Same problem here using Ubuntu Hardy and Wednesday's trunk on three separate 
computers. Incoming calls segfault every time on each 3.1.1 build. Outgoing 
calls are ok. [email protected] echo test works but all incoming calls segfault.

Craig Albrecht
Taylor University Broadcasting Inc
Indiana, USA

-----Original Message-----
From: [email protected] [mailto:[email protected]] On 
Behalf Of [email protected]
Sent: Friday, February 27, 2009 10:58 AM
To: [email protected]
Subject: ekiga-list Digest, Vol 31, Issue 59

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Today's Topics:

   1. Re: direct IP to IP SIP call without SIP registrar,       not
      working in 3.x (Jim Diamond)
   2. Re: direct IP to IP SIP call without SIP registrar,       not
      working in 3.x (yannick)
   3. Re: PTLIB alsa plugin status (yannick)
   4. Re: direct IP to IP SIP call without SIP registrar,       not
      working in 3.x (Damien Sandras)
   5. Re: [ekiga engine] api availability and documentation
      (Damien Sandras)
   6. Re: call failure (Damien Sandras)
   7. Re: WG: Connect to a LifeSize HD System (Damien Sandras)
   8. Re: Incoming call failure with 3.1.1 (Damien Sandras)


----------------------------------------------------------------------

Message: 1
Date: Fri, 27 Feb 2009 08:56:27 -0400
From: Jim Diamond <[email protected]>
Subject: Re: [Ekiga-list] direct IP to IP SIP call without SIP
        registrar,      not working in 3.x
To: Ekiga mailing list <[email protected]>
Message-ID: <[email protected]>
Content-Type: text/plain; charset=us-ascii

On Fri, Feb 27, 2009 at 07:37 (-0000), Dave Higton wrote:

> Patrick Lee wrote:

>> Not sure any one has asked this question before.

>> I was able to make call without registering any SIP registrar in Ekiga
>> ver 2.x.  No need to create SIP account.
>> Just type "sip:192.168.1.34"  (the remote IP address) and the
>> call was setup.
>> And the remote client could be x-lite.

>> But NOT with ver 3.x, the same syntax gave me no response.  It seems
>> like I have to create a SIP account first.

>> My environment is a complete isolated network so I simply can't
>> register in ekiga.net for SIP account.
>> And there is some other SIP clients so I have to stick with
>> SIP protocol.

>> I know SIP registration is a proper SIP operation.  But I love ekiga
>> as it allows pt-to-pt call without setting up a SIP server.  But not
>> true for ver 3.x.  Any solution ?

> It may be worth mentioning that I have the same situation (no SIP
> access to the outside world, therefore Ekiga cannot possibly
> register to anything), the same requirement to do VoIP across our
> own LAN (only), and the same problem.

Dave,

Have you tried connecting to
        h323:192.168.1.34
(or IP as appropriate)?  It works for me both on a LAN and over a wan
with 3.0.[12] and 3.1.0.  Regrettably, I have other show-stopper
problems, but that's another thread.

Patrick, you say you have other SIP clients, but will the h323: thing
work for you in your ekiga-to-ekiga work?

                                Jim


------------------------------

Message: 2
Date: Fri, 27 Feb 2009 15:57:19 +0100
From: yannick <[email protected]>
Subject: Re: [Ekiga-list] direct IP to IP SIP call without SIP
        registrar,      not working in 3.x
To: Ekiga mailing list <[email protected]>
Message-ID: <1235746639.15492.22.ca...@achille>
Content-Type: text/plain; charset=utf-8

Le vendredi 27 f?vrier 2009 ? 08:56 -0400, Jim Diamond a ?crit :
> >> I know SIP registration is a proper SIP operation.  But I love
> ekiga
> >> as it allows pt-to-pt call without setting up a SIP server.  But
> not
> >> true for ver 3.x.  Any solution ?

