on the Snom 360 softphone?
I have Snom
The hint feature is not a well developed general status monitor, it's
just a *static* method to define the state of an extension as the state
of a device for those channel drivers that support the extension state
API.
It can't be dynamically changed (it's not an executable dialplan
element,
Did you enable call waiting?
call_waiting: on
Josh Dady wrote:
I've got an issue on the snoms, and I'm wondering if anyone has some
recent experience with it; I've contacted the one specific reference I
found to it in the list archives, and the person in question didn't
seem to find an answer
on this, which could have been
spent more constructively
with people that have real needs.
Karl Brose
Bruno Hertz wrote:
On Mon, 2005-01-03 at 09:18 -0500, C F wrote:
As for the term contracted that I have been using, thats because I
HAVE BEEN CONTRACTED, I just don't have a commitment for asterisk
Hi List,
Don't deal with this guy (C F shmaltz at gmail.com), he is misleading
people
and just using the replies to show that what ever he is selling to his
clients is popular.
After a couple of exchanges:
C F wrote:
Thanks for your replay, at this time I'm just waiting for a commitment
from
This was posted in a public forum asking for business relationship.
But the intent was for disguised market research which could have been
stated clearly
upfront and there would have been overwhelming positive response
attesting to
the facts and assisting in his quest.
But communicating with
The RFC specification alone is not sufficient, there are many signaling
packages that are defined elsewhere.
Also, RFC 2705 is out of date, see RFC 3435
Leonardo J. Tramontina wrote:
Sirs,
According to RFC 2705 (MGCP), these are the parameters that are used
in the transactions:
The qop method should only be copied from what you receive from the
server. If the server doesn't
send it, don't send it back.
Your pszURI should be the same as the Request-URI
Compare your code with the routine build_reply_digest in chan_sip.c
Kamran Ahmad wrote:
i found that here Method is
Sounds like you are developing an application.
You should read RFC-3261 and RFC-2617
Kamran Ahmad wrote:
Can any one help me in understanding REGISTER message
when i send REGISTER message to asterisk it is
replying 407
with header
Proxy-Authenticate: Digest
realm=asterisk,nonce=1011592446
i want
This is still a nasty design flaw (bug) in Asterisk.
IAX is similarly bugged.
I can only ask you to wait a little bit longer until I post the solution.
Ian Chilton wrote:
Hi,
I have a few accounts with sipgate.co.uk to get some different DiD
numbers. However, when an incoming call comes in, it
No, you just need a phone that allows you to accept more than one call.
Otherwise, you need to set up your dial plan so that the unanswered call
goes to voice mail
or elsewhere perhaps.
Norman Zhang wrote:
Hi,
Currently * is registered to 1 FWD #. If that line is busy people
can't call in? Do I
Oh, I see. This is the realtime connected problem.
Can't say too much constructive about that without info, I'm not a fan
of it.
We need a debug trace of the registration process (SIP trace and *
messages) to debug why it failed,
not just a one-line message, and anything after that is useless,
The criteria are published in RFC 3435, range is from 1 to 999,999,999.
there is no requirement
of starting from 1. Call agents may allocate certain ranges for certain
groups of gateways.
Asterisk (the call agent) simply increments the id numbers monotonically
for each new request.
Most
What registration failure is that?
The only way to tell is a complete SIP trace of what's going on.
The registration timeout on the phone and in Asterisk should be the same,
unless the server goes down and reboots. The server usually has no way
to tell a phone to
re-register (no real need to do
There is no such thing as subscribecontext parameter in SIP.
I have updated the wiki with the correct current information to make
this work.
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+snom
--khb
Peer Oliver Schmidt wrote:
Hi,
does the hint extension work together with the Snom
The configuration file is 'phone.conf'
mode=dialtone
[EMAIL PROTECTED] wrote:
Hi list,
I have some problems to get the QuickNet Internet PhoneJack working.
What .conf files do I have to edit to get a dialtone for the first
test with the standard configs from asterisk?
I have the ixj driver
Why don't you just set up an extension that calls the system application
to execute a Linux script
Then just make a call to that extension, perhaps use disa to
authenticate and done.
Håkan Persson wrote:
Hi!
I have a Asterisk implementation where I would like to start an update
of the PBX
Interesting question and important as passwords are scattered around
everywhere in Asterisk.
A central user credentials database is needed, so that a user can
connect any which way (SIP,
H323,IAX, MGCP,etc) and use the same set of credentials.
I have a prototype implementation coded for SIP,
You're welcome.
It's been submitted.
Cyrille Demaret wrote:
Hi,
Thank you, it's working now!
