Roy Sigurd Karlsbakk wrote:
Asterisk...
Linux...
You get what you pay for. And it's free
Part of the price is the work you have to do yourself or pay a
contractor to do. Open Source is not shrink-wrap.
Trying to grasp the problem report, it seems like a lot of problems derives from
the system as
Chris Albertson wrote:
--- John Todd [EMAIL PROTECTED] wrote:
WipeOut wrote:
One for the gurus..
Obviously not for me, but I'll dare to give it a shot anyway ;-)
Anyway, I decided to go and have a quick read through the SER docs
and in the section about NAT they say that the best way to
To try to summarize and turn this discussion into advice for newcomers, I've
wrote up a rollout-advice page in the Wiki, inspired by all your messages in this
thread. Thank you for participating, even if you weren't aware of it...
http://www.voip-info.org/tiki-index.php?page=Asterisk+rollout+tips
John Todd have started creating a document called Readme.channels that will document
the
syntax of extensions in all channels. I have uploaded his draft to the Wiki, so that
all
of you can help find the syntax, it's not so easy to grasp from reading the source. It
would really be handy to have
For those of you running * on FreeBSD:
I compiled everything and can start. Sockstat -l shows that Asterisk listens on the
correct interfaces and ports. Sniffing, I see registrations coming in to SIP debug,
but nothing seems so reach Asterisk except IAX registration from a peer. Can't dial
Tilghman Lesher wrote:
On Sunday 19 October 2003 11:45, Olle E. Johansson wrote:
For those of you running * on FreeBSD:
I compiled everything and can start. Sockstat -l shows that Asterisk
listens on the correct interfaces and ports. Sniffing, I see
registrations coming in to SIP debug
Eric Wieling wrote:
On Mon, 2003-10-20 at 11:31, Chris Albertson wrote:
Asterisk works perfectly fine in back of a NAT firewall, as long
as all of your SIP phones are also in back of that same fire
wall ;-)
Seriously, I'd fix this if I knew enough about SIP protocol.
Is anyone willing to write
Steve Creel wrote:
You'll want to #include it. This leaves the burden of the [general] and
any static configs on sip.conf but allows the script to blindly write out
from the database to sip_additional.conf
in sip.conf:
#include sip_additional.conf
Eureka! ...is this #include construct a general
Jeremy McNamara wrote:
Panny Malialis wrote:
Is app_meetme broken?
I seem to get invalid conference number all the time :(
You have to have some Zaptel device installed. Like wcfxo or ztdummy
See
http://www.voip-info.org/tiki-index.php?page=Asterisk+timer
for more information!
/O
Help, I'm stuck. Lost in the woods.
I have one Asterisk running on FreeBSD outside on the Wild Internet.
One on the safe inside, behind a NAT firewall.
The inside server registers with IAX to the outer one and can place calls.
The outside one can't register to the one on the inside, since it
Louis-David Mitterrand wrote:
On Mon, Oct 20, 2003 at 03:55:47PM -0500, Tom Walsh wrote:
Also trunking requires that some sort of timing device (digium card or
ztdummy) be in place for trunking. Otherwise trunking is disabled.
What does ztdummy require to work? kernel compile options? Does it
WipeOut wrote:
Olle E. Johansson wrote:
Help, I'm stuck. Lost in the woods.
I have one Asterisk running on FreeBSD outside on the Wild Internet.
One on the safe inside, behind a NAT firewall.
The inside server registers with IAX to the outer one and can place
calls.
The outside one can't
WipeOut wrote:
http://www.voip-info.org/wiki-Asterisk+timer
This will not work on SMP systems (Multiprocessor), where the RTC clock
is used for SMP support.
Symetrical Multi Processing
Fixed. Thank you!
And maybe SMB file sharing needs timers too ;-)
/O
Johnson, Randy wrote:
-Original Message-
From: WipeOut [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 23, 2003 2:12 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX peers and NAT
Olle E. Johansson wrote:
Help, I'm stuck. Lost in the woods.
I have
I've tried to list various files and applications in Asterisk that includes passwords.
http://www.voip-info.org/tiki-index.php?page=Asterisk+password+files
If you know any other file or application with passwords, add to the Wikipage or mail
me
offlist so I can update.
