02:49.52 | *** join/#asterisk-bugs infobot (ibot@rikers.org) |
02:49.52 | *** topic/#asterisk-bugs is Asterisk Bugs Discussion -=- http://issues.asterisk.org/ -=- http://asterisk.org/developers/bug-guidelines - Discussions not related to reported issues should be in #asterisk -=- Join #asterisk-testing for help with testing issues marked Ready for Testing -=- http://xkcd.org/583/ -=- An issue tracker full of activity is a sign of a healthy project |
04:31.46 | *** join/#asterisk-bugs chazzam (~chazz@173-24-239-247.client.mchsi.com) |
06:41.41 | Entomologist | *** CLOSED (18194) [Channels/chan_iax2] [patch] Loading chan_iax2.so allocates >100M RSS |
06:41.42 | Entomologist | Reported by: job |
06:41.43 | Entomologist | https://issues.asterisk.org/view.php?id=18194 |
06:41.43 | Entomologist | ********************************************************* |
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10:47.16 | *** join/#asterisk-bugs Dovid (~Dovid@213.8.118.62) |
10:48.39 | Dovid | can anyone have a look at: https://issues.asterisk.org/view.php?id=17954#131100 |
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13:52.36 | Entomologist | *** CLOSED (18456) [General] [patch] Asterisk HTTP response contains wrong Content-type |
13:52.37 | Entomologist | Assigned to: lathama |
13:52.37 | Entomologist | Reported by: alexo |
13:52.38 | Entomologist | https://issues.asterisk.org/view.php?id=18456 |
13:52.38 | Entomologist | ********************************************************* |
13:54.19 | *** join/#asterisk-bugs Dovid (Dovid@office.mypbxmanager.net) |
14:23.29 | Entomologist | *** CLOSED (18648) [Applications/app_meetme] Conference will not record with DAHDI loaded, but no hardware |
14:23.30 | Entomologist | Reported by: covici |
14:23.30 | Entomologist | https://issues.asterisk.org/view.php?id=18648 |
14:23.31 | Entomologist | ********************************************************* |
14:51.24 | *** join/#asterisk-bugs elguero (~miguel323@12.187.84.162) |
14:54.51 | seanbright | leifmadsen: |
14:54.52 | seanbright | M18714 |
14:54.53 | MuffinMan | [new] [Asterisk] Codecs/General 0018714: rev. 305040: asterisk is not initialized because the g729a-codec is crashing reported by Netview https://issues.asterisk.org/view.php?id=18714 |
14:54.55 | seanbright | digium support, yes? |
14:55.07 | file | yes |
14:56.09 | seanbright | k |
14:57.08 | Entomologist | *** CLOSED (18714) [Codecs/General] rev. 305040: asterisk is not initialized because the g729a-codec is crashing |
14:57.08 | Entomologist | Reported by: Netview |
14:57.09 | Entomologist | https://issues.asterisk.org/view.php?id=18714 |
14:57.09 | Entomologist | ********************************************************* |
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15:25.37 | Dovid | anyone know's if this isa bug or just how Asterisk is supposed to work ? http://lists.digium.com/pipermail/asterisk-users/2011-January/258628.html |
15:32.31 | elguero | Issue 18692 was orginally setup to take care of a compile error in app_voicemail.c when ODBC storage was enabled. At the same time, I added a patch to enable ODBC resequencing. But it looks like tilghman took care of the original compile problem early this morning. Should I setup another issue that takes care getting ODBC resequencing to take place since that now seems like a different issue and we close out this issue? |
15:32.33 | MuffinMan | [new] [Asterisk] Applications/app_voicemail 0018692: [patch] Compile Error - odbc_storage enabled reported by elguero https://issues.asterisk.org/view.php?id=18692 |
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16:31.04 | Entomologist | *** CLOSED (18644) [Applications/app_confbridge] hint for confbridge fails |
16:31.05 | Entomologist | Reported by: seandarcy |
