On 26 Mar 2014, at 19:14, Mickael MONSIEUR mickael.monsi...@gmail.com wrote:
Hello,
When I get a SIP INVITE as follows:
INVITE sip:s@10.1.0.191:5060 SIP/2.0
Max-Forwards: 69
From: 0475XX sip:1053...@sip.domain.com;tag=as7df9ab18
To: sip:02XX@IP:5060
Contact: sip:1053212@IP:5060
6 jun 2013 kl. 17:41 skrev Daniel Pocock dan...@pocock.com.au:
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
will be fixed when there's someone
that needs to
fix it and provides funding for a developer to do it or have developer
resources, fix it and contribute
the code back to the project.
/O
- Miguel Baptista
On 3/13/2013 10:06 AM, Olle E. Johansson wrote:
; With the current situation, you
; With the current situation, you can do one of four things:
; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1
; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
; c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0
; d) Listen on
12 mar 2013 kl. 16:54 skrev Grant Bagdasarian g...@cm.nl:
Hello,
I’m noticing strange behavior in one of our Asterisk nodes where the ACK is
always sent to the proxy, but RR is not enabled for calls.
The proxy drops the ACK.
I’m using the AMI interface to originate a call:
Action:
10 mar 2013 kl. 03:04 skrev Nick Khamis sym...@gmail.com:
Hello Everyone,
I have gone through a few really good tutorials from the OpenSIPS
site, Asterisk resources etc.. The unanswered question (and final
piece of our puzzle) is if it's possible to have a register free
environment in an
6 dec 2012 kl. 16:54 skrev Danny Nicholas da...@debsinc.com:
Not sure about this since I use the 10/11 branches and not 1.8, but I think
you need to use the deprecated call-limit for BLF and the new busylimit for
the other features you need.
use in a database - or that you query from the dialplan with the realtime
function.
However, as stated earlier, this doesn't work in the SIP authentication that is
based on
the data in peers and users.
Regards,
/Olle
--
* Olle E. Johansson - o...@edvina.net
* Kamailio SIP Masterclass Miami
in a controlled way. There are just a few situations where you actually
benefit from having type=friend and match object names with Caller ID numbers.
/O
--
* Olle E. Johansson - o...@edvina.net
* Kamailio SIP Masterclass Miami FL December 2012
* http://edvina.net/training
23 okt 2012 kl. 22:31 skrev Kristian Kielhofner k...@kriskinc.com:
Hello everyone,
Does anyone know of a Wireshark AMI (Asterisk Manager Interface) dissector?
Decode as telnet and display filter telnet.data kind of work but TCP
reassembly can't happen without a better understanding of
running 1.4.43
The connection is alive and good and working. however, I see a bunch of
401 Unauthorized messages using wireshark - then it eventually registers
again
just fine.
Why would it not successfully register the first time - every time?
Jerry
---
* Olle E. Johansson - o
31 aug 2012 kl. 09:18 skrev Frederic Van Espen frederic...@gmail.com:
On Fri, 2012-08-31 at 00:11 +, Andrew White wrote:
Is realtime an option for you to install?
Andrew,
Realtime is not an option actually. We have a whole system built up that
generates configuration files.
The
31 aug 2012 kl. 13:13 skrev Steve Davies davies...@gmail.com:
On 31 August 2012 07:49, Olle E. Johansson o...@edvina.net wrote:
24 aug 2012 kl. 16:18 skrev Steve Davies davies...@gmail.com:
Hi SIP Gurus,
I've tried to find the relevant RFCs, but am struggling. I can find
the odd
31 aug 2012 kl. 16:58 skrev Shitian Long longst...@gmail.com:
Do you think it is a good way to use Manager API command action to
implement this feature?
No. The command action should be avoided since the output from the CLI commands
is not
made for parsing by applications and may change too.
17 aug 2012 kl. 03:15 skrev Phillip Frost:
On Aug 16, 2012, at 6:25 PM, Tiago Geada wrote:
forward to a Local extension that has dialplan requiring the
acknowledgement?
