(timerId>0)
timeKillEvent(timerId);
and now I can sipxReinitialize at will :)
Best regards,
stipus
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ty fix, and
I haven't currently been able to figure out where the real problem is.
Best regards,
stipus
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n, it's at least the second time that DTMF is broken by
untested merges, and I can't afford spending 2 days of work each time a
change is commited.
Thanks,
stipus
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I have seen the problem using G711A and G711U.
I have to check the RTP payload size, but I'm not using any non standard
value, and didn't change anything on this side.
stipus
- Original Message -
From: "Alexander Chemeris" <[EMAIL PROTECTED]>
To: "stipus
call".
Any idea ? How could I solve this problem ?
Thanks in advance,
Stipus
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D] (it's a talking clock), I
get a 404 not found from the proxy.
Is there any way to make this work with Sipxtapi ?
I noticed that when X-Lite is configured with "Send outbound calls via
Target Domain", it just works.
Best regards,
stipus
- Original Message -
From: "Cha
Thanks a lot for the idea. Not setting a proxy at all seems to work !
stipus
- Original Message -
From: "Charlie Hedlin" <[EMAIL PROTECTED]>
To: "stipus" <[EMAIL PROTECTED]>
Cc:
Sent: Sunday, November 12, 2006 1:34 AM
Subject: Re: [sipxtapi-dev] X
This solution solves my problem 100%.
There is no need to spend some time analysing X-Lite SIP traces.
Thanks again !
- Original Message -
From: "Charlie Hedlin" <[EMAIL PROTECTED]>
To: "stipus" <[EMAIL PROTECTED]>
Cc:
Sent: Sunday, November 12, 2006 1:
dditional fixes.
Thanks for all bug tracking and patches. However, I don't think there is a
need for yet another branch specific to wxCommunicator.
Is there a good reason your fixes cannot be included into the main branch ?
TIA,
stipus
___
s
llId);
getRemoteAddress(&remoteAddress);
CpMultiStringMessage message(CpCallManager::CP_REFIRE_MEDIA_EVENT, callId,
remoteAddress, NULL, NULL, NULL,
MEDIA_PLAYBUFFER_STOP, MEDIA_CAUSE_NORMAL, mediaType) ;
mpCallManager->postMessage(message);
}
Best regards,
stipus
- Original
ld be glad to hear it ... because I'm really
puzzled !
Best regards,
stipus
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le, I switched to the sipxtapi branch.
stipus
- Original Message -
From: "Jaroslav Libak" <[EMAIL PROTECTED]>
To: "stipus" <[EMAIL PROTECTED]>
Cc:
Sent: Wednesday, June 13, 2007 10:58 AM
Subject: Re: [sipxtapi-dev] SipxCallPlayBufferStart and jerky outp
A few month ago, I wrote a mprDecodeInbandDtmf.cpp that works quite well.
I can try to post it here if you are interested.
stipus
- Original Message -
From: "Daniel Sigurgeirsson" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, June 13, 2007 6:33 PM
Subject: [sipxtapi-dev] Va
> > A few month ago, I wrote a mprDecodeInbandDtmf.cpp that works quite
well.
> >
> > I can try to post it here if you are interested.
>
> I believe we all would be interested. So, please, post it to issue tracker
too.
>
> I suppose, it detect DTMF, mixed with audio stream, right? Does it use
> som
/**< Incompatible destination -- We
were unable
to negotiate a codec */
MEDIA_CAUSE_DTMF_START,/**< A DTMF tone has started */
MEDIA_CAUSE_DTMF_STOP,/**< A DTMF tone has stopped */
MEDIA_CAUSE_DTMF_INBAND
} SIPX_MEDIA_CAUSE ;
I
evel of interop here from
.Net to C++...
If I have enough time, I'll try to create a mprSpeech.cpp and directly call
SAPI from the C++ side.
I may be able to make a quick test using mprFromFile ...
stipus
- Original Message -
From: "Daniel Sigurgeirsson" <[EMAIL PROTECTE
ave been removed
from latest builds.
Where would you start from if you had to do this ?
It would be really nice if someone could put me on track.
