>
>
>> As to the testing, I'm totally all for it. In every testing
>> scenario, as
>> we push the driver to its limits, we can profile chan_sip to see
>> where
>> it is spending most of its cpu cycles, and see if any significant
>> improvements can be made...
>
> +1. Again, I would rather see
16 feb 2008 kl. 02.36 skrev Terry Wilson:
>> Maybe we can start creating a collection of SIPP tests to run various
>> scenarious.
>> I would like to test how registrations and subscriptions affect the
>> stack too.
>
> I have written a tool in perl that allows you to take a SIP pcap
> capture (li
14 feb 2008 kl. 22.14 skrev SVN commits to the Digium repositories:
> Author: russell
> Date: Thu Feb 14 15:14:37 2008
> New Revision: 103692
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=103692
> Log:
> Add a pretty hefty document on setting up and testing distributed
> events, and
14 feb 2008 kl. 21.25 skrev SVN commits to the Digium repositories:
> Author: murf
> Date: Thu Feb 14 14:25:11 2008
> New Revision: 103686
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=103686
> Log:
> Adding some documentation for testing sip channel performance using
> sipp.
Murf!
31 jan 2008 kl. 15.43 skrev Steve Davies:
> On Jan 31, 2008 10:39 AM, Johansson Olle E <[EMAIL PROTECTED]> wrote:
>> Late answer, found in my mail queue. Obviously did not get sent on
>> the
>> airport...
>>
>> /O
>>
>> 23 jan 2008 kl. 21.0
22 jan 2008 kl. 18.50 skrev Kevin P. Fleming:
> Sebastian Damm wrote:
>
>> Asterisk gets a call from PSTN (via Sangoma PRI card). It sends out
>> an
>> INVITE to the appropriate SIP peer. The SIP peer answers with 180
>> without SDP, but still sends some RTP packets (silent). I don't
>> know
22 jan 2008 kl. 17.37 skrev Atis Lezdins:
> On 1/22/08, Johansson Olle E <[EMAIL PROTECTED]> wrote:
>>
>> 22 jan 2008 kl. 15.32 skrev Atis Lezdins:
>>
>>> On 1/20/08, Johansson Olle E <[EMAIL PROTECTED]> wrote:
>>>> This also touches my
22 jan 2008 kl. 15.32 skrev Atis Lezdins:
> On 1/20/08, Johansson Olle E <[EMAIL PROTECTED]> wrote:
>> This also touches my earlier questions about what realtime SIP
>> buddies
>> really are.
>> Today, we have an unsatisfactory mix between the old sip bud
16 jan 2008 kl. 15.56 skrev Atis Lezdins:
> On 1/16/08, Leif Madsen <[EMAIL PROTECTED]> wrote:
>>>
>>> I agree for real devices. However i wonder - why i can't change
>>> state
>>> for Local channels.
>>>
>>>
>> Funny enough, I had this same issue today within Queue(). I'm using
>> queue_member
13 jan 2008 kl. 23.09 skrev Robert Moskowitz:
> I have discovered a challenge with Asterisk running with a public IP
> address and thus not needing NAT, when working with a service that of
> course knows there has to be a NAT there
>
> I come here to the developer's list, as this requires som
Steve,
This is work that has been on my wishlist for a very long time. Thank
you!
I hope to get time to test drive this soon and have some ideas on how
to do it.
We need to discuss the dialog matching to se fix it once and for all,
there's a lot
of open issues I have there on my list from
2 jan 2008 kl. 19.57 skrev Raj Jain:
> Now it makes sense. I agree that you do need to extend SIP one way
> or another to acheive the correct key telephone system
> functionality, because RFC 4235 support alone is not sufficient for
> many use cases.
>
> I personally think this is very valu
2 jan 2008 kl. 15.58 skrev Raj Jain:
> Gene,
>
> I'm trying to better understand what you've developed. It seems that
> you're trying to share Asterisk SLA trunks w/ a different system
> (Broadsoft). Is the following a correct representation of your
> architecture?
>
>
> | SLA Trunks
>
27 dec 2007 kl. 19.54 skrev Kamanashis Roy Shuva:
> Hi,
>
> The patch is not successful adding this feature. I think there is more
> to be added. Or the patch should follow a good design. I mean I am not
> satisfied with this patch anyway.
>
> Here I have found the flash support for sip.