On 16 August 2012 21:12, Phil Frost p...@macprofessionals.com wrote:
I'd like to allow my users to forward their
8 aug 2012 kl. 14:07 skrev Kevin P. Fleming:
On 08/08/2012 06:30 AM, Kannan wrote:
Where can I get a complete set of RFCs and other specifications
supported by Asterisk?
To my knowledge there is no such list. In addition, Asterisk (like many other
pieces of software) does not claim
10 jul 2012 kl. 20:50 skrev Kevin P. Fleming:
On 07/10/2012 03:24 AM, Olle E. Johansson wrote:
The Asterisk SIP channel has no knowledge about interfaces and can't
bind to a specific interface for communication. In fact, it's a well known
bug that if you have multiple interfaces
11 jul 2012 kl. 00:26 skrev James Lamanna:
On Mon, Jul 2, 2012 at 12:13 AM, Olle E. Johansson o...@edvina.net wrote:
No.
This is probably because you are using phone numbers as names of devices
with type=friend in sip.conf.
That's generally a bad idea.
The SIP channel matches
will autocongest
Yes, that should propably change.
/O
Thanks,
Elliot
On Thu, Jul 5, 2012 at 12:31 PM, Olle E. Johansson o...@edvina.net wrote:
4 jul 2012 kl. 13:32 skrev Elliot Murdock:
Hello,
I am trying to get clarity with the sip.conf timer configuration. The
current configuration states
6 jul 2012 kl. 23:18 skrev Felix Salfelder:
Hi there.
i am seriously stuck with an asterisk and sip problem.
the following sip.conf works with respect to some_peer:
[general]
bindaddr = x.y.z.w
nat = no
[some_peer]
type=peer
host=somehost
secret=somesecret
some other
7 jul 2012 kl. 21:07 skrev Mikhail Lischuk:
Thomas Perron писал 07.07.2012 21:48:
exten = s,n,Dial(SIP/16175551212)
sip.conf
[general]
;register = 125010155:funnyti...@sip3.voipvoip.com/125010155
register = 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11
;
[incoming]
9 jul 2012 kl. 15:24 skrev Sergio Serrano:
Hi all
I hope that someone of you can solve this. Right now I'm stuck!
I'm using asterisk with some SIP extensions. Basically I want to
establish a call between desktop voip phone (ext 181) and embedded sip
system (ext 182)
All I can see
4 jul 2012 kl. 13:32 skrev Elliot Murdock:
Hello,
I am trying to get clarity with the sip.conf timer configuration. The
current configuration states:
;--- SIP timers
; These timers are used primarily in
22 jun 2012 kl. 21:59 skrev Bruce B:
Thanks. Want to secure everything and anything possible.
1- Can both SIP over TLS and SRTP work in conjunction to each other?
Yes. As Kevin said, SIP over TLS only secures the signalling. And it secures it
hop-by-hop so every server in the middle
can
Hello!
I will be running an Asterisk SIP Masterclass - the last one - in Barcelona in
June. During this week, I will organize a dinner for everyone working with or
interested in Asterisk, Kamailio and other Open Source platforms for realtime
communication. It's June 13th somewhere in Barcelona
16 apr 2012 kl. 15:31 skrev Matthew Jordan:
It's not a bug - decrementing the CSeq header field value is directly in
violation of RFC 3261. From section 22.2:
When a UAC resubmits a request with its credentials after receiving a
401 (Unauthorized) or 407 (Proxy Authentication
9 jan 2012 kl. 09:02 skrev Ronald Cepres:
Hi all,
I've been trying to register a SIP user agent to an Asterisk server using
OpenSIPS as SIP router. The functionality is working fine. However, Asterisk
uses the IP address of the OpenSIPS server as the peer IP address. How can I
use the
16 dec 2011 kl. 18:12 skrev Barry Miller:
On Fri, Dec 16, 2011 at 05:02:11PM +0100, Olle E. Johansson wrote:
16 dec 2011 kl. 02:03 skrev Barry Miller:
So is there a way for the dialplan to determine which device caused SIP to
auto-register an extension?
