Re: [asterisk-dev] how does a channel driver know the result of codec negotiation ?

2007-10-15 Thread Olle E Johansson
14 okt 2007 kl. 21.47 skrev Luigi Rizzo: > On Sat, Oct 13, 2007 at 09:37:20AM +0200, Kevin P. Fleming wrote: >> Luigi Rizzo wrote: >> >>> i know - didn't want to make the story too long :) >>> (i suppose it is also a problem to pass the information across >>> a chain of asterisks) >> >> Yes, that

Re: [asterisk-dev] "lock up" on 1.4.13

2007-10-15 Thread Hendrik Visage
Uploaded the "core show locks" just before the system becomes unusable on Thursday to the bugtracker id=10956 On 10/12/07, Hendrik Visage <[EMAIL PROTECTED]> wrote: > http://bugs.digium.com/view.php?id=10956 Oh, please change the ASterisk version to 1.4.13 and put the correct module (I read CORE/

[asterisk-dev] app_dial.c code question

2007-10-15 Thread asterisk
In apps/app_dial.c in the function "wait_for_answer", single is set if it's only a a single destination AND not OPT_MUSICBACK and not OPT_RINGBACK. Why do MUSICBACK and RINGBACK remove this "single" setting? Andrew ___ --Bandwidth and Colocation

Re: [asterisk-dev] :SPAM: About Ringing Time out

2007-10-15 Thread Mayank Mathur
What config u've tried in your exten.conf and where u put in ??? Did u give it for 90 sec ?? Pl'z remember that u've to restart/reload Asterisk again after making changes. > Hello I have tried your configuration and not working. Still ringing for > 30s > only. Please help me. Shoud I redit the

Re: [asterisk-dev] russell: branch 1.4 r85717 - /branches/1.4/apps/app_queue.c

2007-10-15 Thread asterisk
I opened a bug... please post your info there. http://bugs.digium.com/view.php?id=10987 Quoting Martin Vít <[EMAIL PROTECTED]>: > i've tested this revision and i have the same results but my asterisk > installation crashes asap chan_sip loads > > *** glibc detected *** double free or corruption

Re: [asterisk-dev] [svn-commits] russell: branch 1.4 r85717 - /branches/1.4/apps/app_queue.c

2007-10-15 Thread Martin Vít
i've tested this revision and i have the same results but my asterisk installation crashes asap chan_sip loads *** glibc detected *** double free or corruption (fasttop): 0x0821a3c0 *** backtrace shows nothing usefull #0 0xb7df19e7 in ?? () #1 0xb7efaff4 in ?? () #2 0xb6e1fbb0 in ?? () #3

Re: [asterisk-dev] [svn-commits] russell: branch 1.4 r85717 - /branches/1.4/apps/app_queue.c

2007-10-15 Thread asterisk
This version just makes my asterisk process crash after 15 seconds or so. I'm on my way out the door so I don't have time to open a real bug now. Andrew Quoting SVN commits to the Digium repositories <[EMAIL PROTECTED]>: > Author: russell > Date: Mon Oct 15 15:59:27 2007 > New Revision: 85717

[asterisk-dev] About Ringing Time out

2007-10-15 Thread FRANCOIS
Hello I have tried your configuration and not working. Still ringing for 30s only. Please help me. Shoud I redit the chan_sip.c file? If yes, which variable should I need to change? And how? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mayank Mat

[asterisk-dev] automon

2007-10-15 Thread Clod Patry
hi guys, I just saw a patch went in for bug 6353 ( http://bugs.digium.com/view.php?id=6353 ) to be able to define the prefix for automon files and I don't think this is a good idea. I really dont see why we should have 2 channel vars for that ( TOUCH_MONITOR_FILENAME and TOUCH_MONITOR_PREFIX). Wh

[asterisk-dev] Fwd: [svn-commits] russell: trunk r85605 - in /trunk: ./ channels/chan_sip.c

2007-10-15 Thread Johansson Olle E
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

Re: [asterisk-dev] Tab-completion borked in trunk

2007-10-15 Thread Jared Smith
On Sun, 2007-10-14 at 10:59 -0700, John Todd wrote: > Additionaly, it does cause some functionality problems in an obscure > way since I don't know of any way to get a full SIP channel name > other than via tab completion - other methods truncate the name ("sip > show channels" doesn't provide a

Re: [asterisk-dev] :SPAM: Re: :SPAM: Re: :SPAM: About Ringing Time out

2007-10-15 Thread Diego Iastrubni
Giving Monty Python links to wikipedia is so lame. YouTube ownz: http://www.youtube.com/watch?v=wZ7YedEopp4 On 10/15/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > Wow, this is getting like the Monti-Python sketch. SPAM SPAM SPAM :-) > > http://en.wikipedia.org/wiki/Spam_(Monty_Python) > > Not

Re: [asterisk-dev] :SPAM: Re: :SPAM: Re: :SPAM: About Ringing Time out

2007-10-15 Thread Tzafrir Cohen
Wow, this is getting like the Monti-Python sketch. SPAM SPAM SPAM :-) http://en.wikipedia.org/wiki/Spam_(Monty_Python) Not to mention you capitalize SPAM just like SPAM.com: http://spam.com/legal/spam/ -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +97

Re: [asterisk-dev] :SPAM: Re: :SPAM: Re: :SPAM: About Ringing Time out

2007-10-15 Thread FRANCOIS
I have tried but not working. I still got the following answer: No Body Picked up 3 ms. This error from app_dial.c ( dial_exec) What should I do then? Thanks for your help -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mayank Mathur Sent: Monday, Octo

Re: [asterisk-dev] :SPAM: Re: :SPAM: Re: :SPAM: About Ringing Time out

2007-10-15 Thread Mayank Mathur
Add where ur adding all your extensions.. or add in default context . It depends on your Dialplan, where u r using yur extensions... > Thanks for your reply. > But in which section of extensions.conf file should I add this command? > If not which config file should I need to edit then? > I add