>
> http
27 dec 2007 kl. 19.54 skrev Kamanashis Roy Shuva:
> Hi,
>
> The patch is not successful adding this feature. I think there is more
> to be added. Or the patch should follow a good design. I mean I am not
> satisfied with this patch anyway.
>
> Here I have found the flash support for sip.
>
> http
27 dec 2007 kl. 12.39 skrev Kamanashis Roy Shuva:
> Hi,
>
>I wanted to send flash using sip info dtmf .. it says "Don't know
> how to indicate condition 9"!!. it seems that sip dtmf supports flash
> . So why not using 'f' character as a part of stream for flashing . I
> think a little change
7 dec 2007 kl. 17.45 skrev Igor A. Goncharovsky:
> Hi!
>
> Johansson Olle E wrote:
>> Incoming calls are handled this way
>>
>> * First, we match on peer object name with the From username
>> * Then we try to match on IP/Port
>> * If we can't match, we
7 dec 2007 kl. 09.08 skrev Steve Langstaff:
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of
>> Johansson Olle E
>
>> I've created a branch called "kill_the_user" that has no
>> type=friend or type=user, only peers.
>>
&g
6 dec 2007 kl. 15.29 skrev Victor Toofic:
> El Thu, Dec 06 de 2007 a las 09:53 +0000, Johansson Olle E comentaba:
>> What does it mean? Any documentation anywhere on how to implement it?
>> Is the previous duration of 2600 for signal 1 cut off to 305?
>
> The only thing
As agreed with russellb, I've created a new branch for implementing
two things:
- Caller ID names in utf8 for SIP, IAX2 and jingle
- Caller ID domains for SIP and Jingle
The domains will be used for SIP2SIP calls primarily, as domain is
part of the
address and should not be replaced in the ou
Friends,
Philippe Sultan, one of the active developers in the area of our
Jabber/XMPP integration, was recently elected a member of the XMPP
Standards foundation.
This is a very good thing for the Asterisk project as we move forward
with integration between the Asterisk call states and XMPP
While browsing the bug tracker today, I found a patch for adding more
"concise" commands to the SIP channel.
My personal opinion is that I don't like adapting the CLI for machine
parsing. If we're about to do that, we might
as well convert all CLI listings in one big janitor project. But we
Russell,
Changing a released product?
Hmmm. That's a new policy... :-)
/O
Vidarebefordrat brev:
> Från: SVN commits to the Digium repositories [EMAIL PROTECTED]>
> Datum: måndag 15 okt 2007 18.59.54 GMT+02:00
> Till: [EMAIL PROTECTED], [EMAIL PROTECTED]
> Ämne: [svn-commits] russell: trunk r85
2 jan 2007 kl. 14.55 skrev Kevin P. Fleming:
[EMAIL PROTECTED] wrote:
Author: oej
Date: Tue Jan 2 07:50:51 2007
New Revision: 49152
URL: http://svn.digium.com/view/asterisk?view=rev&rev=49152
Log:
Update sample config
Modified:
trunk/ (props changed)
trunk/configs/features.conf.sa
2 jan 2007 kl. 12.32 skrev Anton:
This is one of the ways, thanks for clue! but why not to add
this possibility to a mainstream, that appropriate
X-headers would be setup and read by the asterisk itself.
For example, a SIP channel may register two prefixes, like
SIP - for current dial scheme, a
/DestAsterisk/[EMAIL PROTECTED])
In big system being able to chose a context is very tasty
thing, making dialplanning easier in many aspects.
Regards,
Anton.
On 29 December 2006 15:50, Johansson Olle E wrote:
29 dec 2006 kl. 07.17 skrev Anton:
Guys,
Have anyone thought about implementing a k
Well, why don't you just not switch to SIP?
(he, he. Couldn't resist)
/O ;-)
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Well, as Kevin said, this is only software.
There is a way to do this, that I presented at Astricon a year ago...
We could create virtual peers that we register in the ${HINT} for the
extension. Dial the HINT
and you will dial all registered peers. One peer per registration
makes Asterirsk h
31 okt 2006 kl. 10.06 skrev Anu Gupta:
May I know the bug resolution process and efficieny of the project
Visit our bug tracker, http://bugs.digium.com where you will find
documentation
of the process and can judge yourself on how we're handling it. We do
need
more help - remind yourself
27 okt 2006 kl. 12.55 skrev Raffaele Porzio:
Hi everyone, I'm working with Asterisk 1.4-beta3 and ARA for my
Univesity thesis, to enable jingle support into an administrative
framework for asterisk developed in our lab. It's possible to map
jabber's and gtalk's user from the ARA database,
inbound or outbound registrations?