Sometime in the future,
WipeOut wrote:
Olle E. Johansson wrote:
Here is basically the way mine is setup.. names changed to protect the
innocent.. :)
Maybe you can spot what you are missing..
PBX1- insidepbx (behind NAT)
---iax.conf--
register = user:[EMAIL PROTECTED] ; Server on static IP
[outsidepbx
WipeOut wrote:
First off, can AGI scripts be created using PHP??.. This is where our
skills are and since PHP can be run from a command line it would be
easier to create and maintain..
Oh, flame war warning. There's been a lot of discussion on this before,
mostly about the necessity to use PHP
James Sizemore wrote:
To be a true ip tel softswitch, Asterisk would need SS7 support.
No one is working on SS7 signaling for Asterisk.
I can't say anything about the progress, but OpenSS7 has as a work
item to connect Asterisk and SS7.
http://www.openss7.org
/O
Jonathan Hogg wrote:
OK. I've tried trawling the archives, but I'm not getting very far. I've got
an Asterisk box behind a NAT which I want to register with a SIP provider.
If you've travelled around the archives, you should now that this is a FAQ.
At this moment, Asterisk behind a NAT can't
Are there corresponding AGI commands for the Asterisk commands?? eg a
command to dial instead or using the Asterisk Dial command..
Not one-to-one correspondence. Some of the basic ones you need are there.
ANSWER
AUTOHANGUP time
CHANNEL STATUS [channelname]
EXEC application options
GET DATA
One more question: What are agents, and what are they good for? Help and
Wiki don't reveal much... I am starting to think we'd really need to get
an overview of the * features and have that documented (without all the
details, just to get the big picture which makes a start a lot (!)
easier).
Ken Godee wrote:
I'm a little new around here but..
From what I've been working on...
Setting up agents in the agents.conf file
allows you to then assign agents in your
call queues as a members.
[cut]
Thank you!
http://www.voip-info.org/tiki-index.php?page=Asterisk+Agents
A very, very
Philipp von Klitzing wrote:
Hiya!
I hear you. I've started a brief introduction on
http://www.voip-info.org/tiki-index.php?page=Asterisk+introduction
Do you think I'm totally off the road or on the way to what you're looking for?
Wow - Olle (and others here), I have been around for a week or
I've currently got Asterisk running behind NAT with iconnecthere and it
works with incoming and outgoing calls. All I did enable nat in sip.conf
(nat=1) and authenticate against natrelay.deltathree.com. The only
'special' thing I can see about my setup is that the NAT device supports
UPNP. I
Jan Janak wrote:
I experimented a little bit and Asterisk behind NAT with SIP works. I created
an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.
Great! I copied your information for other users to the Wiki.
Alexandru Coseru wrote:
Hello..
There is another way of doing SIP auth other then manually add the user
passwords to sip.conf ?
There are scripts that generate the text files and a patch that adds
the functionality to hide passwords and replace them with MD5 digests,
but as far as I know,
My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD server. On a slower
CPU
linux system, Asterisk runs at 0.1% - both without any active channels...
Any ideas, anyone recognizing the problem?
/O
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Rich Adamson wrote:
I experimented a little bit and Asterisk behind NAT with SIP works. I created
an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.
Great! I copied your information for other users to the Wiki.
Rich Adamson wrote:
My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD server.
On a slower CPU linux system, Asterisk runs at 0.1% - both without any
active channels...
Any ideas, anyone recognizing the problem?
Is 'top' suggesting that * is actually consuming 98%?
Yes, on
Rich Adamson wrote:
My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD server.
On a slower CPU linux system, Asterisk runs at 0.1% - both without any
active channels...
Any ideas, anyone recognizing the problem?
Is 'top' suggesting that * is actually consuming 98%?
Yes,
Jan Janak wrote:
I experimented a little bit and Asterisk behind NAT with SIP works. I created
an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.
I can confirm that Asterisk behind NAT can call out to IPtel.org
...and users connected to iptel.org can call me, if
Philipp von Klitzing wrote:
You will probably have to use canreinvite=no in the UA definitions in
the SIP.conf for those two phones..
In your case you want the opposite: canreinvite=yes
A try to sort out these kind of opposite messages:
When asterisk connects two SIP phones, it tries to be in
Perry E. Metzger wrote:
Olle E. Johansson [EMAIL PROTECTED] writes:
My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD
server. On a slower CPU linux system, Asterisk runs at 0.1% - both
without any active channels...
