Friends,
i am happy to see so many participating in testing bugs and
discussing new features in the bug tracker.
But please, try to keep the comments in the bug tracker related to
the issue at hand.
Other issues are better discussed on the mailing list.
The bug tracker is not a generic disc
Error 1
make[1]: Leaving directory `/root/test-this-branch/channels'
make: *** [subdirs] Error 1
This is fixed. Run svn update and you will get new code.
Thank you for reporting this. Keep testing!!!
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Train
9 mar 2006 kl. 11.20 skrev santosh y:
I'm very new to Asterisk, I'm tracing the Asterisk code,
i'm feeling difficulty in understanding the code, so please tell me
where i can get the documentation of the code and,
design and architecture of the code
_
http://www.asterisk.org/doxygen
http://ww
A very important fact about being a bug marshal:
*** YOU DON'T HAVE TO BE A DEVELOPER!
We need people that work with the process in the bug tracker and make
sure that everything is kept up to date, that we get all the
information we
need to track down a bug, maybe spend some time testing stuf
http://bugs.digium.com/view.php?id=6355
Would be really nice to get some feedback on this work - it is part
of the
test-this-branch branch and a separate branch. Check the bug report for
more information about what it is and helps you with.
Thanks,
/Olle
__
; branch. Check it out from
http://svn.digium.com/svn/asterisk/team/oej/siptransfer
There are tons of comments within chan_sip that tells you more about
this work.
At some point, I will integrate this into the test-this-branch branch
as well, but not yet.
Regards,
/Olle
* Olle E Joh
Paul Cadach wrote:
Hello,
Olle E Johansson wrote:
The RTCP branch includes improved support of RTCP, but also a
reporting facility we do not use currently. Would it be useful to add
this to a channel variable - or even better a CDR variable - so you
can add it to CDRs and make reports based
24 feb 2006 kl. 16.39 skrev Dave Cotton:
On Fri, 2006-02-24 at 15:47 +0100, Olle E Johansson wrote:
Dear Asterisk user and developer,
I am sorry, but I have to tell you that there is a long boring
weekend
ahead of you. Nothing to do, bad weather and boring sports on TV.
What
to do?
I
27 feb 2006 kl. 13.00 skrev BJ Weschke:
On 2/27/06, Andreas Sikkema <[EMAIL PROTECTED]> wrote:
The RTCP branch includes improved support of RTCP, but also a
reporting facility we do not use currently. Would it be
useful to add
this to a channel variable - or even better a CDR variable - so you
The RTCP branch includes improved support of RTCP, but also a
reporting facility we do not use currently. Would it be useful to add
this to a channel variable - or even better a CDR variable - so you
can add it to CDRs and make reports based on it?
/Olle
24 feb 2006 kl. 18.18 skrev Saul Diaz:
Olle E Johansson wrote:
Denis Smirnov wrote:
On Wed, Nov 23, 2005 at 02:08:51PM +0100, Olle E. Johansson wrote:
OEJ> I had a patch that did the opposite, and that's where we are
going.
OEJ> I want to have "register=yes" with
(Sent to all Asterisk-related mailing lists I can find, my apologies for
being desperate :-)
--
Friends,
The developer team for Asterisk not only consists of coders - a very
important part are the testers, those that test new code and give
feedback.
For a fe
21 feb 2006 kl. 21.10 skrev Kevin P. Fleming:
Andrew Kohlsmith wrote:
Well I suppose that a lot of us just don't have any idea what this
feature
really gets us... Even with BKW's valetparking I never truly
understood the
utility of multiple parking lots in one PBX unless you were
hostin
Tzafrir Cohen wrote:
On Fri, Feb 10, 2006 at 02:56:59PM +0100, Michael Prochaska wrote:
Olle E. Johansson schrieb:
...write an RFC :-)
i don't think that this is necessary :-)
The MD5 is in the SIP RFC, and I've never seen anyone using SHA.
no, md5 is NOT in the SIP RFC. H
Michael Prochaska wrote:
Olle E. Johansson schrieb:
...write an RFC :-)
i don't think that this is necessary :-)
The MD5 is in the SIP RFC, and I've never seen anyone using SHA.
no, md5 is NOT in the SIP RFC. HTTP digest authentication is not
automatically md5
You are
5 feb 2006 kl. 11.33 skrev asterisk_dev:
Hi;
I've seen on bugs.digium.com the main developer is going to be
busy, and I'd like to take over this. I've experience developing
protocols over Linux (have done OSPF and BGP implementations). Do
everybody agree?
