The bug tracker includes several issues relating to Path (RFC 3327)
support. It is not clear which version actually included the patch and
which versions are working.
Could anybody update these issues in Jira with a brief comment about the
supported versions?
Reminder: speaker's deadline this Friday, 27 November at 23:59 UTC
We have already received several really exciting talk proposals
but there is still time for people to propose talks or encourage
friends or colleagues to speak.
Many other dev-rooms also have a deadline in the next few days and
Contact
===
For discussion and queries, please join the free-rtc mailing list:
https://lists.fsfe.org/mailman/listinfo/free-rtc
The dev-room administration team:
Daniel Pocock <dan...@pocock.pro>
Ralph Meijer <ral...@ik.nu>
Asterisk is mentioned in quite a few places in the RTC Quick Start Guide[1]
I've put up a blog today about my work on this book and some
questions[2] for discussion.
I'd be particularly interested in any feedback from the Asterisk
community about just how Asterisk fits into the federated SIP
I'll be in Norfolk, VA for xTupleCon in October
On 15 October, there will be two events for WebRTC:
14:15 a talk about the xTuple WebRTC extension at xTupleCon
- must register for xTupleCon to attend this
17:30 a technical / developer workshop at xTuple's offices
- free, anybody
I've seen the following appear in some tests with Asterisk 11.11:
WARNING[3938][C-0003]: chan_sip.c:10535 process_sdp: Rejecting
secure audio stream without encryption details: audio 54908
UDP/TLS/RTP/SAVPF 109 0 8 101
Specifically, it always happens from a Firefox 24 host but it works
I'm using v11.11
I tried setting:
force_avp=yes
in a SIP peer in sip.conf and it seems to be ignored.
The WebRTC client sends an INVITE with RTP/SAVPF and Asterisk is still
sending back 183 and 200 responses with the UDP/TLS/RTP/SAVPF string
Are there some limitations with this option or
On 21/07/14 15:12, Daniel Pocock wrote:
On 21/07/14 14:33, Joshua Colp wrote:
Daniel Pocock wrote:
I've now replicated my setup on a host with a single IPv4 address and I
am still having trouble with the ICE negotiation.
I am trying to call from Jitsi to Asterisk through a Prosody XMPP
On 22/07/14 18:20, Joshua Colp wrote:
Daniel Pocock wrote:
snip
FYI, I'm using the Debian packages, latest is 11.10.2~dfsg-1~bpo70+1
Has the chan_motif / xmpp / ICE stuff changed significantly in 12.x
releases?
Nope.
Is there any way I can enable ICE debugging?
Not within 11
I've now replicated my setup on a host with a single IPv4 address and I
am still having trouble with the ICE negotiation.
I am trying to call from Jitsi to Asterisk through a Prosody XMPP
server. Asterisk successfully registers with the XMPP server and
appears to be available in the buddy list
On 21/07/14 14:33, Joshua Colp wrote:
Daniel Pocock wrote:
I've now replicated my setup on a host with a single IPv4 address and I
am still having trouble with the ICE negotiation.
I am trying to call from Jitsi to Asterisk through a Prosody XMPP
server. Asterisk successfully registers
I have a multi-homed machine (quite a few IP addresses on one of the
interfaces)
For SIP I found that using externaddr in sip.conf would make it much
more reliable with ICE and RTP using the correct IP
Is there an equivalent setting for XMPP / motif.conf? I saw bindaddr in
gtalk.conf but it
If I understand correctly, setting
encryption=no
means that Asterisk will make outgoing calls without encryption, but
will be happy to accept incoming calls regardless of whether the caller
wants encryption or not
If encryption=yes, then Asterisk not only uses encryption for the
outgoing calls
The sample config files in the Asterisk distribution and packages are
really good for getting the demo up and running quickly, for example, to
extend the demo to run behind a WebRTC proxy only required about 6 lines
of extra code to define a peer in sip.conf and enable TCP
However, I'm not sure
Kamailio has both a ha1 and ha1b column in it's user schema:
ha1 = H(A1) = MD5(user:realm:password)
ha1b = H(A1b) = MD5(user@realm:realm:password)
This is intended to support some devices that append @realm to the user
and/or to allow users to put either user-part only or user@domain
into the
Is the template capability in sip.conf compatible with realtime sip.conf
entries such as users in a database?
I notice that contrib/realtime/mysql/sippeers.sql and the wiki page
don't mention a template column:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
On 06/06/13 15:51, Daniel Pocock wrote:
Is the template capability in sip.conf compatible with realtime sip.conf
entries such as users in a database?