Any ideas, anyone recognizing the problem?
On the BSDs, your
and slided into development, such things happens.
/Olle
Good luck
--- Olle E. Johansson [EMAIL PROTECTED] wrote:
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk on FreeBSD
Date: Mon, 27 Oct 2003 21:25:04 +0100
Perry E. Metzger wrote
Chris Albertson wrote:
Aha. It may be connected to this error message, then:
messages:Oct 26 21:18:15 NOTICE[137382912]: File sched.c, Line 209
(sched_settime): Request to schedule in the past?!?!
I read somewhere that I could ignore this message, therefore I just
didn't include it in my earlier
Béasse Christophe wrote:
Hi,
In voicemail. i declare :
1000 = 1234, ,[EMAIL PROTECTED]
I don't see what's password 1234 is for ?
password for what ?
Where this password is used ?
Where this password is defined ?
This is a pin code used by the VoiceMailMain application to let
Peter Zeltins wrote:
Checking e-mail this morning it looks like we have two independent
fixes that both do what has been suggested in this thread.
No need for a third except posibly a merge of the two.
Would you care to elaborate? I don't see anything in asterisk-users, and no
mention of
Béasse Christophe wrote:
I have some troubles with voicemail sending message to my email address.
Do I have to configure sendmail to use Voicemail ?
I found that the use of sendmail was hardcoded into voicemail. Therefore,
I've submitted a patch to voicemail2 that let's you configure mail
http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ
Welcome to add short questions and answers!
/O
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http://lists.digium.com/mailman/listinfo/asterisk-users
David J Carter wrote:
I have a FWD number and wish to connect it to Asterisk to receive my FWD
calls.
See the Asterisk FAQ at
http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ
You'll find pointers to several Asterisk - FWD configurations there.
/Olle
WipeOut wrote:
David Sussman wrote:
Apologies if there is a cleanly written and searchable FAQ that I
could be
directed to. I have no problem to RTFM if I can find the FM...
Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R
servers that do not support X windows under
Shoval Tom wrote:
Either it's not working, or I don't know what I'm doing. It's giving me the
error sox: effect '.gsm' is no known!
Let's say I need to convert file 1.wav to 1.gsm.
How do I apply this command to it?
FAQ. See
http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ
(I've just
Shoval Tom wrote:
Olle, I can't reach the faq page, and haven't been able to for the last four
days.
I'm getting 504 gateway timeout errors.
Gateway timeout indicates something with your web proxy ...or?
I've been able to reach the Wiki all weekend, I've updated and created
several pages...
I
CW_ASN wrote:
Here is my example. I'm using a lot of times a day.
?php
$socket = fsockopen(192.168.0.53,5038, $errno, $errstr, $timeout);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: admin\r\n);
fputs($socket, Secret: blabla\r\n\r\n);
fputs($socket, Action: Command\r\n);
WipeOut wrote:
Shoval Tom wrote:
And how will all us newbies make the linux box as secure as possible?
The quickest way is to setup an IPTABLES firewall.. You will need ports
5060 and 1 to 2 open for a default Asterisk install using SIP
only..
Visit the Wiki page
Peer Oliver schmidt wrote:
I have not been able to reach www.voip-info.org as well using my default
settings.
Upon researching the problem I tried
nslookup www.voip-info.org
which returns an IP address of
192.168.168.3
which is obviously wrong. This is the answer from my local DNS server
Andrew Kohlsmith wrote:
tested the 911 capability of * and, using an extension trick given from the
#asterisk IRC channel, dialling 911 just plays You will dial 911 in 5
seconds. If this was done in error, hang up now before actually zapping a
trunk line (if all are busy) and dialling out.
Martin Pycko wrote:
You can port forward the 5060 SIP port and use externip keyword in
sip.conf to have it working behind a NAT.
Martin,
Could you please explain the use of the new externip keyword.
Is it a [general] keyword or something configurable for SIP host/peers/friends?
Thank you!
/Olle
Rich and I have updated the Wiki page Asterisk rollout tips with advice on how to
plan
and implement your Asterisk rollout. This page is based on many discussions on the
mailing list, so don't be surprised if your comment or thought is included in the
text. Thank you for your input!
Martin Pycko wrote:
It's new. It prevents asterisk from putting the private IP in the messages
that asterisk sends with SIP.