Great.
The patch in the bug tr
2 feb 2006 kl. 17.20 skrev Enzo Michelangeli:
Wasn't the "s" extension supposed to mean "the null extension"?
http://www.digium.com/asterisk_handbook/extensions.conf.html says:
s: Defines how to route a call when no other routing information
has been
received. On a PRI or local FXS line, we
...and here's a SIP draft.
http://www.ietf.org/internet-drafts/draft-jesske-sipping-tispan-requirements-02.txt
This is not the account code, more a hint of what the provider will
charge for the call as I understand it now after some research.
/O
___
Koopmann, Jan-Peter wrote:
Hi,
The AOC information tells you the exact cost for the call while the
accountcode
> specifies where the cost should be charged to in your company. This
might explain
> our misunderstanding in bug 6152. There currently seems to be no
way
of storing
> the AOC i
Barry Flanagan wrote:
Hi,
Can someone explain to me or point me in the direction of documentation
for the domain support feature in 1.2.x?
Specifically I need to be able to have sip users who are authenticated
as [EMAIL PROTECTED]
For authentication, we only look at
1) Whether the domain is
Barry Flanagan wrote:
Hello,
I have a situation where I need to differentiate between registrations
by users where there might be clashes on the left hand side (username)
portion of the SIP From URI. (for a multi-domain virtual hosting system)
It seems that only the username portion is used for
Denis Smirnov wrote:
Why asterisk binary links with ssl, where only res_crypto must be linked
with it?
I create RPM-distribution for Asterisk and don't wont that asterisk
package requires openssl, but crypto can be used when subpackage with
res_crypto installed.
We use OpenSSL for authenticati
Please try my multiparking and the hints for parkinglots in chan_local,
both available in the issue tracker.
/O
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Koopmann, Jan-Peter wrote:
On Monday, January 30, 2006 4:19 PM Olle E Johansson wrote:
What is the difference between AOC and our use of Accountcode in
Asterisk?
The AOC information tells you the exact cost for the call while the accountcode
> specifies where the cost should be char
Fernando Romo wrote:
Dear developers:
I start to write a CSTA (Computer Supported Telecommunications
Applications) protocol wrapper for the API Manager.
My Goal is interact with a Matra PBX and other brands supporting
CSTA-III protocol to interact CTI and Dialer applications.
Exists any effort
Friends in the developer community,
We now have over 250 open issues in the issue tracker. We need all the
help we can get to move through this and clear it out so it's manageable
again.
We need your help with
* Testing
* Code reviews
* Portability issues
All confirmations of a test makes i
Luigi Rizzo wrote:
The following uncommented code in res_features.c::ast_park_call()
is extremely expensive because it has O(N*M) cost, where N is the
size of the parking lot, and M is the number of parked calls:
for (i = 0; i < parking_range; i++) {
x = (i + parking_offs
Bruno Rocha wrote:
Hi!
I've configured two Asterisk 1.2.1 machines: astA.domain.tld and
astB.domain.tld
astA sip users: mary ([EMAIL PROTECTED]), john ([EMAIL PROTECTED])
astB sip users: alice ([EMAIL PROTECTED]), mary ([EMAIL PROTECTED])
All users are behind NAT and using their respective as
SIP calls are not connected to each other. In the Asterisk architecture,
the SIP "call" is connected to an owner channel. Each call has a pair
of a "technology" driver structure (tech_pvt) - like IAX, SIP, H323, ZAP
and they connect to a generic Asterisk channel that is the "owner".
So when Bob ca
Luigi Rizzo wrote:
> i think i have seen it mentioned before but cannot find a relevant
> patch in mantis...
>
> I was wondering if there is a way to grab the register info
>
> register => user[:secret[:[EMAIL PROTECTED]:port][/contact]
>
> from a peer section, e.g. using a syntax like
>
Call through the local channel and add the alert_info headers.
/O
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John Todd wrote:
>
> Has there been any progress in making IM integration into Asterisk?
Yes, work is going on.
We have no support for SIMPLE messaging outside of a call, only some
amount of line notification. Within a call, you can send text with
sendtext() in the dial plan to SIP phones that su
< Arnaud > wrote:
> We are looking into mobile users that change their IP address in the
> middle of a VoIP call. We use SIP and Asterisk does not reinvite the
> two ends i.e. Asterisk acts as a B2BUA and sees both the SIP and the
> RTP traffic.