Not really, unless someone else
17 dec 2011 kl. 10:11 skrev Darrick Hartman:
The AstLinux Team is happy to announce the release of AstLinux 1.0.0. This
release includes significant changes and improvements over past releases.
Specific upgrade and new installation instructions are available at:
http://www.astlinux.org
16 dec 2011 kl. 11:29 skrev James Courtier-Dutton:
Hi,
I have a situation where unfortunately, I cannot use IAX for trunks,
and need to instead use SIP trunks.
Is there any way to fit the voice data from more than one simultaneous
phone call into a single IP packet over the SIP trunk.
I
16 dec 2011 kl. 02:03 skrev Barry Miller:
Hi all,
In sip.conf:
[general]
regcontext = autoreg
[devabc]
regexten = 543
creates exten= 543,1,Noop(devabc) in context autoreg when devabc
registers. But I can't use exten= _5XX,2,Dial(SIP/${EXTEN}) in the
dialplan, because there's
Friends,
SIPit is an event organized by the SIP Forum and partners. It has been running
for 15 years twice a year, making sure that SIP clients and servers
interoperate. By testing, we also find issues with the myriad of RFCs in this
area and correct them. Testing interoperability is
Friends,
While working with the manager interface, I noticed that an originate action to
a non-existing extension had a strange behaviour. Instead of generating an
error, which would happen in most VoIP channels and Dahdi, Asterisk started
looking for extension s as a fallback.
For as long
18 sep 2011 kl. 22:23 skrev Catalin S.:
Hello Eric,
Is about outgoing calls from multiple devices with the same username at aprox
same time. The overwritten is for incomming calls. I want to prevent using
the same account in multiple devices at same time. The solution with IP will
not
19 sep 2011 kl. 01:51 skrev Zeeshan A Zakaria:
This DTMF problem has always been there and there is no real solution for it,
other than using those expensive PBX systems like that from Avaya, Cisco,
etc. This problem happens when you are sending inband DTMF tones. Via
softphone you are
20 sep 2011 kl. 15:34 skrev Danny Nicholas:
Just my .02 - fix Originate since the Original Asterisk book, page 125
paragraph 1 says s = start. If s is not really start, I'm going to
scrap my 3+ years of dialplan writing and change all of my simple dialplans
to read exten= start,1,blah
20 sep 2011 kl. 15:34 skrev Danny Nicholas:
Just my .02 - fix Originate since the Original Asterisk book, page 125
paragraph 1 says s = start. If s is not really start, I'm going to
In the first edition, page 82, it actually says When a call enter a context
without a specific destination
8 sep 2011 kl. 17:17 skrev Kevin P. Fleming:
Honestly, I'm not really sure that there is a practical solution here. ISDN
overlap dialing was intended for 'dumb' phones, and SIP phones aren't 'dumb'
:-)
That's a quote that goes to my quote storage layer.
/O ;-)
--
8 sep 2011 kl. 17:26 skrev Andrew Latham:
On Thu, Sep 8, 2011 at 11:21 AM, Olle E. Johansson o...@edvina.net wrote:
8 sep 2011 kl. 17:17 skrev Kevin P. Fleming:
Honestly, I'm not really sure that there is a practical solution here. ISDN
overlap dialing was intended for 'dumb' phones
6 sep 2011 kl. 22:30 skrev Leif Madsen:
On 02/09/11 11:27 PM, Joseph wrote:
In asterisk 1.4 I had:
exten = s,n,Answer()
exten = s,n,SetMusicOnHold(default)
But in 1.6 1.8 I think don't need to use: SetMusicOnHold(default)
(beside it is deprecated) as it is default.
In 1.6 and UP I think
7 sep 2011 kl. 15:59 skrev Daniel Tryba:
Looking at the history of the list I don't expect any answer but lets
try anyway:
Does anybody use overlap dialing from SIP devices to asterisk? Does
anybody have a working example?