Do you think that this patch will be included in the next cvs versions?
Sincerely,
Cyrille Demaret
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Is the SIPquest server sending the 401 Unauthorized message verbatim as
you printed it here?
I.e. is the WWW-Authentcate header broken up into several lines like that?
If so, how man spaces are actually at the beginning of each new line?
Continuation lines are allowed in SIP, but I think it's
Just checked the RFC, and it does say that a tab is acceptable.
SIP header field values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
white space, including folding, has the same semantics as SP. A
recipient MAY replace any
Try turning on pedantic mode in sip.conf
pedantic=yes
My X-tens only encode for the # (pound) character, not the '*'
Stanley Cline wrote:
Has anyone else had issues with Asterisk rejecting calls from X-Lite
softphones when the dialed number contains the * or # keys (e.g., dial #86 on
X-Lite
)
on responses to outgoing requests.
Stanley Cline wrote:
On Wed, 10 Nov 2004 15:53:12 -0500, I wrote
On Wed, 10 Nov 2004 14:36:04 -0500, Karl Brose wrote...
Try turning on pedantic mode in sip.conf
pedantic=yes
That fixed it! :)
It turns out that pedantic=yes fixed the X-Lite
You will also need to change a file in Asterisk core, to make it right,
I think it was file.c.
Adam Fineberg wrote:
Original Message
Subject: [Asterisk-Users] Voicemail questions
Date: Tue, 9 Nov 2004 15:16:40 -0500
From: Chris Armour [EMAIL PROTECTED]
Reply-To: Asterisk
The syntax for the register command is
register=username:secret:[EMAIL PROTECTED]:port/extension
Benjamin on Asterisk Mailing Lists wrote:
On Sat, 6 Nov 2004 06:33:15 -0800 (PST), Girish Gopinath
[EMAIL PROTECTED] wrote:
AFAIK, the 050 in the From header acts as a
Try creating a symbolic link for the INBOX directory
(varspoolpath/asterisk/voicemail/context/user/INBOX) of one user
to the INBOX directory of the second user. VM announcment files stay
separate, INBOX is shared.
Leah Newmark wrote:
Is there a way to set up a voicemailbox for multiple
Benjamin on Asterisk Mailing Lists wrote:
IAX is so vastly superior to SIP, that the comparison shouldn't be
things like VHS versus Betamax, but it should be more like horse
carriages versus motorcars.
And what to you base such an assertion on?
Would you care to elaborate on the technical
NONSENSE
Benjamin on Asterisk Mailing Lists wrote:
On Thu, 28 Oct 2004 14:45:46 -0600, Ryan Courtnage [EMAIL PROTECTED] wrote:
Yep, you can do this, just requires some port forwarding and special
considerations in sip.conf.
You are missing the point. There is no *solution* to SIP NAT
are
dangerous together?
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl Brose
Sent: October 29, 2004 5:49 PM
To: Benjamin on Asterisk Mailing Lists; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Suggestion
You can only restrict the range of ports used, in rtp.conf.
I suppose restricting it to 2 ports starting on even number might do it,
but if you're not using SIP on one end, how are you going to start a call?
You need to have at least rudimentary call control for SIP invite and SDP
exchange, and
The problem you are having is due to the way chan_phone was designed.
The distributed driver does not buffer the entire phone number dialed
and then send it on to the PBX,
like a SIP phone would, but instead scans the dial plan after every
digit is entered to look for a match.
The solution is to
Please see my rely for the related topic
Eric Jacksch wrote:
Perhaps this will help...
I have a phone connected to a QuickNet PhoneJack card. When I pick it up, I
get a dial tone. When I dial a certain number of digits, the call is
processed by Asterisk.
The question: How does Asterisk
No Brian,
The old driver scans the ENTIRE dial plan on EVERY digit dialed so no
matter where, if you have a
. wildcard in the plan, it will match always on the first digit dialed.
It is the driver that does this.
If you use a SIP phone, or any technology that presents a complete dial
string,
-
[EMAIL PROTECTED] On Behalf Of Karl Brose
Sent: Sunday, September 05, 2004 4:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Wildcards and variable number of digits
No Brian,
The old driver scans the ENTIRE dial plan on EVERY digit dialed so no
matter
It's not clear how you are making the call.
You should be able to call directly from either phone to the other by
dialing 5011 or 5012, respectively, if
your context local indeed contains the those extensions, which is not
clear from your configuration excerpts.
But it seems you are calling
Craig Guy wrote:
cancallforward=yes
There is no such function in distributed chan_sip.c,
ergo there can't be such a configuration parameter.