I notice that contrib/realtime/mysql/sippeers.sql and the wiki page
don't mention a template column:
https://wiki.asterisk.org/wiki/display
On 03/06/13 23:04, Daniel Pocock wrote:
On 03/06/13 19:18, Jason Parker wrote:
On 06/03/2013 12:03 PM, Daniel Pocock wrote:
I tried building manually from the source RPM
Before running rpmbuild, I installed libsrtp-devel and I notice that
res_srtp.so is generated during the build
However
As mentioned in the thread about MP3, I found that the rpmbuild process
demands network access, e.g. to access the mp3 code in SVN.
Some people need to build on isolated networks though
I've attached a patch that allows the MP3 code to be placed in /tmp
before the build starts, then svn will
Given the recent announcement about Google slimming their support for
public interconnection with XMPP, can anybody comment on where this
leaves the XMPP support in Asterisk?
In particular, I notice many of the references to XMPP on the wiki link to
I've now prepared a blog about my experience setting up Asterisk 11 with
repro as a SIP proxy for WebSocket clients:
http://danielpocock.com/using-resiprocate-to-connect-asterisk-webrtc
In particular, the focus is on the use of packages because that makes it
faster for more people to deploy
On 04/06/13 18:37, Tzafrir Cohen wrote:
On Tue, Jun 04, 2013 at 12:49:35PM +0200, Daniel Pocock wrote:
As mentioned in the thread about MP3, I found that the rpmbuild process
demands network access, e.g. to access the mp3 code in SVN.
Some people need to build on isolated networks though
On 04/06/13 19:13, Tzafrir Cohen wrote:
On Tue, Jun 04, 2013 at 06:44:43PM +0200, Daniel Pocock wrote:
On 04/06/13 18:37, Tzafrir Cohen wrote:
On Tue, Jun 04, 2013 at 12:49:35PM +0200, Daniel Pocock wrote:
As mentioned in the thread about MP3, I found that the rpmbuild process
demands
On 07/08/12 23:11, Rusty Newton wrote:
On 8/7/2012 7:27 AM, Paul Belanger wrote:
On 12-08-07 03:31 AM, ml asterisk wrote:
Hi,
I used to install asterisk on debian squeeze with digium repository.
The last build of asterisk available is 1.8.11.1.
Is this repository discontinued ?
Since
I tried installing the Asterisk 11 RHEL6 packages from packages.asterisk.org
I followed this guide:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
The SRTP support appears to be missing though. I notice libsrtp was not
automatically installed as a dependency, and no srtp module
Building from the source RPM I get an error
mISDNuser-devel is needed
I was able to obtain all the other build dependencies from EPEL 6, but
that one doesn't appear to existing in EPEL or in packages.asterisk.org
I then tried adding --nodeps to the rpmbuild command:
rpmbuild
On 03/06/13 18:46, Jason Parker wrote:
The packages currently do not support SRTP.
I tried building manually from the source RPM
Before running rpmbuild, I installed libsrtp-devel and I notice that
res_srtp.so is generated during the build
However, the rpmbuild fails for other reasons (see
On 03/06/13 19:18, Jason Parker wrote:
On 06/03/2013 12:03 PM, Daniel Pocock wrote:
I tried building manually from the source RPM
Before running rpmbuild, I installed libsrtp-devel and I notice that
res_srtp.so is generated during the build
However, the rpmbuild fails for other reasons
I've just done a test with a WebRTC client connecting to the repro proxy
with the SIP messages relayed over TCP to Asterisk
Asterisk successfully answers the call using SAVPF, SRTP and ICE.
The client is greeted by the demo
This was tested in the Asterisk 11 environment described in my
On 31/03/13 23:43, Joshua Colp wrote:
Daniel Pocock wrote:
I'm trying to call from DruCall to Asterisk and I get this error:
WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F
103 104 111 0 8 107 106 105 13 126'
== Problem setting up ssl connection:
error:
On 01/04/13 22:06, Joshua Colp wrote:
Daniel Pocock wrote:
Thanks for the fast reply. I agree backporting full support for AVPF
would not be justified for many use cases (including my own). What I
was more curious about is whether the F can be tolerated (in other
words, ignored
On 17/12/12 13:34, Joshua Colp wrote:
Barco You wrote:
Dear All,
Hola,
I use sipml5 to register two users from browser and the two clients
are successfully connected. But when I made a call from one of the
users, the other user doen'st have call notification and for a while the
I've set up a peer to use G.722 only and tried to make it talk to an
Asterisk box
Asterisk always rejects the call with the following error:
[Jan 14 22:20:16] WARNING[32653]: chan_gtalk.c:1343 gtalk_newcall:
Capabilities don't match : us - 0x4 (ulaw), peer - 0x1000 (g722),
combined - 0x0
On 14/01/13 23:31, Joshua Colp wrote:
Daniel Pocock wrote:
I've set up a peer to use G.722 only and tried to make it talk to an
Asterisk box
Asterisk always rejects the call with the following error:
chan_gtalk was written to only support a limited number of codecs, not
the full set
For those using Debian/Ubuntu (and anybody else is welcome of course),
there is a mini-DebConf in Paris this weekend:
http://fr2012.mini.debconf.org/
There is a presentation at 16:00 about Debian's role in establishing an
alternative to Skype, this will look at some of the packages
On 20/08/12 16:23, Administrator TOOTAI wrote:
Hi,
I have to connect 3 asterisk servers,each of them being TLS server for
his clients and connected in both way in TLS with both others asterisk,
each having hi own Common Name. Is this possible?