>
> We've been searching chan_sip.c, rtp.c and other
> Where's O'reilly, I'm gonna write the second Asterisk book. It's gonna be
> titled "The Asterisk Developer's Handbook: all the info Digium isn't keeping
> from you, but isn't making easy to get, either" :-)
:-)
So as you find out that stuff, please add it to the Doxygen docs. We
have to st
Jason Pyeron wrote:
> On Sat, 22 Oct 2005, Olle E. Johansson wrote:
>
>> Jason Pyeron wrote:
>>
>>> On Fri, 21 Oct 2005, Olle E. Johansson wrote:
>>>
>>>> I fail to see how you MUST have this to get shared line apperances...
>>>> Ple
Short answer is yes, the work with timer t1 was just a start to
implement the rest of the series of SIP timers. Won't be in 1.2, but
work will continue and I would gladly accept any kind of support for
this work, including money :-)
We also need to implement correctly timers for keeping SIP sessio
Jason Pyeron wrote:
> On Fri, 21 Oct 2005, Olle E. Johansson wrote:
>
>> I fail to see how you MUST have this to get shared line apperances...
>> Please explain.
>
>
> That is how it is done in pbx's which support it, ex SNOM 4S
>
Does not mean we have to
Sherwood McGowan wrote:
> Olle,
>
> Thanks for your input. I'll definitely be just adding a section in the
> sip.conf page that says "** New In 1.2+ **". The problem with incominglimit
> and outgoing limit (I haven't changed over to call-limit, will be today)
> causing registration problems is thi
Gunnar Schaller wrote:
> Hi,
> A nice feature would be the useragent in realtime database. Difficult
> or simple change?
>
We can do everything, but the real question is how many database actions
we want per call/registration?
/O
___
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Michal Olejnik wrote:
> I didn't get answer for my question on -users list so I try to get
> answer here. I have Asterisk 1.0.9, when INVITE contain "From: "1234
> <1234>" ;" ${CALLERIDNUM} is 1234. Is it
> correct or is it bug?
>
Well, the sender told us to use the caller ID 1234, so it is not a
Seems like all my IAX2 connections are dead until I change from RSA to
MD5 auth with CVS head...
I vaguely remember someone else reporting this somewhere, but can't find it.
Is it only me or is someone else seeing this?
/O
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After months in the bug tracker, we've finally committed a lot of
changes to the SIP Subscribe subsystem in Asterisk cvs head.
* It now works even if you reload the dial plan
* It does not accept subscriptions to extensions without hints
* It will terminate subscriptions if the hint does not exist
Chee Foong wrote:
> Hello,
>
> I have tested with the vendor, they send a CANCEL instead of BYE. Another
> problem came out:
>
> When the other end send CANCEL, asterisk response with "487 Request
> Terminated" then a "200 OK" for the CANCEL.
> The problem now is the other end expect "200 OK" fir
Chee Foong wrote:
> But I still dont understand why asterisk response to the BYE with OK since
> it is not permited at the stage and leave the channels hanging.
>
As I reported in another mail, I just found out that the caller can send
a BYE on a non-answered call, I was wrong. We just have to can
Mikael Magnusson wrote:
> On Thu, Aug 18, 2005 at 10:50:42PM -0500, Kevin P. Fleming wrote:
>
>>Chee Foong wrote:
>>
>>>But I still dont understand why asterisk response to the BYE with OK since
>>>it is not permited at the stage and leave the channels hanging.
>>
>>You are correct, Asterisk's res
harry gaillac wrote:
> Hello,
>
> I patched asterisk cvs head sources with
> http://juraj.bednar.sk/work/software/asterisk/messaging/
> and presnce patch without success.
>
> asterisk send "405 method not allowed" to sender.
> I use polycom ip300.
THat is a response to the polycom's PUBLISH req
Mikael Magnusson wrote:
> On Thu, Aug 18, 2005 at 10:50:42PM -0500, Kevin P. Fleming wrote:
>
>>Chee Foong wrote:
>>
>>>But I still dont understand why asterisk response to the BYE with OK since
>>>it is not permited at the stage and leave the channels hanging.