16:31.05 | Entomologist | https://issues.asterisk.org/view.php?id=18644 |
16:31.06 | Entomologist | ********************************************************* |
16:33.29 | *** join/#asterisk-bugs russellb_ (~russell@asterisk/digium-open-source-team-lead/russellb) |
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18:08.23 | leifmadsen | russellb: omg 47 new issues |
18:08.34 | leifmadsen | this was 33 when I was done on Friday... I'm back right where I started |
18:25.50 | *** join/#asterisk-bugs Dovid (~Dovid@213.8.121.90) |
18:29.36 | leifmadsen | seanbright: ping? |
18:29.40 | leifmadsen | M18152 |
18:29.42 | MuffinMan | [new] [Asterisk] Addons/chan_mobile 0018152: Channel Variable SMSSRC not set properly reported by menschentier https://issues.asterisk.org/view.php?id=18152 |
18:29.56 | leifmadsen | that looks like a really easy issue to close perhaps? |
18:35.03 | leifmadsen | Dovid: your link from earlier about 17954 please file as a separate issue with console output, SIP debugging, and the dialplan used to recreate the scenario. I don't believe they are related. |
18:36.55 | Dovid | leifmadsen: Thank you. does it look like a bug ? |
18:37.03 | leifmadsen | shrugs |
18:37.11 | leifmadsen | hard to tell without seeing the information requested |
18:37.12 | Dovid | leifmadsen: It has been this way since 1.2.X. never looked in to it till now that i needed to. |
18:37.18 | leifmadsen | I think it might be a configuration issue |
18:37.18 | Dovid | will file one early tomorrow. thanks |
18:37.40 | leifmadsen | my gut instinct tells me it isn't a bug |
18:37.43 | Dovid | leifmadsen: how so ? Asterisk gets the BYE, responds to it yet does not hang up till the second leg does |
18:37.44 | Dovid | ok |
18:37.47 | Dovid | will post tomorrow |
18:37.55 | leifmadsen | shrugs |
18:42.30 | leifmadsen | russellb: https://issues.asterisk.org/view.php?id=17954#131134 |
18:42.32 | leifmadsen | regression? |
18:54.22 | file | one channel is basically sitting there in limbo until the other is done doing what it needs to do (for example playing a sound file to the channel) |
18:54.31 | file | then it comes out of limbo and the hangup is seen |
18:54.40 | file | but while in limbo it isn't watched |
19:09.27 | Entomologist | *** CLOSED (18673) [Channels/chan_gtalk] when using chan_gtalk the tos bit set in either the sip.conf and or the iax,conf is ignored |
19:09.27 | Entomologist | Reported by: stevenla |
19:09.28 | Entomologist | https://issues.asterisk.org/view.php?id=18673 |
19:09.28 | Entomologist | ********************************************************* |
19:20.53 | leifmadsen | file: so not a bug, and is just the way it works? |
19:21.00 | file | oui |
19:21.05 | leifmadsen | Dovid: see above |
19:21.15 | leifmadsen | file: you're talking about Dovid's issue right and not the regression? |
19:21.19 | file | oui |
19:21.22 | leifmadsen | (possible regression) |
19:42.34 | Entomologist | *** CLOSED (18678) [Applications/app_queue] [regression] Transfer from queue agent does not change his state |
19:42.34 | Entomologist | Reported by: jamicque |
19:42.35 | Entomologist | https://issues.asterisk.org/view.php?id=18678 |
19:42.35 | Entomologist | ********************************************************* |
19:47.56 | Entomologist | *** CLOSED (18682) [Applications/app_dial] When using u and g in dial if the call is bridged there is no audio once called hangs up |
19:47.56 | Entomologist | Reported by: Dovid |
19:47.57 | Entomologist | https://issues.asterisk.org/view.php?id=18682 |
19:47.57 | Entomologist | ********************************************************* |
19:51.43 | Dovid | leifmadsen: WHy was the issue closed ? |
19:52.19 | leifmadsen | that is the issue we were talking about earlier right? |
19:52.24 | Dovid | <PROTECTED> |
19:52.29 | leifmadsen | ok fine |
19:52.55 | leifmadsen | looked like it |
