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| 08:48.49 | ixyd_ | is it possible to customise the "my view" in mantis? i would like to see my monitored issues above of the unassigned etc... |
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| 09:02.07 | schmidts | hello |
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| 14:41.53 | philm | Hi |
| 14:44.05 | philm | I'm try to create an channel that works with raw rtp streams. I started with chan_multicastrtp. I have it sending out and rtp stream, but can figure out why it won't receive an rtp stream. |
| 14:45.10 | philm | I don't even see anything coming in on with "rtp set debug on", just going out. Any ideas? |
| 14:51.53 | kpfleming | yes. something is wrongg. |
| 14:54.45 | philm | When I create a new ast_rtp_instance_new do I need to do anything special for it to receive a stream? |
| 14:55.42 | kpfleming | no. |
| 14:55.55 | philm | I'm doing a ast_channel_alloc and ast_rtp_instance_set_remote_address after that. |
| 14:55.58 | philm | hmmm |
| 14:58.14 | philm | have any suggestions on where I can look. Between streaming rtp and "rtp set debug on". |
| 14:58.28 | philm | I know .read on the channel isn't being called. |
| 14:58.49 | kpfleming | not off the top of my head, no. your best bet is probably to post the code somewhere and ask on the asterisk-dev mailing list for someone to take a look at it. |
| 14:59.52 | philm | Will do. kpfleming, Thanks for trying. |
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| 17:28.47 | alecdavis | https://reviewboard.asterisk.org/r/1185/ fixes a few pickup() issues. |
| 17:29.06 | alecdavis | deadlock mainly in 1.8 and trunk. |
| 17:30.17 | alecdavis | removes pickup success/failed beep from dialplan pickup(), pickup feature code unchanged. |
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| 18:39.08 | pabelanger | So, I know the merge patch for branches is forward, but how would somebody backport a revision from trunk into 1.8? The patch should have been applied to 1.8 too |
| 18:40.15 | BMJ | Asterisk SCF Developer Call today at 5:00 EDT. Info here: https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Developer+Call+-+04212011+-+1700+EST |
| 19:24.34 | russellb | pabelanger: just manually |
| 19:24.41 | russellb | pabelanger: and then "record8trunk" on trunk |
| 19:24.53 | russellb | to note that the code from that rev is already in trunk |
| 19:26.00 | pabelanger | russellb: danke |
| 19:26.17 | russellb | np |
| 19:33.43 | alecdavis | any objection to committing https://reviewboard.asterisk.org/r/1185/ to trunk? |
| 19:33.49 | alecdavis | to get soem testig doen. |
| 19:34.27 | alecdavis | ^^ to get some testing done. typed slower :) |
| 19:39.41 | pabelanger | I might we wrong, but I don't think we merge code into trunk for testing :) |
| 19:39.46 | pabelanger | gah! |
| 19:39.52 | pabelanger | s/we/be |
| 19:46.29 | alecdavis | it's good tested code. just need a wider audience. |
| 20:10.40 | alecdavis | ast_pickup_call since the birth of 1.8 r278985 when chan_dahdi is involved has a choppy/distorted beep http://pastebin.ca/2049122 |
| 20:11.13 | alecdavis | ast_pickup_call works fine with 1.6.2 and 1.4 |
| 20:18.54 | tzafrir | asterisk <= 1.6.2 defaults to sending out SIP/TCP calls in SSLv2? http://svnview.digium.com/svn/asterisk?view=revision&revision=191177 |
| 20:19.46 | tzafrir | (I'm trying to understand what this means, exactly) |
| 20:22.22 | BMJ | Asterisk SCF Developer call at 5:00 Eastern. Info here: https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Developer+Call+-+04212011+-+1700+EST |
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| 20:57.28 | alecdavis | pabelanger: if your running 1.8 or trunk, you can test the pickup() issue in a sip only environment, with pickupsounds enabled. |
| 20:58.10 | alecdavis | it will deadlock any 1.8 release since it's birth way back in July 2010. |
| 20:58.52 | pabelanger | alecdavis: what about creating a test for the testsuite to automate it? |
| 21:00.23 | alecdavis | any template code for writing tests? |
| 21:02.36 | pabelanger | alecdavis: http://svn.digium.com/svn/testsuite/asterisk/trunk/tests/ |
| 21:03.46 | alecdavis | and then I'll have to commit my proposal, to make the bamboo failure go away. |
| 21:03.49 | alecdavis | ;) |
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