To add to your question: Does anyone have a phone that supports
7 sep 2011 kl. 16:20 skrev Andrew Latham:
On Wednesday, September 7, 2011, Olle E. Johansson wrote:
7 sep 2011 kl. 15:59 skrev Daniel Tryba:
Looking at the history of the list I don't expect any answer but lets
try anyway:
Does anybody use overlap dialing from SIP devices
31 aug 2011 kl. 14:42 skrev Kevin P. Fleming:
On 08/31/2011 02:46 AM, Jaime Lozano wrote:
Hello,
I agree with you, I'm not explaining the problem in a proper manner,
because of my lack of Asterisk knowings. I send the Wireshark captures.
3com telephones take the timezone TZ:7200 from the
29 aug 2011 kl. 15:05 skrev Kevin P. Fleming:
On 08/28/2011 01:56 AM, Tzafrir Cohen wrote:
On Thu, Aug 25, 2011 at 07:36:53PM +0100, Chris Hastie wrote:
Hi
I've just added direct support for AMI to a forthcoming version of
TBDialOut, a Thunderbird extension for dialling direct from
26 aug 2011 kl. 14:06 skrev Jaime Lozano:
Hello,
In which file do I use SIPAddHeader()?
Please consider that the packet goes from the PBX to the telephone, and what
I want is not a header because the TZ: 7200\n is in the *message body* not
in the *message header*.
That's no longer a SIP
12 aug 2011 kl. 14:51 skrev Kevin P. Fleming:
On 08/11/2011 02:03 AM, Jim Boykin wrote:
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Next week I'll be in the hot city of Madrid doing Asterisk/Kamailio training -
The SIP master class.
Maybe we can organize a voip nerd dinner on Thursday evening? If you're
interested, please e-mail me off list and I'll send out more details later.
Greetings
/Olle
PS. E-mail off list :-)--
31 maj 2011 kl. 14.49 skrev Benny Amorsen:
Jeff LaCoursiere j...@sunfone.com writes:
Hasn't anyone managed to solve this with something better than a
caching DNS server, which seems to only last a short while? What
exactly is going on that is failing?
If your recursive DNS server
23 maj 2011 kl. 23.36 skrev Paul Belanger:
On 11-05-23 05:30 PM, Elliot Murdock wrote:
Hello,
I am wondering how to send a call to a specific IP address that is different
than the host of the URI. For example, an invite to the URI is
j...@phone.com needs to be sent to the IP address
24 maj 2011 kl. 12.19 skrev Tony Mountifield:
One of our customers has an Asterisk conference bridge connected to a
SIP trunk from an ITSP. Yesterday, they had two inbound calls that
didn't get hung up properly. From the tcpdump SIP trace that we have
running continuously, I can see that no
5 maj 2011 kl. 18.30 skrev Ira:
At 07:56 AM 5/5/2011, you wrote:
So how can we fix this? How can we get more people involded? What makes
projects like FedoraTesting[3] and DebianTesting[4] popular? How can the
Asterisk project reproduce their success?
Well, it's not a lot of people
5 maj 2011 kl. 05.28 skrev Flavio Goncalves:
My 2 cents. All these problems seem to be lack of focus. Digium,
please stop doing everything to everyone. Too many versions, too many
features, too many code, too many bugs. Following the Pareto's
principle, 80% of the users use only 20% of the
5 maj 2011 kl. 06.33 skrev Olivier:
2011/5/5 Flavio Goncalves fla...@asteriskguide.com
snip
but stuffing Asterisk with
many new features on each version does not seem to be contributing to
the stability of the code or the migration to newer versions.
yes but it seems to me that code
5 maj 2011 kl. 12.04 skrev Paul Hayes:
On 05/05/11 05:41, Cary Fitch wrote:
Flavio E. Goncalves
www.asteriskguide.com http://www.asteriskguide.com
Compare to which version of Windows… Patches are a never ending process
Cary Fitch
I think this attitude is half the
5 maj 2011 kl. 14.08 skrev bilal ghayyad:
Hi All;
When the endpoint register on Asterisk or initiate a call, so they exchange
the sip username and password. What is the possibility that this will be
capture by the hacker and how to avoid this problem?