Thanks,
stipus
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bject();
SipxCallPlayBufferStart(m_handleIntPtr, bufferIntPtr, byteArray.Length, 0,
false, false, true);
- Original Message -
From: "Daniel Sigurgeirsson" <[EMAIL PROTECTED]>
To: "stipus" <[EMAIL PROTECTED]>;
Sent: Thursday, June 14, 2007 4:16 PM
Subject: RE: [sipxtapi-d
I compared the SIP traces between a call with normal behavior, and a call
crashing sipxtapi.
The difference I found on 2 traces is that I'm getting a RE-INVITE, and the
crash happens just after sipxtapi receives the RE-INVITE .
--> INVITE
<-- TRYING
<-- OK
--> ACK
--> INVITE *** crash ***
Here i
I'm also getting these messages in release mode, and my app crashes... I
have been trying to debug yesterday night, but couldn't find the problem.
First I thought it was a mismatch in msvc-8 projects, as the HAVE_SPEEX...
preprocessor directives are mismatched between projects (in some project
it'
16:17:18.108000Z":3831:KERNEL:CRIT:klnta:::sipXtapi:"OsM
sgPool::FindFreeMsg 'MediaSignals' queue size (64) exceeds hard limit
(64)\n"
stipus
- Original Message -
From: "Daniel Sigurgeirsson" <[EMAIL PROTECTED]>
To: "Alexander Chemeri
7;MediaSignals' queue size (64) exceeds hard limit
(64)\n"
and my app crashed again.
And I checked your other question: If I call from another phone number that
works, I don't get any soft limit exceeded message. But when I call from
this number, sipxtapi doesn't receiv
nexb=no\r\n
a=rtpmap:101 telephone-event/8000\r\n
a=fmtp:101 0-15\r\n
a=ptime:20\r\n
\r\n
--unique-boundary-1--END\n"
- Original Message -----
From: "stipus" <[EMAIL PROTECTED]>
To: "Daniel Sigurgeirsson" <[EMAIL
tapi:"OsMs
gPool::FindFreeMsg 'MediaSignals' queue size (64) exceeds hard limit (64)\n"
And crash.
- Original Message -
From: "Daniel Sigurgeirsson" <[EMAIL PROTECTED]>
To: "stipus" <[EMAIL PROTECTED]>; "Jaroslav Libak&
new MimeBodyPart(this, partStart, partLength);
And I got partStart by changing getMultipartBytes() to return this value as
an additional output parameter.
I'll try to publish a clean patch ASAP... but now it's working, and sipxtapi
doesn't crash anymore when receiving multipart/mixed bodies in IN
Here I have 2 VS2005 solutions open
- One for SIPXTAPI
- One for my .NET project (using SIPXTAPI).
1) Make sure you are compiling SIPXTAPI with DEBUG turned on.
2) Copy SIPXTAPID.DLL and SIPXTAPID.MAP to your .NET project BIN directory.
3) Run your .NET project. Make sure it loads the Debug DLL
For this problem, I have (manually) applied the patch XMR-70
http://track.sipfoundry.org/browse/XMR-70
There is a ZIP file with a Non Windows Message Queue. This NWMQ is then used
in place of the windows message queue in the speaker thread. This solved the
problem for me.
stipus
- Original
I haven't changed thread priorities I have just installed the Non
Windows Message Queue.
stipus
- Original Message -
From: "Alexander Chemeris" <[EMAIL PROTECTED]>
To: "stipus" <[EMAIL PROTECTED]>
Cc: "Alexander Boreham" <[EMAIL
tiple files (.h and .cpp) ...
Hemmm sorry :(
stipus
- Original Message -
From: "Alexander Chemeris" <[EMAIL PROTECTED]>
To: "stipus" <[EMAIL PROTECTED]>
Cc: "Daniel Sigurgeirsson" <[EMAIL PROTECTED]>; "Jaroslav Libak"
<[EM
Mixers in the flowgraph create this problem... ?
stipus
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I still couldn't figure how to select several files for a patch ... so I
created 2 patches ...
One for HttpBody.h
One for HttpBody.cpp
Best regards,
stipus
- Original Message -
From: "Alexander Chemeris" <[EMAIL PROTECTED]>
To: "stipus" <[EMAIL PRO
Thanks !