I set up 2 asterik's , one server and the
On 20/08/12 21:11, Danny Nicholas wrote:
This is all nice and good but the documentation all assumes that you are on
a Debian box and use MYSQL. What about us SUSE/Postgresql folks?
They are both good questions, and there are partial answers:
SUSE:
reSIProcate can be built from source on a
On 20/08/12 22:53, Danny Nicholas wrote:
I'm fond of the tar-config-make method that Asterisk uses. Is this possible
for reSIPprocate? If so can you provide a link?
http://www.resiprocate.org/ReSIProcate_1.8_Release
You can access the download directory (use the 1.8.5 tarball) or SVN
Given the limitations around Asterisk's TLS support, and all the
benefits of using a SIP proxy, I've put together a rough guide about how
to use the repro SIP proxy as a front-end for Asterisk connectivity with
TLS peers:
http://www.opentelecoms.org/using-repro-with-asterisk-or-freeswitch
It
On 11/08/12 01:26, Paul Belanger wrote:
Is Digium officially endorsing 1.8.13 for wheezy in any way?
No. Digium nor the Asterisk Project has anything to do with the package
within Debian. In fact, most of the work is done by Tzafrir.
I'm not referring to the actual packaging processes, but
Debian 7 is currently in the `freeze' status with 1.8.13 - that means
Debian 7 is very likely to release 1.8.13 and be carrying it for the
next 2-3 years (typical lifetime of a Debian release)
I run 1.8.8. TLS has a bug: it fails to receive BYE over the TLS
connection from my Polycom phone.
I
Debian is very conservative about accepting updates during the `freeze'
process - they will most likely want to see a 1.8.13.2 release with ONLY
the most essential fixes
a) is anyone else aware of these bugs?
b) what essential changes should go into 1.8.13.2 for Debian?
We don't need to
On 06/08/12 13:48, Daniel-Constantin Mierla wrote:
* http://asipto.com/u/68
The tutorial focuses on how to use Asterisk's database structure to
perform authentication in Kamailio SIP server, along with user location,
nat traversal, instant messaging, presence, a.s.o., offloading
On 06/08/12 02:59, Vladimir Mikhelson wrote:
Have you tried 1.8.15?
I'm trying 1.8.13 because that is the versions currently scheduled for
release in Debian 7 (wheezy)
http://packages.debian.org/wheezy/asterisk
If 1.8.15 contains definite solutions for TLS problems, then either
a) they can
Package: asterisk
Version: 1:1.8.13.0~dfsg-1+b1
Severity: important
On 05/03/12 10:47, Wolfgang Pichler wrote:
Hi all,
i have had sip TLS with an own signed certificate (using the
ast_tls_cert script) running on asterisk-1.8.8 - i then have updated
to 1.8.9.3 - and now i get the message
I've recently released a dlz ENUM module for the bind9 nameserver:
http://www.opentelecoms.org/dlz-ldap-enum
Basically, it handles ENUM queries from Asterisk, FreeSWITCH, repro,
Kamailio, Lumicall, searches for the phone number in ENUM, and if found,
returns the email address as both a SIP
On 07/02/12 05:29, Gordon Messmer wrote:
On 02/06/2012 03:27 PM, Josh wrote:
Why do you see binding to 0.0.0.0 to be a security risk?
Purely because a response from Asterisk can be received as a result of a
connection on *any* interface on the system/machine. If I have Asterisk
confined to,
* And, is it necessary to use both my server specific certificate and
the intermediate certificate on the telephones or will the telephones
only require the server specific certificate?
The phones should already have the root certificate for Geotrust, you
should not deploy intermediate roots
On 01/02/12 10:58, Stuart Elvish wrote:
Thanks for the clarification. I have looked at Polycom's website and
saw which phones have the latest firmware (or at least a firmware that
supports TLS) available.
Didn't get around to the testing with the chained certificate but will
try again
On 31/01/12 16:16, Gilles wrote:
Hello
To cut down on the number of hackers trying to break into an Asterisk
server, I'd like to simply move the SIP port from the standard UDP
5060 to something non-standard.
Something more appropriate for your goal might be a move to TLS, it is
definitely
I've raised a bug report about this here:
https://issues.asterisk.org/jira/browse/ASTERISK-19268
I'm just wondering who else has been investigating RFC 5922 style
certificate practices?
Which CAs have been able to provide appropriate certificates?