We never exchange passwords in
5 maj 2011 kl. 14.17 skrev Alex Balashov:
Bilal,
On 05/05/2011 08:08 AM, bilal ghayyad wrote:
When the endpoint register on Asterisk or initiate a call, so they
exchange the sip username and password. What is the possibility that
this will be capture by the hacker and how to avoid this
5 maj 2011 kl. 15.11 skrev Paul Hayes:
On 05/05/11 14:04, Jonas Kellens wrote:
Hello list,
what does this mean :
[May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered
elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the cause
code, buddy. The cause code!!!
5 maj 2011 kl. 16.35 skrev Gilles:
On Tue, 3 May 2011 13:48:34 -0500, Shaun Ruffell sruff...@digium.com
wrote:
I know this thread is dead but: I do not believe this should go into the
DAHDI
kernel modules.
I agree. It's just too bad Dahdi is unable to report how an outgoing
call is
Here is the thing, there is nothing stopping 'the community' today from doing
this. In fact, we already have a testsuite [1] in place, running each
subversion commit and producing results for the last year. But this is only
one type of testing; automated, we also have unit tests built
4 maj 2011 kl. 19.44 skrev Robles Román, José Miguel:
By the way, I like the implementation in iax.conf (auth=md5 ...
secret=x), it seems more flexible, and it enables the use of other hash
functions or other security algorithms.
The SIP protocol does not support any other hash
2 maj 2011 kl. 18.09 skrev Hans Witvliet:
On Thu, 2011-04-28 at 11:25 -0400, Bruce B wrote:
Hi everyone,
How can I introduce some distortion, echo, chopping sound and all
other bad quality things that can happen to a SIP trunk? I have plenty
of bandwidth and crisp clear lines so the
I don't think there's anything inherently wrong with the bug tracking system.
It's more of a resource issue with many conflicting priorities. Officially
letting off some of the pressure from older branches does help. I would like
to be making faster progress through bug reports and
29 apr 2011 kl. 01.49 skrev Leif Madsen:
Well the issue is that we currently have over 900 open issues in the Asterisk
project alone, and with only one primary bug marshal (myself) sometimes things
accidentally get closed if it looks like a configuration issue.
What's the reason that we only
28 apr 2011 kl. 16.53 skrev Russell Bryant:
- Original Message -
PS. Please don't start a discussion about 1.8 quality in this thread,
that's a separate issue. I just want to know what you think about
closing 1.4 support now. If you want to discuss 1.8 quality, start a
new thread.
Friends,
We have a discussion on asterisk-dev about the maintenance of the 1.4 branch.
According to the release plans, support for 1.4 was scheduled to close in April
2011 - basically now. After that, only security patches would be committed.
This is already a delay from the original plan
I(me, my opinion, my feelings, my commercial view) am on the side of
dropping support for 1.4 and 1.6. 1.8 had some major issues which are
resolved/being worked on with more energy as older platforms are shut
down. If a large enough security issue showed up, I hope we would all
try to do
Friends,
After having spent many years working with the Asterisk SIP channel driver and
the SIPv2 protocol, I have finally realized that this is a dead end. It's
getting nowhere and it's way too complicated to set up, run and support in
working code.
After realizing this, I started a new
16 mar 2011 kl. 14.13 skrev Benny Amorsen:
Kevin P. Fleming kpflem...@digium.com writes:
Why do you need a Local channel to do this? If extension 234 exists in
some context, the Dial() statement in that extension can dial
SIP/234-foo and SIP/234-bar itself.
Good point.
It can be a
3 jan 2011 kl. 00.26 skrev Bryan Field-Elliot:
Normally, no matter which Asterisk server an ATA connects to, we get our
database fields filled out correctly, such as regseconds, lastms,
ipadr, etc. However, with some ATA's we are experiencing a problem as
follows:
1. ATA reaches its
11 nov 2010 kl. 23.25 skrev Baha @ SH:
Hello
How can I run the sip service on asterisk on another port beside 5080?
I mean asterisk will still take sip requests on port:5080 and another custom
port, lets say port:6080
For UDP, we only have one port. You have to select.
