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| 03:34.22 | kn0x | anyone familiar with building asterisk inside linux-vservers? |
| 03:34.51 | kn0x | I followed some old instructions to trick asterisk with zaptel by copying libetonezone, zaptel.h and some others to the guest, but i couldnt find a dahdi.h and it configure isnt recognizing dahdi support.. I have dahdi running on the host and the /dev/dahdi/* devices are available inside the guest |
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| 09:19.06 | Kchehab | do asterisk sip realtime enabled on V 1.6.0.6 |
| 09:24.55 | Gugge | ? |
| 09:25.10 | Gugge | i dont even understand what you wanted to ask there .... |
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| 09:42.25 | flohack | Hi! Im trying to fix the segfault bug #15848, can someone help me with locking policy in chan_sip.c ? |
| 09:44.23 | flohack | I pasted the relevant code here: http://pastebin.com/m24f63e67 |
| 09:45.22 | Kchehab | Gugge i load res_realtime.so and did all config to have all sip peers integrated with mysql database |
| 09:45.51 | Kchehab | but it didnt work i was asking if this feature sip peers real time enabled on asterisk 1.6.0.6 |
| 09:46.19 | flohack | p->owner is already NULL on line 23. and what does the code try to do with the statement on lin 31? Why could the bridge we want to get a lock on change? |
| 09:49.01 | atis_work | flohack: do you use attended transfer? |
| 09:49.13 | flohack | atis_work: yes, from a queue call |
| 09:49.28 | atis_work | oh, ok, that explains why i don't have any issues |
| 09:49.36 | atis_work | i do a lot of blind transfers for queue calls |
| 09:49.53 | atis_work | flohack: you should mention that in the issue |
| 09:50.07 | flohack | atis_work: I'll add a note |
| 09:50.42 | Gugge | Kchehab: yes, sip realtime works in 1.6.1.6 |
| 09:50.43 | flohack | atis_work: could the bridge be freed by someone else between line 29 and 31? How can I check if the bridge is still there? |
| 09:50.59 | atis_work | chan->bridged |
| 09:51.04 | atis_work | or __bridged |
| 09:51.09 | atis_work | i don't remember exactly |
| 09:51.22 | atis_work | or there was ast_bridged_channel |
| 09:51.41 | atis_work | you should generally check that every time you lost lock on channel |
| 09:51.58 | flohack | atis_work: that was the plan :-) |
| 09:52.40 | atis_work | hmm, it's weird that p is assigned without any locks |
| 09:52.45 | atis_work | but i'm no pro |
| 09:53.19 | flohack | atis_work: ? which line? |
| 09:53.26 | atis_work | first in function |
| 09:53.58 | atis_work | flohack: gdb comes handy in debugging values at crash |
| 09:54.19 | atis_work | just use: "frame 1" to go to sip_hangup |
| 09:54.30 | atis_work | and "p ast" |
| 09:54.32 | atis_work | p p |
| 09:54.36 | atis_work | p is for print |
| 09:54.57 | flohack | atis_work: thanks, I'm pretty confident with using gdb :-) |
| 09:56.15 | flohack | atis_work: I already confirmed that chan is really NULL on line 29 |
| 09:57.05 | flohack | atis_work: I'll create a proposal and come back for review, thanks for your help! |
| 09:57.45 | atis_work | flohack: btw, how's T38 for you in 1.6.1? |
| 09:57.57 | atis_work | i couldn't get it working reliably with my provider |
| 09:58.14 | flohack | atis_work: working like a champ (in production, a few hundred faxes a work) |
| 09:58.31 | flohack | atis_work: a week |
| 09:58.58 | flohack | atis_work: I'm using a VoIP gateway which is right next to asterisk, never tried it with a SIP provider :-) |
| 10:01.23 | flohack | atis_work: I'd propose http://pastebin.com/m11604adf to fix it |
| 10:15.14 | atis_work | flohack: i won't be able to confirm that fix is good.. i'm just a newbie in C |
| 10:18.42 | flohack | atis_work: Which revision of the 1.6.1 branch are you using for T.38? I had to upgrade to revision 209515 to make it work (there are a lot of improvements from the released 1.6.1) |