19:52.57 | Dovid | i need to test in a non NAT enviroment as perpabelanger |
19:53.06 | Dovid | the issue today is that the channel is in limbo |
19:54.02 | Dovid | and if you thought it was the same issue why did you close it ? |
19:55.04 | leifmadsen | Dovid: because file already responded that your other issue is not likely a bug |
19:55.11 | leifmadsen | <file> one channel is basically sitting there in limbo until the other is done doing what it needs to do (for example playing a sound file to the channel) |
19:55.11 | leifmadsen | <file> then it comes out of limbo and the hangup is seen |
19:55.11 | leifmadsen | <file> but while in limbo it isn't watched |
19:55.18 | leifmadsen | anyways the issue is reopened |
19:55.23 | leifmadsen | and I set it to feedback |
19:56.43 | Dovid | ok. but why is it in limbo. if Asterisk gets the BYE and responds to it shouldnt it then hang up. why does it wait for the second leg to hangup b4 hanging up the first leg ? |
19:58.08 | Corydon76-home | Dovid: because it needs to coordinate the final hangup code between the two ends |
19:59.06 | Dovid | so why wont asterisk send a BYE to the second leg ? why does it insist on waiting for a BYE from LegB of the call |
20:00.07 | Dovid | the only difference than a regular call is that here they are not bridged but asterisk should know they are the same since if it gets a positive result from the go sub it will bridge them. if this is the case when the first leg sends a BYE why doesent asterisk send the same to the second call ? |
20:02.27 | Corydon76-home | Dovid: once you provide the additional information, we can make that determination |
20:03.28 | Dovid | Corydon76-home: No problem. so SIP debug, core verbose 15, core debug 15 and dial plan in bug report ? |
20:03.31 | Corydon76-home | Or, you can work on fixing it, yourself, and provide the fix back |
20:04.14 | Corydon76-home | Dovid: for completeness, I'd also add in a packet dump |
20:04.58 | Corydon76-home | The stuff internal to Asterisk only helps us in determining what Asterisk is doing. If there's something that Asterisk is ignoring, then we need the packet dump |
20:05.01 | Corydon76-home | (also) |
20:05.31 | Dovid | Corydon76-home: If only I knew C............... I have a friend who knows C but not much about Asterisk. How hard you think it would be to figure out the code ? Also can you reccomend anyone that I can pay to have a look at it ? right now all I have to go on is reporting it here. if i could fix it myself I would ;). |
20:05.41 | Dovid | When you say packet dump you mean tcpdump or something in Asterisk ? |
20:06.02 | leifmadsen | M18689 |
20:06.03 | MuffinMan | [new] [Asterisk] CDR/General 0018689: Calls didn't appear in CDR reported by DJ Kill https://issues.asterisk.org/view.php?id=18689 |
20:06.05 | Corydon76-home | tcpdump, or (better) wireshark |
20:06.19 | leifmadsen | I have no idea what to ask for, but I don't think he has provided enough information |
20:06.41 | Corydon76-home | tcpdump can drop important parts of the packet, depending upon the default configuration of the distribution |
20:06.56 | Dovid | so wireshark on the local machine would be better ? |
20:07.37 | Corydon76-home | Wireshark is one of those programs which never seems to provide too little information |
20:07.52 | Dovid | lol. ok. I thought tcpdump and wireshark worked the same. |
20:08.12 | Corydon76-home | They do, but tcpdump's default output may have truncated packets |
20:08.16 | Dovid | dont have a linux box here local. will do it once I am in the office tomorrow and then create a bug report. will do the same for the other bug |
20:08.19 | leifmadsen | I prefer using tshark |
20:08.28 | Dovid | ok. I can use tshark as well if you want. |
20:08.31 | leifmadsen | (over tcpdump) |