/O
--
10 nov 2010 kl. 21.48 skrev Hans Witvliet:
On Wed, 2010-11-10 at 08:38 +0100, Olle E. Johansson wrote:
6 nov 2010 kl. 15.30 skrev Hans Witvliet:
Hi all,
As stated in the subject, slightly off-topic, as it is not directly a
Asterisk issue, but more SIP in general
Because security
10 nov 2010 kl. 02.38 skrev Brett Woollum:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9
6 nov 2010 kl. 15.30 skrev Hans Witvliet:
Hi all,
As stated in the subject, slightly off-topic, as it is not directly a
Asterisk issue, but more SIP in general
Because security in general, and specifically identification becomes
more and more a subject for more concern, and Asterisk is
31 okt 2010 kl. 13.43 skrev Paul Belanger:
On Sat, Oct 30, 2010 at 6:22 PM, Brian Capouch bri...@palaver.net wrote:
I wonder if anyone out there has a perspective on this. There are a
welter of tickets out there on the matter, most of them closed.
I'm actually able to reproduce this
2 nov 2010 kl. 17.19 skrev Olivier:
Hi,
In Europe many Telcos implement power-save mode
(See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to
'Activation / Deactivation' for more information).
Would you agree to have this feature added to the ones already discuused for
Friends,
After listening to Mark Summer's keynote at Astricon (hopefully soon on the
Astricon web site) I think we should come back to the discussion he started.
Mark talked about using Open Source in general and Asterisk in particular in
third world projects as well as in emergencies in other
11 aug 2010 kl. 15.49 skrev Leif Madsen:
On 10-08-10 04:11 AM, Olle E. Johansson wrote:
26 jul 2010 kl. 18.13 skrev Leif Madsen:
On Asterisk 1.6.2, your only option for distributing device state is with
res_ais. I've used it in a labbing system and it works well -- the caveat
26 jul 2010 kl. 18.13 skrev Leif Madsen:
On 10-07-26 10:45 AM, Mathieu wrote:
Hello,
as I'm looking for a solution (with asterisk 1.6.2) , my
investigations leaded to :
- res_ais = libais corosync. (each node need to run corosync / aiexec)
- res_jabber = libjabber iksemel. (each
Further to Steve Edward's comment, I think things would be more
obvious if the help system was improved slightly, for instance:
If you were trying to figure out the commands dealing with peers, you
would be able to type:
*CLI help peer
No peer command found. Possible alternatives:
,
including the Asterisk Diva Allison Smiths' attractive voice. In addition to
Allison
Digium added the voice of the southern gentleman Danny Wyndham and the
Swenglish dialect of Asterisk developer and guru Olle E. Johansson, one that
was recognized with a strange smile by all Asterisk developers
25 mar 2010 kl. 13.14 skrev Michelle Dupuis:
I can't find this in the wiki/email history..but I'm sure it's based asked
before.
The port range define in rtp.conf - is that for connections initiated by
asterisk? Or the port range asterisk listens on? Or both?
These are the ports
24 mar 2010 kl. 16.48 skrev Karl Fife:
Steve Edwards wrote:
It may not be as intended, but from a user standpoint, it seems
logical
and convenient to establish policy in [general] and make exceptions in
the entities as needed.
Right... for when you have one policy. When you have two
23 mar 2010 kl. 22.20 skrev Kevin P. Fleming:
Steve Edwards wrote:
It may not be as intended, but from a user standpoint, it seems logical
and convenient to establish policy in [general] and make exceptions in
the entities as needed.
Right... for when you have one policy. When you
21 mar 2010 kl. 18.22 skrev Philipp von Klitzing:
Hi Olle!
The work I started during Christmas - Named ACL's - is a starting point
that other developers can use to develop all kind of schemes.
http://www.voip-forum.com/asterisk/2010-01/manageable-access-control-lists
-asterisk-nacls/
22 mar 2010 kl. 14.54 skrev Kevin Sandy:
On 3/21/2010 4:05 AM, Olle E. Johansson wrote:
17 mar 2010 kl. 16.37 skrev Kevin Sandy:
We're having an odd issue with codec negotiation from one of our
SIP providers. Here's the basic situation.