The Contact is certainly saved, and that's the interesting part.
Why should the From header be interesting?
/O
27 okt 2006 kl. 07.58 skrev Luigi Rizzo:
Any interest in adding the username from the REGISTER message
in the stored peer state ? It seems that we
26 okt 2006 kl. 08.00 skrev Luigi Rizzo:
On Wed, Oct 25, 2006 at 07:16:11PM -, asterisk-
[EMAIL PROTECTED] wrote:
Author: oej
Date: Wed Oct 25 14:16:10 2006
New Revision: 46252
URL: http://svn.digium.com/view/asterisk?rev=46252&view=rev
Log:
Somewhat ugly code to try to fix issue #7608.
S
17 okt 2006 kl. 14.13 skrev Roy Sigurd Karlsbakk:
yes - using the IP address in a sip URI is possible, but your
Provider [asterisk]
uses the IP Adress of the SwyxServer in the uri ~INVITE
sip:[EMAIL PROTECTED]:65002 SIP/2.0
thats the wrong part - he must use a) his realm or b) his public IP
17 okt 2006 kl. 13.26 skrev Roy Sigurd Karlsbakk:
when dialing a number like this 00123456789:
asterisk make this,
SIP:[EMAIL PROTECTED]:5070
but what i need is,
SIP:[EMAIL PROTECTED]
asterisk put the port on the SIP address, any idea on how to
remove the
port from the sip address..
A
17 okt 2006 kl. 13.25 skrev Roy Sigurd Karlsbakk:
trying to connect asterisk to a Swyx (dot com) server, I cannot
send the call through to the swyx, always getting a 404. The swyx
people tell me:
These call is rejected on swyxware side, because the provider
uses a non
rfc compatible kin
Please try using fromdomain= and see what happens. Thanks.
/Olle
17 okt 2006 kl. 12.13 skrev Alexandre Almeida:
I have a peer with this configuration, it's used as trunk for outgoing
calls.
[xxyy]
username=123456
type=peer
secret=1423241
port=5070
outboundproxy=my.proxy.com
host=gateway.ser
16 okt 2006 kl. 15.11 skrev Brian Candler:
On Mon, Oct 16, 2006 at 01:28:13PM +0200, Johansson Olle E wrote:
I've tested it now, and in my SVN build, which is about 10 days out
of date,
it appears to be broken.
Full description posted at http://bugs.digium.com/view.php?id=8152
In su
16 okt 2006 kl. 15.06 skrev Tzafrir Cohen:
On Mon, Oct 16, 2006 at 06:43:53AM -0500, Rich Adamson wrote:
What I'm worried about is trying to fix it in the current code,
since it
will change quite a few things that are needed today,
and break backwards compatibility. I've tried, but failed,
16 okt 2006 kl. 13.52 skrev Morten Isaksen:
On 10/16/06, Johansson Olle E <[EMAIL PROTECTED]> wrote:
13 okt 2006 kl. 18.09 skrev Kevin P. Fleming:
BUT, that's exactly what we're doing in Asterisk 1.4 - all IFs and
BUTs regarded, if there's only two SIP endpoints in the
16 okt 2006 kl. 13.26 skrev Brian Candler:
On Mon, Oct 16, 2006 at 11:51:42AM +0200, Johansson Olle E wrote:
Asterisk really does look and smell like a SIP proxy in this case.
No. We never ever forward SIP messages. As you see, this involves two
different
SIP calls, with separate Call-IDs
22 sep 2006 kl. 09.45 skrev Alexandr Olekhnovich:
I'm sorry for that question, but I really do not know if this trouble
was solved.
I have to make Presence support in Asterisk but the next choise is
not universal
exten => num1, hint, uid1
exten => num2, hint, uid2
...
I must write down
12 sep 2006 kl. 15.23 skrev <[EMAIL PROTECTED]> [EMAIL PROTECTED]>:
Hello,
I use asterisk svn-trunk .
I wish asterisk to forward all sip requests from non
local domains to a proxy .
For example asterisk handle domainA a sip agent send a
invite to a domainB .