Here is a single patch for both files :)
stipus
- Original Message -
From: "Daniel Sigurgeirsson" <[EMAIL PROTECTED]>
To: "stipus" <[EMAIL PROTECTED]>; "Alexander Chemeris"
<[EMAIL PROTECTED]>
Cc: "Jaroslav Libak" <
tand
what's happening...
stipus
- Original Message -
From: "Keith Kyzivat" <[EMAIL PROTECTED]>
To: "stipus" <[EMAIL PROTECTED]>
Cc:
Sent: Tuesday, July 24, 2007 6:21 PM
Subject: Re: [sipxtapi-dev] Bug in mpCallFlowGraph with DISABLE
OK, I just found the audio glitches problem...
I disabled LINEAR_COMPLEXITY_BRIDGE in MprBridge.h --> No more audio
glitches !
I fear there must be another problem in the new bridge implementation !!!
stipus
- Original Message -
From: "stipus" <[EMAIL PROTECTED]>
working very well with one connection per call.
stipus
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>>The changes made in revision 9386 have caused STUN failures on the RTP
>and
>>RTCP ports. The SIP call control port still receives STUN responses
>>correctly. I've made many Wireshark traces and can see that the STUN
>>requests are correctly made to the server and that the correct STUN
>>response
to get MEDIA_PLAYBUFFER_START/STOP and
MEDIA_PLAYFILE_START/STOP notifications from MprFromFile as it was working
in the sipxtapi-media-update branch...
Regards,
stipus
- Original Message -
From: "Alexander Chemeris" <[EMAIL PROTECTED]>
To: "stipus" <[EMAIL PR
> Not sure if this is going to help you. But I did find a
> problem in SdpBody.cpp in SdpBody::addCodecsOffer(). Basically
> 'szTransportString' was being used to fill transport in SDP but is being
used
> unintialized. I fixed the problem by using 'UDP' directly instead of that
> variable. Thats th
ched patch.
Thanks a lot for this patch !
stipus
- Original Message -
From: "Jaroslav Libak" <[EMAIL PROTECTED]>
To: "Anshuman S. Rawat" <[EMAIL PROTECTED]>
Cc: "stipus" <[EMAIL PROTECTED]>;
Sent: Thursday, July 26, 2007 5:17 PM
Subject: Re:
s NWMQ for monthes in my app, and I didn't
find any problem (yet).
stipus
- Original Message -
From: "Alexander Chemeris" <[EMAIL PROTECTED]>
To: "Alexander Boreham" <[EMAIL PROTECTED]>
Cc:
Sent: Friday, July 27, 2007 7:44 PM
Subject:
Here are the files I use for the NonWin32Queue.
The .cpp is in sipXmediaLib\src\nw32
The .h is in sipXmediaLib\include\nw32
stipus
- Original Message -
From: "Jaroslav Libak" <[EMAIL PROTECTED]>
To: "Alexander Boreham" <[EMAIL PROTECTED]>
Cc:
Sent:
od (again with a pMediaEventListener in the flowgraph, and new
OnPlayBufferStart/Stop methods in SipConnection.)
I'm very interested in implementing this "the right way", and if possible
with the new notification system.
stipus
- Original Message -
From: "Alexander C
tion
object as MpdPtAvt).
I'm now going to see how I can add a second Notification (dedicated to
INBAND dtmf)... but I fear this would require many changes.
stipus
- Original Message -
From: "stipus" <[EMAIL PROTECTED]>
To: "Alexander Chemeris" <[EMAIL PR
It seems the notification signal uses 31 of the 32 bits
first bit is (keyup/keydown)
second bit seems unused
Then there are 14 bits for the DTMF key
Then there are 16 bits for the detected tone duration.
Maybe I could use the second bit set to 1 for INBAND DTMF ?
What do you think ?
stipus
EDIA_CAUSE =
MEDIA_CAUSE_DTMF_INBAND
When I get a MEDIA EVENT= MEDIA_REMOTE_DTMF, I look at the MEDIA_CAUSE
It can be one of
MEDIA_CAUSE_DTMF_START, /**< A RFC2833 DTMF tone has started */
MEDIA_CAUSE_DTMF_STOP, /**< A RFC2833 DTMF tone has stopped */
MEDIA_CAUSE_DTMF_INBAND // INBAND DTMF detected
I agree with you, but when I coded InBand DTMF, I didn't want to break
existing code.
stipus
- Original Message -
From: "Jaroslav Libak" <[EMAIL PROTECTED]>
To: "stipus" <[EMAIL PROTECTED]>
Cc:
Sent: Tuesday, July 31, 2007 10:35 PM
Subject: Re:
ay:
ToneGen-->FromStream-->FromFile-->ToneFileSplitter--> etc
If FromFile works, there is no reason ToneGen does not. There must be
another bug somewhere...
stipus
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> > In the flowGraph resources are linked this way:
> >
> > ToneGen-->FromStream-->FromFile-->ToneFileSplitter--> etc
> >
> > If FromFile works, there is no reason ToneGen does not. There must be
> > another bug somewhere...