Hmmm. According to the sip.conf example:
[general]
externip = 200.201.202.203 :Address that we're going to put in SIP messages if we're
behind a NAT
Does this apply
Christoph Loibl wrote:
however while running asterisk without any clients online my system-load is
about 1 - and asterisks seems to run all the time. is this the default
behavior?
Check the list archives at http://lists.digium.com
I reported the same problem a while ago and got no solution, but
Philipp von Klitzing wrote:
Olle, www.voip-info.org still resolve to 192.168.168.3 from here, and
many other places (like our branch office, my home dial-up account, my
parents dial-up account)
It's been fine for me (Gin ermany, both via university (GWin) and T-
Online), no problems at all, so
Stephen R. Besch wrote:
Gavin Hamill wrote:
Hullo again, all :)
If you're using * to run telephony in a real business environment, can I
trouble you to write a short paragraph about the setup, and how you've
found the migration / daily use?
Well, it's not technically a business, but we are
Thank you, Steven!
http://www.voip-info.org/tiki-index.php?page=Asterisk+setup+success+1
ADIT 600
What is that?
Zhone Zplex 10b
And that?
/Olle
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...and to solve another problem, there's my suggestion on support for outbound SIP
proxy.
http://bugs.digium.com/bug_view_page.php?bug_id=359
There are corporate networks that use a SIP proxy proxy as an ALG, application layer
gateway,
for all outbound and inbound SIP traffic in the DMZ.
Philipp von Klitzing wrote:
http://www.skype.com/
seesm to be the latest craze... anyone have any knowledge of their
technolgy use etc ??
- closed source
- WinXP and 2k only
- peer-2-peer, i.e. they route foreign calls through your client (and
bandwidth) if that helps the calling parties
In
Alastair Maw wrote:
On 05/11/03 10:14, Olle E. Johansson wrote:
As I understand it they must not be fully peer-to-peer even if they
use your bandwidth, there has to be media servers in their network,
handling calls. Or?
No, the whole point is that it's completely decentralized. More
Steven Critchfield wrote:
On Wed, 2003-11-05 at 04:55, Roy Sigurd Karlsbakk wrote:
Is there any information available about ZapRAS other than the fscking
source?
just pointing out for those interested in reading that I have written up
my recent experience over on the -dev list where I read
Michael Manousos wrote:
A newer version of asterisk-oh323 is available. This version
features a set of channel variables and improvements in audio frame
handling. People that have reported clicks or choppy sound, in
some cases, should try this version.
Download from:
Gavin Hamill wrote:
It would seem an odd question, but I'm trying to put together a little
presentation on 'Why Asterisk?' and need to list Pros and Cons I've
got plenty of Pros (including the availability of commercial support),
but the only Con I can think of is 'Relatively few
Ryan Tucker wrote:
Since it's all the craze, I might as well post our current Asterisk
usage. :-)
Thank you!
http://www.voip-info.org/tiki-index.php?page=Asterisk+setup+success+3
/Olle
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First, please do not use friends. It will confuse you. We receive calls from
type=user
and send calls to type=peer (that is the general idea, anyway... :-)
For an incoming call, we first match on users based on the username part of the
From:
sip address. So if the call comes from sip:[EMAIL
Brian West wrote:
Ok to cut confusion here
Its:
Variable: _ALERT_INFO
In CVS head.
In stable, it's still ALERT_INFO.
The same applies to VMXL_URL that is now _VXML_URL in CVS head.
This is due to a change in app_dial where you now are able to
set any variable in the new call leg created by dial()
Karl Brose wrote:
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,
http://edvina.net/broadvoice/
We now have an update to the patch and new configuration instructions.
Apply the patch to a an unpatched copy of chan_sip.c, recompile and
reconfigure.
Note that if you followed the earlier instructions on adding a host
entry to /etc/hosts, you will have to remove
Reid A. Forrest wrote:
I don't know why but my * is not accepting Register Messages.
Have you seen this kind of problem before??
I need help!
Thank in advance.
*CLI Nov 19 15:42:11 NOTICE[12893]: chan_sip.c:4869
register_verify: Peer
Time to reboot and re-start Asterisk, well, hrrm, monthly, news.
It's been a hectic fall with a lot to do, both before and
after Astricon.