>>
>>You are correct, Asterisk's res
> There's also Integrated Services Digital Network (ISDN) User Part (ISUP)
> to Session Initiation Protocol (SIP) Mapping" aka
> http://www.ietf.org/rfc/rfc3398.txt and "Internal Cause Code Consistency
> Between SIP and H.323" aka
> http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_f
John Todd wrote:
>
>
> While there are certainly arguable points here, I'd suggest that this is
> relevant to the comparison:
>
> http://www.zvon.org/tmRFC/RFC3398/Output/index.html
>
> This defines a standard for ISUP-to-SIP mapping, which could possibly be
> used for PRI-to-SIP mapping, as ma
Kevin P. Fleming wrote:
> Olle E. Johansson wrote:
>
>> I would say that the core problem is that we send NOTIFY messages
>> WITHOUT a preceeding SUBSCRIBE for notifications, so there's no
>> transaction to match. The NOTIFY has nothing to do with the actuall call
&
Kevin P. Fleming wrote:
> Chris A. Icide wrote:
>
>> In the following message, there is a tag after the From: header. What
>> is the function of this tag? The MWI indication comes right after the
>> call that left a voicemail, but the tag isn't the same as the tag on
>> the previous call. As yo
Please check the latest cvs. I kind of remember this was fixed in CVS
head, but since this is the busy day before Astricon Europe, I might be
totally wrong and more confused than usual :-)
/O
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ht
The real solution is *not* to mix extensions in the dial plan (phone
numbers) with device names. That will always lead to some sort of
problem. In some cases, it also will confuse the manager (me or you).
You need different name spaces for extensins and device names (peers/users).
Extensions are w
Mikael Magnusson wrote:
> Hi,
>
> It's often mentioned on the list that Asterisk is an user SIP agent and not a
> proxy. But why is then Proxy-Authenticate (and 407 Proxy
> Authentication Required) used instead of WWW-Authenticate
> (and 401 Unauthorized) in chan_sip. According to section 22.1 Fr
Steve Clark wrote:
> Hello List,
>
> Has any work been done or being done in asterisk to support rtp
> redundancy as described
> in rfc 2198?
>
Please describe this a bit more!
/O
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>
> The username field is used only for users. I suspect it would use it
> if you changed the above type=peer to type=friend.
>
>
>>So, are both intended behaviours or should I file a bug report ?
>
>
> Yes. ;-)
Tilghman,
The username field is *not* for users, it's for peers. Please read th
Leif Madsen - Certified Asterisk Consultant wrote:
> On 5/20/05, Russell Bryant <[EMAIL PROTECTED]> wrote:
>>Tilghman Lesher wrote:
>>>I'm generally against trying to change the function syntax at this point
>>>in time, unless someone comes up with a really nice alternative syntax.
>>
>>I agree. I
Leif Madsen - Certified Asterisk Consultant wrote:
> On 5/17/05, Tilghman Lesher <[EMAIL PROTECTED]> wrote:
>
>>As soon as you nest that deep (3 or more levels), it's going to look
>>ugly, no matter what the syntax. It's not that big a deal; move on.
>
>
> I disagree. By having curly braces you
Steve Davies wrote:
> Hi,
>
> A couple of days ago I posted regarding SIP 'SUBSCRIBE's failing from
> our snom190 phones here:
> http://lists.digium.com/pipermail/asterisk-users/2005-May/106545.html
>
> I did some more analysis, and it seems that the problem is caused
> because each of our ph
Steve Davies wrote:
> On 5/13/05, Steve Davies <[EMAIL PROTECTED]> wrote:
>
> [snip]
>
>>The failure was due to an inability to authenticate the SUBSCRIBE, and
>>this was happening when the phone was trying to SUBSCRIBE using
>>credentials from "SIP/line1", and find_peer(...) was returning
>>cred
Brian Capouch wrote:
> Preston Garrison wrote:
>
>> Source code is your documentation :)
>>
>
> All 200K+ lines of it!!
>
> Good luck. . . there aren't many comments either. This approach keeps
> the "academics" away, with all the attendant slowdown in development
> that entails.
>
And all cal
Nicolás Gudiño wrote:
Event: Hold
Holdstatus: On | Off
Event: ParkedCall
PCstatus: Parked | Unparked | GivingUp
Why not just "Status" or "State"? There's not really any particular
reason why each event needs its own state header name, is there?
I totally agree...
Do you agree that we should use
Kevin P. Fleming wrote:
Olle E. Johansson wrote:
Event: Hold
Holdstatus: On | Off
Event: ParkedCall
PCstatus: Parked | Unparked | GivingUp
Why not just "Status" or "State"? There's not really any particular
reason why each event needs its own state header name, is th
For those of you that missed the start of this thread:
Please read:
http://edvina.net/asterisk/alphanumericextensions.pdf
It's a proposal by me and Leif Madsen on how to be able to support
International character sets in Asterisk in a good way. The reason I
published this document now was the IAX
Brian West wrote:
Please make the changes so that configs & tools in the old
format are still working . Tia , JimL
--
You do realize that at some point we have to break backwards
compatibility moving forward. Otherwise we get stuck working around
BAD IDEAS that were done very badly in
It's worth enduring the "Don't change the subject!!" tirades, IMO.