I had a better look at this, and DTMF tones only work after codecs are
in
I have been trying to stress test my application using Sipp today.
http://sipp.sourceforge.net/index.html
The scenario is very simple:
|(1) INVITE |
|-->|
|(2) 100 (optional) |
|<--|
|(3) 180 (optional) |
|<--|
|(
nd you have to make slight changes to
sipxCallCreate() if you really want virtual lines.
stipus
- Original Message -
From: "Amith Nambiar" <[EMAIL PROTECTED]>
To: "David Sargrad" <[EMAIL PROTECTED]>
Cc:
Sent: Tuesday, August 07, 2007 5:49 PM
Subject: Re: [si
A preliminary patch has been published a week ago for the DNS issue, but
it's not commited to svn yet.
You can just get the patch from the list archive, and apply it.
stipus
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Monday, August 06, 2007 7:34 PM
Subject: Re
When you get a new call event, you get the line id ... as you know the line
identity you registered for this id, you should know which phone number has
been called.
stipus
- Original Message -
From: "Daniel Sigurgeirsson" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, Augus
MAL, mediaType
- MEDIA_PLAYBUFFER_START, MEDIA_CAUSE_NORMAL, mediaType
- MEDIA_PLAYBUFFER_STOP, MEDIA_CAUSE_NORMAL, mediaType
How can the resource notification message result in a sipxtapi event being
fired ?
Thanks in advance,
stipus
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I'm also very interested in early media support.
However, I fear it won't be as simple as a config change.
stipus
- Original Message -
From: "Anthony G" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, October 03, 2007 11:12 PM
Subject: [sipxtapi-dev] sipXPhone -
opinion ?
Thanks,
Stipus
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You need to install the VS2005 SP1 redistribution package on the deployment
computer.
You can easily download this from the MS download site.
This installs new versions of the side by side MSVC runtime libs.
Best regards,
stipus
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[
7;t forget to change this parameter on all sipxtapi projects (sipxtapi,
portlib, calllib, stacklib, medialib...)
Regards,
stipus
-Message d'origine-
De : stipus [mailto:[EMAIL PROTECTED]
Envoyé : mercredi 27 août 2008 09:19
À : 'sipxtapi-dev@list.sipfoundry.org'
Objet : RE: [sip
Or use the Sipxtapi function:
sipxConfigSetAudioCodecByName( sipxInstance, « PCMA » ) ;
rgds,
stipus
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Paulo
Vicentini
Envoyé : jeudi 9 octobre 2008 18:34
À : sipxtapi-dev@list.sipfoundry.org
Objet : Re: [sipxtapi-dev
it's clearly stated in their policy:
http://www.digium.com/en/docs/G729/g729policy.php
" The indemnification for patents and copyright infringement of the G.729
standard covers Digium's products only."
If you find any other way, I would be glad to hear from you.
stipus
You can use Wireshark to capture all IP packets, and then decode & play
received and sent RTP (if its in one of the supported formats only G711
as far as I know).
You can access the RTP replay feature from the wireshark statistics / VOIP
calls menu.
stipus
De : [EMAIL PROTE
As long as you didnt use Wireshark to capture and replay the RTP, you cant
be sure that this RTP doesnt contain silence
.
Its a 10 minute test (time to download and install wireshark and capture a
few packets), and then you can be sure if its a problem within sipxtapi or
not.
stipus
.
Regards,
stipus
De : sipxtapi-dev-boun...@list.sipfoundry.org
[mailto:sipxtapi-dev-boun...@list.sipfoundry.org] De la part de Daniel
Sigurgeirsson
Envoyé : mercredi 4 novembre 2009 00:07
À : sipxtapi-dev@list.sipfoundry.org
Objet : [sipxtapi-dev] SIP Trunk
Hi all,
could anyone explain in
I would like to know if someone ever worked on a SAPI5 voice recognition
resource for the Call FlowGraph ?
Regards,
Stipus
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