| 10:19.46 | atis_work | flohack: well, i'm using 1.6.1.6, but i think i had few fixes applied from callweaver |
| 10:22.16 | flohack | atis_work: You could try vanilla asterisk from the 1.6.1 branch revision 209515, that's what works here with a third party T.38 implementation (teles.de PRI box) |
| 10:25.25 | atis_work | well, i think 1.6.1.6 has all the T38 patches currently out there |
| 10:26.59 | flohack | atis_work: probably, yes |
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| 12:38.53 | kazaa_lite | hi all |
| 12:42.44 | kazaa_lite | where are the release candidates for 1.4.26? I have explored http://downloads.asterisk.org/pub/telephony/asterisk/ but could not find it anywhere |
| 12:45.55 | Marquis42 | kazaa_lite: Why do you need RCs for 1.4.26? |
| 12:46.24 | kazaa_lite | to commit suicide:p |
| 12:46.36 | Marquis42 | Heh... |
| 12:47.27 | kazaa_lite | well there can be millions of reasons.... but it takes me to another discussion rather than demystifying where the rcs have gone? |
| 12:50.15 | Marquis42 | The reason I ask I guess is not to get into a philosophical discussion, really. AFAIK, the policy is that the RCs are pulled when the release is made. And since 1.4.26.2 is released, the former RCs are no longer there. |
| 12:53.28 | atis_work | kazaa_lite: http://lists.digium.com/pipermail/asterisk-commits/2009-July/thread.html |
| 12:53.39 | Marquis42 | OTOH, if you really need it you can check out the tag from Subversion. |
| 12:54.00 | kazaa_lite | where is the svn for asterisk? |
| 12:54.07 | kazaa_lite | that would make my life much easier |
| 12:54.26 | Marquis42 | http://www.asterisk.org/developer/resources/svn |
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| 12:54.56 | Marquis42 | http://svn.digium.com/svn/asterisk/tags/1.4.26-rc1/ <--- that would, for example, be the path for checking out 1.4.26-rc1 |
| 12:54.58 | atis_work | aah, rc's aren't removed from svn |
| 12:55.22 | Marquis42 | Nope, since they're just tags anyway there is no particular reason to remove them. |
| 13:02.50 | kazaa_lite | cool |
| 13:02.52 | kazaa_lite | love you |
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| 17:22.22 | atis_work | can anybody fix WARNING message in app_controlplayback? It should actually be VERBOSE or DEBUG |
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| 19:47.19 | moy | isn't using select()/FD_SET and friends in Asterisk effectively limiting file descriptor usage to FD_SETSIZE that typically is 1024? |
| 19:55.25 | moy | particularly res/res_timing_thread.c |
| 19:55.43 | moy | I think it should use poll instead of select in read_pipe() |
| 19:58.53 | moy | I am pushing lots of rtp sessions that surely exceed 1024 limit, at some point after exceeding the 1024 limit asterisk starts consuming lots of cpu apparently some threads loop like crazy, I believe is due to select() returning constantly file descriptors as ready due to fd_set overflow |
| 20:01.47 | moy | how does it sound if I create a new compilation flag to replace ast_select() implementation using poll ? or something like that? |
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| 22:56.26 | CcRnp | can anyone help me ! i want to use video on asterisk and it seems like i dont have H.263 codec install on my system where can i get source of this codec ? |
| 22:57.52 | kpfleming | you don't need the codec to handle video calls with asterisk |
| 22:58.23 | kpfleming | you only need that to connect dissimilar video calls together |
| 22:58.25 | kpfleming | and those questions should be asterisk in #asterisk |
| 22:58.34 | CcRnp | thank you |
| 22:58.57 | russellb | kpfleming does Asterisk support in #2,000 |
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| 23:20.54 | snuff-work | mm how goes the astricon.. |