At this time, we're preparing for two Astricon shows in 2005. And no,
we haven't made a decision on where to run the European Astricon,
not yet.
I am preparing
Andy Reinke wrote:
SIP SECURITY WARNING
[general]
contex=sip-unauthorized
If you spell this right, all calls from unknown SIP devices will be sent to the
context you set here. If you do not set a context in the general section of
sip.conf, default will be used.
This is the way you configure how to
Public Dump wrote:
For reasons unknown to me, SER and subsequently a Microsoft Live
Communcations Server 2005 seems to have problems, matching a SIP ACK
request from asterisk to the ongoing SIP transaction, I have attached
the complete log, but the essential lines are:
That's a bug in
Welcome to the Asterisk users community!
Asterisk.org is a fast moving project. New code is added every day.
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Our community is also growing
Doug Reid - Stormcorp wrote:
HI
I got the same problem that only started lately. I have to do a
stop start to get the phones registered again. One site out of 12
with the same spec.
Running CVS head or a stable release?
Please show us SIP debug on first registration and then failed
Mick Hastings wrote:
Hi Folks,
cheers for all the great info on the list.
I need to create a SIP NOTIFY message to reboot my Cisco 7960 phones but I
dont know how.
The admin guide gives an example of the packet (attached), I have tried a
few web searches and found some cool
little programs that
Alexander Lopez wrote:
OK with CVS-HEAD-12/22/04-09:47:32 incoming (to ata) sip calls on a
ATA0186 now seam to work fine. However transfers still do not work.
With CVS-HEAD-12/22/04-12:46:47 transfers still do not work.
You need to download the fix from the bugtracker until Markster
approve and
*** SIP Channel fixed in CVS stable
---
During a few days there's been a buggy SIP channel in CVS STABLE, but
not in the 1.0.3 release tarballs on the FTP server and mirrors. We have
now removed the patch that was integrated by mistake so CVS should be
ok again.
Karl Brose wrote:
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
The subscribecontext is part of chan_sip2, not in standard Asterisk
For incoming calls, Asterisk matches peer's on IP, meaning that the
first peer it finds will match. This is the *last* one you have in
sip.conf. The context given in that peer must have *all* extensions you
need for incoming calls, which is the extension at the end of the
register= line in the
Friends!
I have recently discovered that chan_sip, chan_sip2 and chan_sipx all
lack support of SIP multicast. This has a major impact on my network,
since I haven't got the bandwidth needed to call all of you and send you
this message. With that feature missing, I have to go back to old
John Baker wrote:
tad wrote:
hi folks.
this may be common knowledge, but i haven't seen it documented. so...
it seems that one cannot pass multiple arguments to agi scripts? for
example, a line like this in extensions.conf:
http://bugs.digium.com/bug_view_page.php?bug_id=664
Please test
Welcome to the Asterisk users community!
It's great to have you participating in this Open Source project
- building an Open Source PBX. Here are a few things to know and
remember while working with the project.
** The mailing list is growing
The lead
://lists.digium.com/mailman/listinfo/asterisk-users
--
Olle E. Johansson, Edvina.net AB, [EMAIL PROTECTED]
- Phone +46 8 594 788 10, Cell phone: +46 70 593 68 51
- IP phone: sip:[EMAIL PROTECTED]
- Address: Runbovägen 10, SE-192 48 Sollentuna, Sweden
- Web: http://edvina.net
There is a lot of MGCP-related reports, bug fixes, patches in the bug tracker that
needs help to be tested, fixed, evaluated, thrown away or added to the CVS.
So if you're using MGCP, please visit the bug tracker, apply an MGCP filter and help
us moving those bug reports forward or closing them.
Iain Stevenson wrote:
--On Wednesday, March 24, 2004 11:13 am -0600 Steven Sokol
[EMAIL PROTECTED] wrote:
I have seen a number of postings cross this list that mention the
possibility of standards-tracking IAX2 with the IETF (generating an RFC,
etc.). Has that gone anywhere? What would it
Adam Hart wrote:
Olle E. Johansson wrote:
An informational RFC documenting the protocol would be a good start,
it would
make it more open but not an IETF product. Security specialists would
get something
to read and analyze. A VOIP protocol with RSA authentication,
implemented today
Hal A. Lightwood wrote:
I've successfully installed Asterisk and have Microsoft's Instant
Messenger connecting. We can make VoIP calls between clients without a
problem, however we cannot send text instant messages between clients.