Is asterisk able to check the d
30 aug 2006 kl. 17.37 skrev Kevin P. Fleming:
- Andrea Spadaccini <[EMAIL PROTECTED]> wrote:
So without a disclaimer my code is likely to be ignored? :(
Not likely, guaranteed. We do not look at undisclaimed code for
copyright and other licensing reasons, and in the near future you
w
30 aug 2006 kl. 17.10 skrev John Lange:
On Wed, 2006-08-30 at 09:38 -0400, Jared Smith wrote:
(I know what you're thinking -- I'd like to be able to use regex
matches too -- but that's probably overkill at this stage.)
Its inevitable that Asterisk will need regex matching at some
point.
18 aug 2006 kl. 15.11 skrev Wasim Baig:
http://www.ietf.org/internet-drafts/draft-guy-iax-01.txt
expires on sep 7, 2006
what happens then? or is the expiry date superflous?
moreover what needs to be done to make IAX more than a draft,
i.e. a real honest to goodness, meaty RFC
Ed Guy can ans
4 jul 2006 kl. 11.23 skrev Mailing Lists:
Hi,
Would it be a lot of work to change app_queue so that SIP agents
that is
busy and have callwaiting enabled will be included in the ringall list
for next call in queue? I have looked in the source code but it's not
obvious for me where this lis
27 jun 2006 kl. 04.22 skrev Paul Cadach:
Hello,
Johansson Olle E wrote:
Are you sure? All channels that supply 'ast_rtp_bridge' as their
bridge method should already support it.
That was news to me. Cool if that's the case!
Never seen or heard about it before.
I stand corr
21 jun 2006 kl. 18.02 skrev md:
Hello,
if you use redirect manager action while the channel is hanging up
Asterisk crash and the last log is "Putting channel ..." in channel.c.
Is there a solution for this?
´
If you find a bug, always visit the bug tracker to see if someone
else have fou
6 jun 2006 kl. 12.05 skrev Kev Jackson:
Hi all,
I've just done a very very simple change to the code as suggested
by the janitor projects list[1], and I'm unsure where to upload the
code to. I have experience in open source with apache/jakarta
(though no deep experience of C coding, so
1 jun 2006 kl. 16.51 skrev Mike Fedyk:
Olle E Johansson wrote:
Friends,
I finally committed the last piece of the new SIP transfer support
code. This greatly enhances the support of SIP
transfers - or at least is meaning to. The code has been tested on
1.2 for almost a year in production,
17 maj 2006 kl. 10.06 skrev <[EMAIL PROTECTED]>
<[EMAIL PROTECTED]>:
Hi all,
The current PLC algorithm uses only one pitch period, and
therefore behaves bad during burst lost (Beeps in background),
I have prolonged the pitch period under burst loss situation and
tested it using a small
17 maj 2006 kl. 08.41 skrev Wayne Shaw:
Hi,
Sorry this may be off topic.
Where do I start if I want to build a hard SIP IP phone?
You're right, this is not a question related to Asterisk development.
I would recommend
the SIP implementors mailing list.
/Olle
___
This looks interesting. What about patents and licenses?
/Olle
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31 mar 2006 kl. 02.39 skrev [EMAIL PROTECTED]:
Hello,
Asterisk will not use the tag that was generated by remote site for
the
To: field when sending CANCEL message.
Relevant line of chan_sip.c:
/* Add tag *unless* this is a CANCEL, in which case we need
to send it exactly
29 mar 2006 kl. 01.20 skrev John Todd:
OK, so I've put forward the solution. Someone other than me
should test it. I'd like to get this approved and in SVN TRUNK
before the next freeze so it can be part of the distribution.
Please take a few moments away from Olle's gargantuan list of
28 mar 2006 kl. 15.02 skrev Pavel Jezek:
Hi Olle,
what about incorporating this two patches to your testbranch?
I think two of the most wanted in new asterisk release ;-)
thanks for your great work! :-)
PJ
0004825: [patch] [post 1.2] New codec negotiation algorithm
0006643: [patch] Implement
OK, so I've put forward the solution. Someone other than me should
test it. I'd like to get this approved and in SVN TRUNK before the
next freeze so it can be part of the distribution. Please take a
few moments away from Olle's gargantuan list of test cases and poke
at this for a bit t
13 mar 2006 kl. 16.27 skrev anthony thomas:
Hello all,
We are testing the t38passthrough branch in our
gateways and can not fully understand why the RTP
stream has to go througth the * box disabling native
RTP bridging.
What is the problem in allowing a second RE-INVITE to
switch to T.38?
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