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To
STUN (RFC-3489) is an UNSAF type network protocol (see RFC 3424) that is
used to discover UDP address and port
bindings across network address translators.
(a) Currently Asterisk only supports static configuration of the
external IP address of a NAT.
You need to discover it manually by other
[EMAIL PROTECTED] wrote:
Hi Karl,
I'm suffering with the problem you outlined in (a) regardless of a STUN Server
being used.
Is their anyway around this?
It's not a fault of the STUN server.
Yes, with a little patience there will be a way around this. We are
close to releasing STUN support
This message does not indicate major trouble.
It simply means that the STUN protocol is not using message integrity
generation/checking
which is a checksum method using HMAC hashes.
STUN works ok without it.
muralikrishnan lakshmanan wrote:
Hi friends
I have some doubt in connecting my
The times are measured as a result of qualify=yes and involve sending an
OPTION SIP request to the peer which in turn has to reply
to the request. So this is an application layer ping which naturally
has much greater latency than an ICMP ping.
Brent Franks wrote:
Hello,
I am cruious what
It's correct that neither the SRV lookup is handled correctly or
completely, nore is there in standard distro a way to register with the
proxy for a domain, if those names differ.
It wasn't a difficult task to change this.
If there is interest I might release the patch for this as part of
multiple account registrations by duplicating the line.
Kevin Walsh wrote:
Karl Brose [EMAIL PROTECTED] wrote:
There is also the option of expanding, or better redesigning, the [peer]
sections with proper and logical configuration options
and adding a register=yes flag.
I would prefer
Use Voicemail instead of VoicemailMain
Reto Stauss wrote:
When a user dials 999 he is always asked for the mailbox and has to enter his mailbox
number and password. As I understand this shouldn't happen because the CALLERIDNUM is
passed over to VoicemailMain. It's annoying to have to enter the
Yes, it is something to worry about, because you might run out of RTP
ports or open fd's, depending on your port range in rtp.conf.
Which cvs version are you running?
This behavior was observed by several people for a short period of time
and then seemed to have disappeared with a cvs versions
The relevant configuration file is phone.conf and the channel name is
Phone/phone0
i.e.
exten = 999,1,Dial(Phone/phone0)
Kevin Chew wrote:
Hi all,
I am trying to configure asterisk to work with quicknet phonejack PCI
card. I tried to serach the internet for the relevant .conf files but
no
Hi again,
You can't dial out with a PhoneJack. It's an FXS device only.
For dialing out with a Quicknet product you need the LineJack card.
Kevin Chew wrote:
Hi all,
I am still confused about the way to use asterisk with QuickNet
Phonejack. If I am not wrong, The phonejack card should be using
I compile Asterisk with gcc3.4 occasionally to check for compatibility
(under RH9)
Aside from some compiler warings it compiles just fine.
Your issues seem to be more related to running gcc 3.4 under the newer
kernels
rather than compiling asterisk with newer version of gcc.
Karl Rannseier
Yes, it's a known issue on the bug tracker (#1110), but no solution has
been found to date, afaik.
Julien Levi wrote:
Hello,
I was planning to use the output of asterisk -rx show queues in a
script when I noticed that sometimes asterisk only outputs the first
line of the response. e.g:
Luis,
I tried to simulate your situation using a sip agent (Xten X-Pro) and
having it register to Asterisk with two user ids simultaneously all on
the same LAN.
I cannot replicate your problem. Both id's registered immediately.
Can you test this in your environment replacing the gateway with
: [Asterisk-Users] Sip Registration Problem
Karl Brose wrote:
Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or
not, Asterisk doesn't do it correctly either.
The host should respond with 200/OK if the call could succeed
theoretically if it were an INVITE or else it should send a
404
*/
if (!strcasecmp(cmd, OPTIONS)) {
+ check_user(p, req, cmd, e, 0, sin, 0);
res = get_destination(p, req);
build_contact(p);
/* XXX Should we authenticate OPTIONS? XXX */
Olle E. Johansson wrote:
Karl Brose wrote:
Btw, Ignoring OPTIONS
I posted them for you yesterday.
[EMAIL PROTECTED] wrote:
Dear All,
Any one know the correct SIP setting for the TerraCall?
Thank You.
Cary LEUNG
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To
It's a bug in Asterisk.
I believe it's still open also on the bugtracker. There are a few
reported senarios with these kind of problems.
Some of them where solved with the recent 'ast_gethostbyname' fix. Are
you running a recent version?
Btw, Ignoring OPTIONS is not a valid option (:-) whether
Try using the IP address below directly and not a hostname.
The follow works for me.