From what I can tell this should be possible using IETF SIMPLE
It looks like we have to create an Internet Draft which is assigned to the
relevant working group for revision, questions, comments, more revision,
then it may or may not become an RFC. Unfortunately, the IPTel working
group appears to be made up of people who are heavily invested in SIP.
That's
Hal A. Lightwood wrote:
Thanks for the quick, if not very detailed answer. Obviously I am
interested in this capability, is there some reason we couldn't work on
this? I believe SER might support it (it seems to work between FWD
clients at least), why not asterisk? What would be required to
Lal, Deepak (Contractor) wrote:
I am trying to use Asterisk as a pure voicemail system and have the following
setup:
I have the * setup as a SIP peer to a softswitch. When someone calls a number on
the softswitch and no one picks up the phone, the softswitch forwards the call
to the * using SIP.
Mailling LIst wrote:
Hi guys,
I am a newbie and having problem to enter a conference room. Here is an
extract of my config files:
I had a look on the mailing list archive but did not find anything
regarding this problem. Thanks in advance for your help
This is really a FAQ. You need a
Jakob Strebel wrote:
Yash,
Just yesterday i joined asterisk mailing list. We want to use it in
our company. Can someone tell me which version or type of Linux would
best work for it and also what should be the configuration of machine
(hardware configuration) to install Asterisk?
I did it
Welcome to the Asterisk users community!
It's great to have you participating in this Open Source project
- building an Open Source PBX. Here are a few things to know and
remember while working with the project.
** The mailing list is growing
The lead
Brian Cuthie wrote:
Let's say that I have a call coming in to Asterisk through a TDM400P and
going out through SIP to someone on the Internet. Is there any
configuration option that would allow me to do silence suppression on
the RTP stream generated by Asterisk on behalf of the TDM400P
Larry Keyes wrote:
Hi...I've got two Grandstream phones attached to my Asterisk on the same
subnet. The phones have fixed IP addresses. Asterisk is generated an error
for one of them only, even though both appear to be registered correctly.
The current state of the sip.conf is included below.
Richard Airlie wrote:
At this stage, and someone please confirm that Asterisk is
really working on FreeBSD ? :)
Yes, it's working with some limitations.
See http://www.voip-info.org/tiki-index.php?page=Asterisk%20freebsd
/O
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Richard Airlie wrote:
On Mon, Apr 05, 2004 at 02:33:08PM +0200, Olle E. Johansson wrote:
Richard Airlie wrote:
At this stage, and someone please confirm that Asterisk is
really working on FreeBSD ? :)
Yes, it's working with some limitations.
See http://www.voip-info.org/tiki-index.php?page
Personally, I think VAD is a great service, as well as comfort noise
generation to disguise when VAD is working. I'll always encourage
methods that reduce bandwidth. Most major developers on Asterisk
consider these technologies of low concern since their bandwidth is
unlimited, as they
Steven Sokol wrote:
Having just returned from four days at the VON show in Santa Clara, I
thought I would submit a highlights message. I hope others who attended
the show will take the opportunity to add, as there was far more to see than
I can cover on my own.
Thank you for a good report!
Scott Laird wrote:
On Apr 5, 2004, at 12:10 PM, Olle E. Johansson wrote:
Members of the IETF added information on the to-be-standardized standard,
meaning that SIP with TLS over TCP will be mandatory. We need to start
working
on TCP and TLS support.
Could someone explain to me why anyone
James Golovich wrote:
On Mon, 5 Apr 2004, Scott Laird wrote:
Could someone explain to me why anyone in their right mind would ever
want to run VoIP (or any lossy real-time data) over TCP? Unless I'm
missing something, the effects of packet loss would be almost perfectly
pessimal. Every time
Steven Sokol wrote:
I think Olle Johansson took pictures of the event.
They may already be on the WiKi in fact.
I've uploaded the pictures without editing at
http://www.voip-forum.com/asterisk/von2004/index.htm
Enjoy!
/O
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Scott Laird wrote:
On Apr 5, 2004, at 12:49 PM, Olle E. Johansson wrote:
SRTP protects RTP/UDP media with encryption.
There are concerns that sending positioning within SIP/UDP will reveal
private detailes, like position. Hence the encryption requirement.
The position data needs to be given
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