| 01:57.57 | *** join/#asterisk-dev Kobaz (~kobaz@its.kobaz.net) |
| 02:53.56 | leifmadsen | Micc, once we decide we should work on it :) |
| 02:58.58 | *** join/#asterisk-dev mjordan (~mjordan@75.76.55.191) |
| 02:58.58 | *** mode/#asterisk-dev [+o mjordan] by ChanServ |
| 03:21.58 | *** join/#asterisk-dev leedm777 (~leedm777@50-81-28-152.client.mchsi.com) |
| 05:12.41 | *** join/#asterisk-dev doome_ (~doome@82.150.48.146) |
| 05:44.26 | *** join/#asterisk-dev MrTAP (~MrT@76-10-184-89.dsl.teksavvy.com) |
| 06:21.59 | *** join/#asterisk-dev hehol (~hehol@gatekeeper.loca.net) |
| 06:57.38 | *** join/#asterisk-dev ak77 (c12e4b04@gateway/web/freenode/ip.193.46.75.4) |
| 06:57.42 | ak77 | hello all |
| 07:21.52 | *** join/#asterisk-dev tzafrir (~tzafrir@local.xorcom.com) |
| 07:21.52 | *** mode/#asterisk-dev [+o tzafrir] by ChanServ |
| 07:23.55 | *** join/#asterisk-dev pchero_work (~pchero@109.70.54.56) |
| 08:49.08 | *** join/#asterisk-dev oej (~oej@95.209.24.201.bredband.tre.se) |
| 08:49.08 | *** mode/#asterisk-dev [+o oej] by ChanServ |
| 09:28.23 | *** join/#asterisk-dev oej (~oej@host-95-199-11-44.mobileonline.telia.com) |
| 09:28.23 | *** mode/#asterisk-dev [+o oej] by ChanServ |
| 09:41.34 | ak77 | where to set logging so that ast_debug(1, ...) will be output to console ? |
| 10:41.56 | *** join/#asterisk-dev oej (~oej@95.209.24.201.bredband.tre.se) |
| 10:41.56 | *** mode/#asterisk-dev [+o oej] by ChanServ |
| 11:56.24 | *** join/#asterisk-dev seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 11:56.24 | *** mode/#asterisk-dev [+o seanbright] by ChanServ |
| 12:07.01 | *** join/#asterisk-dev mjordan (~mjordan@75.76.55.191) |
| 12:07.01 | *** mode/#asterisk-dev [+o mjordan] by ChanServ |
| 12:36.10 | opticron | ak77, first, you need to make sure one of your logger channels has the debug flag (logger.conf is where they are defined) |
| 12:37.38 | opticron | if you have to add it, run "logger reload" on the asterisk CLI. then run "core set debug 1" on the CLI |
| 12:45.10 | *** join/#asterisk-dev sgalarneau (~Adium@user-24-96-87-214.knology.net) |
| 13:08.04 | *** join/#asterisk-dev leedm777 (leedm777@nat/digium/x-egisvwurhgdplpob) |
| 13:17.18 | *** join/#asterisk-dev [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:18.05 | *** part/#asterisk-dev mjordan (~mjordan@75.76.55.191) |
| 13:34.30 | *** join/#asterisk-dev sgalarneau (Adium@nat/digium/x-dowwrkbnymiqkycr) |
| 13:34.59 | *** join/#asterisk-dev sgalarneau (Adium@nat/digium/x-lpjcbmrbyzskzubd) |
| 14:10.04 | *** join/#asterisk-dev mjordan (mjordan@nat/digium/x-rpxpeqjqerrsjbdn) |
| 14:10.04 | *** mode/#asterisk-dev [+o mjordan] by ChanServ |
| 14:12.47 | *** join/#asterisk-dev putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:12.47 | *** mode/#asterisk-dev [+o putnopvut] by ChanServ |
| 14:17.13 | *** join/#asterisk-dev sgriepentrog (sgriepentr@nat/digium/x-mfakrwbdfwwpqese) |
| 14:18.55 | *** join/#asterisk-dev kharwell (kharwell@nat/digium/x-ntupogfoaqvxfdlv) |
| 14:19.52 | *** join/#asterisk-dev sruffell (sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:19.52 | *** mode/#asterisk-dev [+o sruffell] by ChanServ |
| 14:40.44 | putnopvut | So having local channels forward -1 frames has caused some tests to start failing. All failures seem to be because expected events are not what they seem anymore. |
| 14:41.03 | putnopvut | I have confirmed that not propagating -1 control frames fixes this issue with the failing tests. |
| 14:41.07 | file | putnopvut, blargh |
| 14:41.24 | putnopvut | Since I had other proposed methods of fixing that problem, should I go with one of those instead? |
| 14:42.26 | file | putnopvut, do you know what ripples that actually created (why events are different)? |