[terracall]
type=friend
host=64.69.76.33
username=##x##
secret=
fromuser=##x##
fromdomain=pc.tt.xten.net
nat=yes
context=terracall-inbound
[EMAIL PROTECTED] wrote:
Dear All,
I had try the new cvs
The asterisk qualify option in sip or iax
sends a test packet to the remote host every minute
and measures the time it takes for a response to
come back. If this time frame is less than what is
configured on the qualify statement or 2000ms if
it is 'yes' than the host is flagged unreachable until
Hi there,
The use of the realm setting in the SIP [general] section is simply to
set the protection domain of the Asterisk server as a registration
server. According to the RFC the string must be globally unique, so we
are finally just following proper protocol. It is not used to as a
Hmm, your call trace doesn't seem to reflect the dial plan you show
us. There is more to this somewhere else.
Probably some misconfiguration?
Steven Kokinos wrote:
Hello-
I am currently testing with a carrier that seems to be having some trouble
around toll-free (800 number) access. While a
I think when you have this setup you need to keep the media path going
through Asterisk at all times.
Your SIP is binding to both ports, internal and external, but that
doesn't correctly set it up for either scenario, localnet calls and
external calls. It won't keep the addresses straight for
It's called caller id.
Paul Mahler wrote:
I need to dial an extension that tells me what extension I'm dialing from.
I'm running a bunch of analog phones off a channel bank to * over a T1. I
have the following in extensions.conf.
exten = 98,1,SayDigits(${EXTEN})
This says the digits the
Sounds like you are using CVS 1.0stable?
Proxy Authentication is broken in that CVS head, and it may not get fixed.
Using development head will fix this.
see also the bugtracker.
is it possible to use PROXY AUTH with */sip ?
Szenario:
UAC = Asterisk ( SIP REGISTRAR/PROXY) = SER ( PROXY ) =
Wasn't this just fixed?
markus monka wrote:
R=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o say.o say.c
say.c: In function `ast_say_number_full_de':
say.c:769: parse error before `int'
say.c:770: `thousands' undeclared (first use in this function)
say.c:770:
Bob,
What I am going to tell you may seem arrogant or what, but I think
you would do yourself a great favor if you figured this one out yourself
by studying the info that is available and ask questions if things don't
work. Your configuration is indeed very simple and with 100% certainty
you will
was posted on a day or two ago
Thorsten Gehrig wrote:
hi
anybody running with german SIPGATE?
my configuration don't works :-(
regards
[EMAIL PROTECTED]
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[EMAIL PROTECTED]
I know it's exciting to get things working, however,
there are some things wrong with your configuration, despite it perhaps
working ok.
Is it really? You can make outbound calls this way?
In your friends definition (friend-sipgate) you don't have a host
specified.
host=sipgate.de
Without that
You may be quite right, I have read parts of the rfc at least, I remember,
but the lure of using cheap existing infrastructure is probably to great.
KHB
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, May 04, 2004 03:20
Subject: RE:
to all cards. Make sure the device=
lines are at the end of the file.
I have a rewritten driver for chan_phone, that corrects all these problems
and lets
you configure each device separately. You are welcome to try it.
Send me a private e-mail if you're interested.
Karl Brose
- Original Message
I can connect to FWD on my asterisk - but FWD only see me as an external
SIP agent and not a SIP client of the FWD network. DOn't know exactly why
- so would luv to compare your conf files.
There is really not much you can do about that with your local
configuration.
The problem is on
Asterisk will accept unauthenticated calls, defaulting to the context
specified in the general section.
Therefore only the call to extension 88 should work.
If both, 77 and 88, are working for you then, yes, something is broken.
- Original Message -
From: Paul Mahler [EMAIL
It doesn't matter what ports you run at.
But the only way to make it work with different ports right now is to change
the driver source
and recompile. That's what I meant by enabling the code.
Regarding the qualify, Asterisk monitors the connection to the host by
sending probes or pings
and
I don't quite understand the problem, so I will
only respond to the PORT=4569 part
Currently the driver ignores the setting and gives
you the message. It will use the default
port of 4569 however. I don't see a
reasonwhy it was disabled,but I am running my IAX channelsat
other
ports just
ldconfig
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, April 18, 2004 19:24
Subject: [Asterisk-Users] libspandsp.so.0
I successfully compiled installed the
spandsp-0.0.1k.tar.gz modules for faxing and
patched the asterisk according to
Here is a problem I am having with IAX registration.
Starting up Asterisk I am registering with IAXTEL with no problem.
But then most of the time when I issue a reload, Asterisk all of a sudden
tries to register to an IP address on my network, although that host is not
even up and not
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