I reshaped our wiki page about NATs,

did you read this troubleshooting?

http://wiki.ekiga.org/index.php/Ekiga_behind_a_NAT_router#I_have_2_Ekiga_behind_a_NAT_using_STUN:_they_can.27t_communicate

And this configuration setup if you're sure to only use Ekiga
exclusively inside the LAN:

http://wiki.ekiga.org/index.php/Ekiga_behind_a_NAT_router#How_to_disable_STUN_with_Ekiga_3.3F

@Damien,
What is the command line to enable STUN back using Ekiga 3.x?



------------------------------

Message: 3
Date: Fri, 27 Feb 2009 16:02:56 +0100
From: yannick <[email protected]>
Subject: Re: [Ekiga-list] PTLIB alsa plugin status
To: Ekiga mailing list <[email protected]>
Message-ID: <1235746976.15492.26.ca...@achille>
Content-Type: text/plain; charset=utf-8

Le vendredi 27 f?vrier 2009 ? 09:37 +0100, Alec Leamas a ?crit :
> Derek Smithies wrote:
> > Hi,
> >
> > On Fri, 27 Feb 2009, Alec Leamas wrote:
> >
> >> Hm... a write operation could be guaranteed to return in finite time
> >> (using non-blocking io + snd_pcm_wait). So couldn't the close method
> >> just mark the chanell as closing, leaving the dirty work to the
> >> "writer" thread and thus avoiding the locks? (Which, otoh, really
> >> isn't a big issue in this scenario). If required, opening could be
> >> handled in the same way, I guess. This would also create the
> >> advantage that the thread could process the jitter buffer data in
> >> parallel with the alsa output, without the need to wait for the IO to
> >> complete. Wouldn't this give a more accurate timing? Also, avoiding
> >> blocking io is a Good Thing IMHO.
> >
> > No.
> > It must be a blocking write. The architecture of opal demands this.
> >
> > The play thread (using play as an example) repeatedly does the following
> >   read rtp packet from jitter buffer
> >   decode
> >   put raw audio to sound device (which delays for up to framesize of
> >                           packet)
> >
> I didn't really make my point clear, sorry. I understand that the write
> method should block, and  my code does this, it's just a question how
> it's implemented. Refactored to a write method:
>
> write( pcm,  chunk)
>     if( closing)
>         close(); return();
>
>     snd_pcm_wait( pcm, timeout)
>         // Blocks until there is a free frame in alsa buffer,
>         // the same time as a blocking write would, using the
>         // the same "timer", but with a timeout option.
>
>     if( timeout)
>         // Underrun? Check status & handle error.
>     else
>         write( pcm, chunk) // non-blocking
>
>
> The basic difference is that this code will never block indefinitely -
> thus making it it possible to remove the locks. Depending on the
> blocking, alsa write implementation it might also give a slightly better
> timing. But I shouldn't count on it.
>

I've no clue if this documentation might help, still the pulse audio
main author refers it as "a guide":
http://0pointer.de/blog/projects/guide-to-sound-apis.html

Especially the section "You want to know more about the safe ALSA
subset?"


> >
> > There was a time when pwlib and openh323 (the old names of ptlib and
> > opal)
> > used non blocking writes to the sound card plus software timers. the
> > software timers were found to not be reliable enough to delay the
> > write thread. Sometimes the delay was hundreds of milliseconds. So
> > openh323 and pwlib were changed to use blocking writes, which gave
> > much better audio performance.
> >
> > to change the operation of the write to be non blocking would have
> > major architectural implications to opal. Let me help you. This won't
> > happen.
> Agreed
>
> Coming back to the other issues. Unfortunately, I'm the victim of
> https://bugzilla.redhat.com/show_bug.cgi?id=481722, having a hard time
> to to test Ekiga. I'll do what I can, though.
>
> _______________________________________________
> ekiga-list mailing list
> [email protected]
> http://mail.gnome.org/mailman/listinfo/ekiga-list
>
--
Me joindre en t?l?phonie IP / vid?oconf?rence ?
sip:[email protected]
Logiciel de VoIP Ekiga : http://www.ekiga.org
http://wiki.ekiga.org/index.php/Which_programs_work_with_Ekiga_%3F