What kind of interoperability testing has
I've just come across this issue:
https://issues.asterisk.org/jira/browse/ASTERISK-17727
I am strongly in support of TLS and I believe this issue will be a
stumbling block for more and more users - because more and more CAs are
using the intermediate certificate chains
For example, the free
On 30/01/12 17:12, Stuart Elvish wrote:
Hi all,
Firstly, apologies if the answer to this question should be obvious.
I have just started working with SRTP and had a self-signed
certificate working perfectly. I have now purchased a CA signed
certificate but can't get it to work properly
- upgrade policy - is it intended that someone who has Debian 6 with
the existing Asterisk 1.6 packages (from Debian's maintainer) can just
upgrade to the Digium package without moving or changing any config?
There is nothing specific about the packages that is going to make this
Hi all,
All the examples I've come across seem to suggest configuring
jabber.conf/jingle.conf/gtalk.conf for a real Google account.
What about the scenario where the Asterisk server should connect to an
account on a private Jabber server and using Jingle (voice calling over
Jabber)?
e.g. for
This effort is not intended to replace packaging of Asterisk in the
official Debian or Ubuntu repositories. Our repositories are for
providing access to major versions of Asterisk that are newer than what
is included. We are exploring ways to work as closely as possible with
the Debian and
For those of you coming to FOSDEM on 24/25 Feb, there'll be a session in
the Debian devroom on Open Source VoIP.
http://www.fosdem.org/2007/schedule/speakers/daniel+pocock
Several VoIP projects will be represented in various ways throughout the
weekend, and there will be some
Jason Lee wrote:
Hi,
I was testing the intel based G729 codec on SVN-trunk-r42453 following
the
new instructions for compiling with SVN trunk and it in preliminary
tests it
works ok for some calls but I found when one end of the call is an IVR or
Music On Hold the sound gets all distorted
the backtrace from a segfault?
On 9/9/06, Daniel Pocock [EMAIL PROTECTED] wrote:
Jason Lee wrote:
Hi,
I was testing the intel based G729 codec on SVN-trunk-r42453 following
the
new instructions for compiling with SVN trunk and it in preliminary
tests it
works ok for some calls but I found when
zap show status
will tell you if Asterisk is really using ztdummy
Make sure you have chan_zap.so enabled in modules.conf (or that it isn't
disabled with a noload declaration)
Nigel Godfrey wrote:
On a new set up Centos 4.4, kernel 2.6.9-42.0.2.EL, yum updated, 2
BRI-HFC cards, no digium
Check sip.conf parameters:
rtptimeout
rtpholdtimeout
David Gagnon wrote:
I would recommend you to call Unlimitel as they have a very good support. Or
just send a copy of your post to : [EMAIL PROTECTED]
David
_
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
Brandon Galbraith wrote:
Steve,
Forgive my ignorance, but why does India institute that policy?
Why does France blow up bombs in the south pacific? Each country can do
as it pleases - unfortunately - but that is also good for us VoIP
carriers because it creates and protects high retail
The Intel IPP based open source release of G.729 and G.723.1 have now
been updated to compile with the following versions of Asterisk:
- Asterisk 1.2.11
- Asterisk trunk - tested with SVN r 42264
The code is at the usual location:
http://www.readytechnology.co.uk/open/ipp-codecs/
If you
A new release of the open source G.729 patch has been issued.
The new URL is:
http://www.readytechnology.co.uk/open/ipp-codecs
The memory leak in codec_g729 is now fixed. This was due to a
problem in a section of code copied from the Intel example. Thanks
to those who assisted in
Hi,
I've tried to request VoIP PSTN numbers from a couple of Australian
companies who are advertising on Google, but neither of them was able to
fulfil despite advertising the numbers on their sites. In fact, I was
disappointed that both of them actually asked me to complete their
online
I was somewhat concerned reading Mark's posting earlier today.
Obviously, things are very bad in the US at the moment. Their
Government even deported Cat Stevens the other day (check
http://news.bbc.co.uk/1/hi/england/london/3686992.stm ).
Clearly, given the fact that Digium contributes so
-- snip --
Had the patch been against the actual g729 libraries the case would have
been clear. Now, the patch is against asterisk to make it interoperate
with the g729 libarary and this may or may not be non-infringing. However,
the distribution of the g729 libraries themselves are almost
DISCLAIMER: This code is free (I am not charging you to use it), but
you might have to pay royalty fees to the G.729 patent holders for using
their algorithm.
I finished this last Saturday and have had it on an Asterisk machine for
5 days without a crash, so I'm hoping that means it's safe to
I'm interested in the g729 diff you posted...
I've applied the patch, but I don't seem to have the prerequisites to
compile it... I tried downloading the other code available from
Intel, but even the 'eval' version won't install without a FlexLM
license (damn license managers...). Am I heading
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