:-)
The problem for me is not the threading, is that in order to survive, I
have to delete a lot of mails without reading them. And without proper
subjects, it's easy to just delete a lot of interesting stuff. So with
your non-n
Tilghman Lesher wrote:
On Saturday 30 April 2005 05:04, Olle E. Johansson wrote:
Since it's obvious that this is an unresolved issue, we should
avoid the issue at this juncture in the IAX2 spec and simply
specify that ASCII is the character format. If at some point in
the future these argu
Kristian Nielsen wrote:
Steve Underwood <[EMAIL PROTECTED]> writes:
There is no need to have ASCII + UTF-8. ASCII is a subset of UTF-8, so
they are fully compatible. Its only when you have 8 bit sets, like the
PC ones, that compatibility is an issue. Just define that all strings
in IAX2 are UTF-8,
Brian Roy wrote:
Since Ollie and Steve wrote this and are on the forum...
It's "Olle"
Two things...
1) Excessive NOTICE messaging in the Console as follows:
Will look into this. It's a result of their 15 second re-registration time,
that will be the effect ONLY if you run your Asterisk behind NAT
John Todd wrote:
At 8:48 AM +0200 on 4/28/05, Olle E. Johansson wrote:
Good morning, bonjour, god dag!
I can't find anything to argue with in your proposal. The inclusion
(well, exclusion) of '@' has been a thorn in my side a few times, and
while I can't speak f
Amazingly, though, they all start with the function named 'sip_call' in
chan_sip.c. Imagine that, who would have thought that the function that
initiates calls would be named 'sip_call'... must be pretty hard for
people to believe, since you weren't able to find it
Chan_sip is also one of the
Tzafrir Cohen wrote:
On Thu, Apr 28, 2005 at 05:15:34PM +0200, Olle E. Johansson wrote:
Steve Underwood wrote:
Hi,
I raised this with Mark ages ago, when I started putting Chinese into
IAX2 messages. I thought it should be specified that all text is Unicode
in UTF-8 form, but he seemed pretty
Steve Underwood wrote:
Olle E. Johansson wrote:
Steve Underwood wrote:
Hi,
I raised this with Mark ages ago, when I started putting Chinese into
IAX2 messages. I thought it should be specified that all text is
Unicode in UTF-8 form, but he seemed pretty indifferent to specifying
anything
Goldenear wrote:
Olle E. Johansson a écrit :
Kevin P. Fleming wrote:
Goldenear wrote:
STUN support for asterisk has been discussed since a long time.. why
isn't it implemented yet ? is it so difficult to add STUN support in
chan_sip ?
You are aware that Asterisk is an open source pr
The separate events used at the moment breaks the command-response
pattern. I would prefer of the events mechanism was used only for
unsolicited events.
I fully agree, but at this time, this is the way it's decided.
/O
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Ast
Marks response was that he doesn't want to see any XML in manager. Period.
This upset me a bit, so in a childish way I simply closed the bug
reports...
I will discuss this with him at von next week, but if you have any
opinions pro and con XML in manager, please respond!
Remember, I am not sugg
Matthew Boehm wrote:
* Realtime SIP friends
--
The SIP realtime objects are users and peers that are loaded in memory
when needed, then deleted. This means that Asterisk currently can't handle
voicemail notification and NAT keepalives for these peers. Other than
that,
most of th
[EMAIL PROTECTED] wrote:
Hello,
I just started using asterisk, and have a question. I have setup two
asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1
Please do not use the -dev list for non-development questions.
THank you.
/O
___
From reading the chan_mgcp I understand a lot of the reload code is
disabled by #ifdefs. Does this mean that mgcp reload does not work or
that it actually works, but that we should remove that redundant code?
I haven't got any MGCP devices so I can't test...
/O
__
Brian West wrote:
Ya it should be up to date with latest CVS.. we have boxes with 7 weeks of
uptime that do 100's of reloads a day and it keeps up ;)
So I guess it's time to update the README...