------------------------------

Message: 4
Date: Fri, 27 Feb 2009 16:24:03 +0100
From: Damien Sandras <[email protected]>
Subject: Re: [Ekiga-list] direct IP to IP SIP call without SIP
        registrar,      not working in 3.x
To: Ekiga mailing list <[email protected]>
Message-ID: <[email protected]>
Content-Type: text/plain; charset=UTF-8

Le vendredi 27 f?vrier 2009 ? 15:57 +0100, yannick a ?crit :
> Le vendredi 27 f?vrier 2009 ? 08:56 -0400, Jim Diamond a ?crit :
> > >> I know SIP registration is a proper SIP operation.  But I love
> > ekiga
> > >> as it allows pt-to-pt call without setting up a SIP server.  But
> > not
> > >> true for ver 3.x.  Any solution ?
>
> I reshaped our wiki page about NATs,
>
> did you read this troubleshooting?
>
> http://wiki.ekiga.org/index.php/Ekiga_behind_a_NAT_router#I_have_2_Ekiga_behind_a_NAT_using_STUN:_they_can.27t_communicate
>
> And this configuration setup if you're sure to only use Ekiga
> exclusively inside the LAN:
>
> http://wiki.ekiga.org/index.php/Ekiga_behind_a_NAT_router#How_to_disable_STUN_with_Ekiga_3.3F
>
> @Damien,
> What is the command line to enable STUN back using Ekiga 3.x?

gconftool-2 -s /apps/ekiga/general/nat/stun_server stun.ekiga.net
--type=string

This is only needed in 3.0 because in 3.20, there is a new option in the
preferences window.
--
 _     Damien Sandras
(o-
//\    Ekiga Softphone : http://www.ekiga.org/
v_/_   Be IP           : http://www.beip.be/
       FOSDEM          : http://www.fosdem.org/
       SIP Phone       : sip:[email protected]




------------------------------

Message: 5
Date: Fri, 27 Feb 2009 16:53:42 +0100
From: Damien Sandras <[email protected]>
Subject: Re: [Ekiga-list] [ekiga engine] api availability and
        documentation
To: [email protected], Ekiga mailing list <[email protected]>
Message-ID: <[email protected]>
Content-Type: text/plain; charset=UTF-8

Le vendredi 27 f?vrier 2009 ? 08:53 +0100, Giampaolo Armellin a ?crit :
> I?ve read on Ekiga web site that the engine is available to be used in
> other projects. Nevertheless, I couldn?t find any reference to a SDK,
> API or documents.
>
>
>
> May anyone tell me where that information can be found?


I'm not sure it is completely doxygenified, but you could try generating the 
help from the headers.
--
 _     Damien Sandras
(o-
//\    Ekiga Softphone : http://www.ekiga.org/
v_/_   Be IP           : http://www.beip.be/
       FOSDEM          : http://www.fosdem.org/
       SIP Phone       : sip:[email protected]




------------------------------

Message: 6
Date: Fri, 27 Feb 2009 16:54:28 +0100
From: Damien Sandras <[email protected]>
Subject: Re: [Ekiga-list] call failure
To: Ekiga mailing list <[email protected]>
Message-ID: <[email protected]>
Content-Type: text/plain; charset=UTF-8

Le vendredi 27 f?vrier 2009 ? 20:52 +1100, Michael Stockenhuber a
?crit :
> Hi,
> I have replied and sent the logs but they are too big and wait for
> moderator approval.