| 14:43.11 | putnopvut | Nope, I don't know the specifics. So far, the affected events seem to relate to hangup cause and some dialbegin/dialend events when local channels are involved. |
| 14:43.47 | file | I think it'd be useful to know, at the very least to expand the comment in case someone else has the idea of removing it |
| 14:45.25 | file | it's sorta ... odd ... that it would break things like that |
| 14:47.58 | putnopvut | Agreed |
| 14:54.31 | putnopvut | Looking at the pjsip/forward_loop test, it's weird because ZERO DIalBegin or DialEnd events are emitted. |
| 14:54.47 | putnopvut | It's not like the information is incorrect. They're just not there at all. |
| 14:56.17 | file | very ... odd |
| 15:27.33 | *** join/#asterisk-dev superscrat (asanders@nat/digium/x-hrsupgzzsbmwgtob) |
| 15:48.55 | *** join/#asterisk-dev rmudgett (rmudgett@nat/digium/x-tucxzetxsyzfralr) |
| 16:01.17 | *** join/#asterisk-dev superscrat (asanders@nat/digium/x-xxitlqcvopligzwb) |
| 16:16.57 | *** join/#asterisk-dev jameswf (uid27319@gateway/web/irccloud.com/x-zchjoliwllovdcjo) |
| 16:20.02 | *** join/#asterisk-dev anonymouz666 (~anonymouz@189-25-24-119.user.veloxzone.com.br) |
| 16:29.03 | *** join/#asterisk-dev fornax (~fornax@p5DD144F4.dip0.t-ipconnect.de) |
| 16:52.28 | *** join/#asterisk-dev babak (uid19622@gateway/web/irccloud.com/x-bhowsecspmxpjwhg) |
| 17:50.45 | *** join/#asterisk-dev fornax (~fornax@p54BD16DF.dip0.t-ipconnect.de) |
| 18:55.49 | *** join/#asterisk-dev sgalarneau (Adium@nat/digium/x-badnixvbxwqxsiqw) |
| 19:02.09 | *** join/#asterisk-dev mjordan (mjordan@nat/digium/x-mhfbkldytdumecrk) |
| 19:02.09 | *** mode/#asterisk-dev [+o mjordan] by ChanServ |
| 19:59.10 | *** join/#asterisk-dev sruffell (sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:59.10 | *** mode/#asterisk-dev [+o sruffell] by ChanServ |
| 20:04.25 | seanbright | so this is a new one... we're getting calls with a To header that looks like this: |
| 20:04.28 | seanbright | To: "" <sip:5556667777;cic=3333@1.2.3.4>;tag=as0948d83c |
| 20:04.51 | seanbright | wtf is cic? |
| 20:05.28 | file | circuit identification code |
| 20:06.01 | seanbright | is there a way to strip that off before chan_sip gets a hold of it? |
| 20:06.05 | file | although in SIP world maybe... carrier identification code! |
| 20:06.27 | *** join/#asterisk-dev fornax (~fornax@85.183.43.221) |
| 20:06.53 | file | um, hack chan_sip to remove it or put a SIP proxy in front and remove it there? |
| 20:07.13 | seanbright | perfect |
| 20:07.15 | seanbright | danke |
| 20:07.26 | file | or drink |
| 20:10.25 | seanbright | well, given those 3 options, you know what i'll opt for |
| 21:02.36 | *** join/#asterisk-dev fornax (~fornax@85.183.43.221) |
| 21:05.36 | *** join/#asterisk-dev sruffell (sruffell@asterisk/the-kernel-guy/sruffell) |
| 21:05.36 | *** mode/#asterisk-dev [+o sruffell] by ChanServ |
| 21:06.52 | *** join/#asterisk-dev prelude2004c (~IceChat9@dsl-67-55-28-3.acanac.net) |
| 21:06.53 | prelude2004c | hey guys |
| 21:07.33 | prelude2004c | can anyone help me and tell me if this is a bug.. i have 2 softphones working with Iax2 ... i get calls and i have one device that i hear the audio just fine in/out and the other i can't talk back |
| 21:07.42 | prelude2004c | but.. the odd part is... if i put them on hold , they can hear the on hold music |
| 21:07.48 | prelude2004c | but when i bring the call back, they can't hear me |
| 21:07.51 | prelude2004c | i dont know if this is a bug or something more |
| 21:07.53 | prelude2004c | trunk=no |
| 21:22.47 | prelude2004c | anyone? |
| 21:39.55 | *** join/#asterisk-dev superscrat (asanders@nat/digium/x-gidnfrewfczuvjbl) |
| 23:56.50 | *** join/#asterisk-dev infobot (~infobot@174-148-182-189.pools.spcsdns.net) |
| 23:56.50 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Hell hath frozen over: Asterisk is moving to Git -=- Follow on Twitter at @AsteriskDev |