/O
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Kevin P. Fleming wrote:
That patch also paves the way for a future upgrade to rwlocks for ASTOBJ
entities instead of mutexes; at this point it does not seem that that
will be possible soon, but I am still researching it. If it can be done,
it will be another big performance improvement, since ma
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 t
http://bugs.digium.com/bug_view_page.php?bug_id=0002859
A patch for chan_sip in cvs head. Please test and confirm your findings to the
bug tracker.
Thank you!
/Olle
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I've uploaded two patches to chan_sip to the bug tracker:
* sipgetheader()
* sipaddheader()
SIPgetheader takes any sip header from the incoming INVITE and adds it
to a variable. With this function, we can safely remove the patch that
was added a few days ago, where we read a header and store it in
Richard wrote:
Hi,
I have sip trustrpid enabled. If I make a call from sip phone to pstn,
the call pulls the right caller-id from “Remote-Party-ID” and sends out
to the carrier. It works perfectly.
I also use a SIP provider for long distance. When the sip call is sent
to the provider, * ignores
Fernando Romo wrote:
2) Make a project wher we make a "Central IAX Directory" (Skype Like)
where we deposit our "contact" channels and a kind of "signature" to try
to make a "trusted" Phone comunity. This derivate two efforts:
Check Dundi, in CVS head of Asterisk. http://www.dundi.com
Mark Spence
Chris Lee wrote:
Reinhard Max wrote:
On Tue, 19 Oct 2004 at 15:38, Chris Lee wrote:
so what I was suggesting was a peer-to-peer fabric that can help
with the membership, naming and call receipt issues so that IAX
could pick up some of the ease of use provided by the supernodes.
I think you are as
Christopher Jacob wrote:
Hey All,
I have kicked this around the -user list and the #asterisk channel and I
believe it to not be possible currently, so I thought I would ask about it's
feasibility on the -dev list.
It would be a useful feature if you could define a mailbox for a sip users
to a box o
http://bugs.digium.com/bug_view_page.php?bug_id=928
This patch adds a way to transfer a variable in the dial plan from the calling
leg of the call to the other end, to the channel created by dial(something).
There are several ways of doing this:
* Adding a _ in front of the variable makes
Kristopher Lalletti wrote:
Hi everyone,
I encountered an interesting/unordinary SIP proxy + nat-transversal
setup for a VOIP termination, here in Montreal, to which I’m trying to
get Asterisk registered as a SIP client without my temporary hack
(because this hack is affecting outbound calls).
Rich Adamson wrote:
It's a known fact that bugs are not being fixed in Stable, and even Mark
has suggested no one should be running Stable in a production environment.
On the other hand, there's not many bugs open in the bug tracker. Feature
requests and patches, but not bugs.
If you are aware of b
Duane wrote:
Olle E. Johansson wrote:
TLS. TLS is the standard for SIP UA-proxy encryption, and S/MIME is
used for
end-to-end auth and encryption.
IETF standard 802.1x uses EAP-TLS which is a slight variation of Cisco's
PEAP, but this is on the link layer not on the TCP layer...
Yes, bu
Dr. Rich Murphey wrote:
RFC 3711 doesn't specify the method for establishing the shared srtp session
key. Likewise, libsrtp appears to require that the two endpoints have
already established a shared key.
Does anyone know of any voip standards for key exchange?
I guess, only guess, that for SIP t
brian k. west wrote:
When you are ready for a test person let me know
I think that sipura added support for SRTP in the latest firmware.
Could be a good interoperability test platform.
/O
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John Todd wrote:
Encryption is a real concern of mine (and my customers.) SRTP is a
great tool, though we'd be well-advised to also have TLS for SIP, and
whole-enchliada-encryption for IAX2.
Two notes on the TLS/SSL topic:
* To get to TLS in SIP, we need TCP. Someone out there, you know who y
Ok, we need to figure out how to negotiate SRTP in SDP/SIP setup.
Jeremy, is H.323 able to use SRTP?
/O
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brian wrote:
WHOO lets give er a shot! :)
GREAT!!!
/O
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Björn Grönesjö wrote:
Greetings Asterisk developers,
I have reasons to believe that Asterisk's sip-stack is modifying the
"Contact:"-header in a response to a REGISTER message in an illegal way.
RFC 3261; 10.3 Page 66
8. The registrar returns a 200 (OK) response. The response MUST
[EMAIL PROTECTED] wrote:
+ context = strdupa(data);
Not found on FreeBSD...
Alloca is found though.
Strdup needs free();
Anyone that can come up with a strdupa function?
/O
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Olle E. Johansson, Ed
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