It won't get approved, there are too many subscribers to this list.
Please post it to pastebin. However, I doubt a gdb backtrace is that
big.
--
 _     Damien Sandras
(o-
//\    Ekiga Softphone : http://www.ekiga.org/
v_/_   Be IP           : http://www.beip.be/
       FOSDEM          : http://www.fosdem.org/
       SIP Phone       : sip:[email protected]




------------------------------

Message: 7
Date: Fri, 27 Feb 2009 16:55:11 +0100
From: Damien Sandras <[email protected]>
Subject: Re: [Ekiga-list] WG: Connect to a LifeSize HD System
To: Ekiga mailing list <[email protected]>
Message-ID: <[email protected]>
Content-Type: text/plain; charset=UTF-8

Le jeudi 26 f?vrier 2009 ? 09:11 +0100, Stefan Stoedtgen a ?crit :
> Hi Ekiga List!
>
> I'm using Ekiga 3.0.2 stable under Windows XP, SP2 with a logitech
> QuickCam.
> I tried to connect to a LifeSize Room HD Conferencing Unit
> (www.lifesize.com) which uses SIP and H323
> for communication.
>
> The connection seems to be established, but then, it breaks down. I
> attached you the error.log.
> Can you help me with this problem? Thanks for your Support!

Breakdown = crash ?

Because I see nothing wrong. Perhaps you should try H.323.
--
 _     Damien Sandras
(o-
//\    Ekiga Softphone : http://www.ekiga.org/
v_/_   Be IP           : http://www.beip.be/
       FOSDEM          : http://www.fosdem.org/
       SIP Phone       : sip:[email protected]




------------------------------

Message: 8
Date: Fri, 27 Feb 2009 16:57:15 +0100
From: Damien Sandras <[email protected]>
Subject: Re: [Ekiga-list] Incoming call failure with 3.1.1
To: Ekiga mailing list <[email protected]>
Message-ID: <[email protected]>
Content-Type: text/plain; charset=UTF-8

Le jeudi 26 f?vrier 2009 ? 10:30 +0100, Alec Leamas a ?crit :
> Damien Sandras wrote:
> > Le mercredi 25 f?vrier 2009 ? 21:02 -0500, Mark T.B. Carroll a ?crit :
> >
> >> Damien Sandras <[email protected]> writes:
> >>
> >>
> >>> Le jeudi 19 f?vrier 2009 ? 00:32 -0500, Mark T.B. Carroll a ?crit :
> >>>
> >>>> I have run through the configuration assistant thing from start to
> >>>> finish. I can call the [email protected] echo test and that works just fine.
> >>>> However, calling the [email protected] callback service has it hang up on me
> >>>> and then ... nothing, though the -d 5 output shows that it is indeed
> >>>> getting a callback initiated from from sip:[email protected] which is then
> >>>> aborted. I am behind NAT and the router forwards incoming UDP from port
> >>>> 5060 to 5100 to the machine I'm using and acts as a gateway to let all
> >>>> my outgoing packets out.
> >>>>
> >>>> Should I gzip my -d 4 output and send it to somebody? I can also sniff
> >>>> packets and send pcap files. I use Debian; software versions are,
> >>>>
> >>> There is a known problem with incoming calls and the current snapshot. I
> >>> will fix it this week-end.
> >>>
> >> Now with the 20090225 one, the incoming call does arrive (yay!), I hit
> >> `accept', and it segfaults.
> >>
> >> I can still send my -d 4 output to somebody. (-:
> >>
> >
> > A gdb backtrace would be more useful.
> >
>
> waiting for some kind of customer "service", taking a backtrace (yes,
> same problem, trunk as of yesterday).

Despite trying 50 times, I can not reproduce it. Are you sure something
is not corrupted?

Eugen, can you reproduce it?
--
 _     Damien Sandras
(o-
//\    Ekiga Softphone : http://www.ekiga.org/
v_/_   Be IP           : http://www.beip.be/
       FOSDEM          : http://www.fosdem.org/
       SIP Phone       : sip:[email protected]




------------------------------

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