We receive an invite from them advertising
Friends,
Daniel and I are running a Kamailio SIP Masterclass this week in Berlin. When
travelling around like this, we often invite the community to come and meet us
in a nice restaurant. We offer good company and fun discussions about Kamailio,
SIP-router.org and Asterisk - but the drinks and
17 mar 2010 kl. 16.37 skrev Kevin Sandy:
We're having an odd issue with codec negotiation from one of our SIP
providers. Here's the basic situation.
We receive an invite from them advertising support for G711, G729, and G723.
In our response, we send back that we support G711 and G729.
19 mar 2010 kl. 03.41 skrev Philipp von Klitzing:
Hey hey!
My first step will be to strengthen the passwords in use, and for the
hardphones to restrict by IP address, but that still leaves the
softphone quite widely open.
Asterisk doesn't differentiate between a hard phone and a soft
12 mar 2010 kl. 10.45 skrev Katerina Borin:
Probably has anyone idea how dtmf payload type could be changed in Asterisk
say to 100?
On Wed, Mar 10, 2010 at 2:53 PM, Katerina Borin katerin.bo...@gmail.com
wrote:
Hello,
I encountered the dtmf problem between my asterisk box (1.4.23) and
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
---
* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
11 mar 2010 kl. 15.17 skrev Philipp von Klitzing:
Is there a way for a client to tell a server where it is registered to
remove the registration?
Yes, it needs to send an UNREGISTER sip message.
There's actually not an UNREGISTER method in SIP.
As Kevin stated, you send a REGISTER with a
8 mar 2010 kl. 11.13 skrev Peter Childs:
On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote:
At an Asterisk CLI use the command manager show commands.
Life is rarely that simple, and this does not really answer the question.
Oh and Channel can mean different things in
Friends,
SIPit is the main interoperability event for all things SIP. It's organized by
the SIP Forum and creates good
feedback to the IETF. Asterisk has been participating in SIPit during many
years and in many variants
- videocaps, Marc Blanchet's IPv6 branch and the standard Digium
27 feb 2010 kl. 08.26 skrev Olle E. Johansson:
26 feb 2010 kl. 22.02 skrev JT:
Hmmm I agree that altering sip.conf with the RTP timeouts are somewhat
of a band-aid to the issue. But in my observations there is one clear
indicator that I am shocked is not used.
When I have done
26 feb 2010 kl. 22.02 skrev JT:
Hmmm I agree that altering sip.conf with the RTP timeouts are somewhat of
a band-aid to the issue. But in my observations there is one clear indicator
that I am shocked is not used.
When I have done this test - pulling the network cable on a device
24 feb 2010 kl. 01.22 skrev Kristian Kielhofner:
On Tue, Feb 23, 2010 at 8:22 AM, Michelle Dupuis supp...@ocg.ca wrote:
We're creating a SIP gateway for a client that will take one leg of a call
in via SIP, and out the other side via H.323. To minimize load on the
gateway, we would like to
/listinfo/asterisk-users
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* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
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22 feb 2010 kl. 07.23 skrev Tilghman Lesher:
open audio {tcp|udp} hostname portno
close audio
If you design something now, I would strongly suggest that we stop using
audio as an attribute. Each call will have multiple media streams - and
already have. You need to be able to select which
23 feb 2010 kl. 03.18 skrev Kevin P. Fleming:
Kirill 'Big K' Katsnelson wrote:
The caveat here is that it is perfectly normal NOT to transmit any RTP
data in case of long silence. This is why the SIP timers were introduced
in the first place: there is no correct way to detect when the
23 feb 2010 kl. 01.47 skrev Kirill 'Big K' Katsnelson:
On 100222 1313, JT wrote:
When a SIP device dials another SIP device...Asterisk connects the calls and
displays the channel information.
If one of those SIP devices hangs up, Asterisk receives the hangup notice
and disconnects the
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