20:08.52 | Dovid | what's better. WireShark or TShark ? |
20:08.58 | Corydon76-home | tshark and wireshark are the same program. It's just that wireshark requires a graphical display |
20:09.09 | leifmadsen | right |
20:09.34 | Dovid | ok. so wireshark kicks tcpdump in the but. got it. would you like just SIP ? or all (including RTP) ? |
20:10.24 | Corydon76-home | See comment above about providing too little information. ;-) |
20:10.50 | Dovid | ok. i will grab it all and upload. dont say i gave too much ;) |
20:12.00 | Corydon76-home | Dovid: given that a developer will load it into wireshark for analysis, eliminating any extra data is easy (just apply a filter) |
20:12.47 | Corydon76-home | It's one of those reasons why it's best to use whatever the developers are using... |
20:14.30 | Dovid | Corydon76-home: I wont argue. I like filtering on capture so I don't grab what I don't need. when you are tracing 20-30 mbits of rtp u dont want to cpature what you dont need |
20:17.05 | leifmadsen | M18692 |
20:17.07 | MuffinMan | [ready for review] [Asterisk] Applications/app_voicemail 0018692: [patch] Compile Error - odbc_storage enabled reported by elguero https://issues.asterisk.org/view.php?id=18692 |
20:19.57 | leifmadsen | if someone is reporting a memory leak, what information should I be asking them for? |
20:46.04 | Corydon76-home | MALLOC_DEBUG, memory show allocations |
20:46.22 | Corydon76-home | Or actually, memory show summary |
20:46.39 | Corydon76-home | memory show summary before and after the leak |
20:48.20 | leifmadsen | awesome thanks |
20:49.01 | leifmadsen | rtautoclear=yes isn't expected to clear registrations from memory when rtcachfriends=yes is in use right? |
20:49.14 | leifmadsen | when a sip peer unregisters, should it be removed from memory? |
20:49.19 | leifmadsen | (when using realtime) |
20:50.07 | Corydon76-home | leifmadsen: Incorrect, and yes. |
20:50.44 | Corydon76-home | rtautoclear=yes only applies with rtcachefriends is on |
20:51.01 | Corydon76-home | otherwise, the peer is cleared from memory at every opportunity |
20:51.12 | leifmadsen | what I mean is rtautoclear=yes is used to remove peers from memory with rtcachefriends=yes when the registration expires, it isn't what causes registrations to be removed from memory when the peer unreg's |
20:51.25 | leifmadsen | ok, then someone has a bug I guess |
20:51.43 | Corydon76-home | Yes, it is |
20:51.56 | Corydon76-home | registrations are otherwise never removed from memory |
20:52.15 | leifmadsen | even when the peer unreg's? |
20:52.19 | Corydon76-home | Even then |
20:52.45 | leifmadsen | ok, the documentation should probably be updated then since they way I read it, rtautoclear is only used when the registration timeout expires |
20:52.54 | leifmadsen | (or every XX seconds if set to a number) |
20:53.07 | Corydon76-home | This is why the realtime infrastructure in SIP needs a massive overhaul to make it do "the right thing" most of the time and bag all of these extra options |
20:53.15 | leifmadsen | aye |
20:56.39 | leifmadsen | thx |
20:57.17 | Entomologist | *** CLOSED (18710) [Applications/app_parkandannounce] Cannot park a call more than once |
20:57.18 | Entomologist | Reported by: gentlec |
20:57.18 | Entomologist | https://issues.asterisk.org/view.php?id=18710 |
20:57.18 | Entomologist | ********************************************************* |
21:02.35 | Entomologist | *** CLOSED (18631) [PBX/General] Asterisk crashes on "dialplan reload" |
21:02.36 | Entomologist | Reported by: greenfieldtech |
21:02.36 | Entomologist | https://issues.asterisk.org/view.php?id=18631 |
21:02.37 | Entomologist | ********************************************************* |
21:08.34 | Entomologist | *** CLOSED (18708) [Channels/chan_sip/General] sip_xmit warning |
21:08.34 | Entomologist | Reported by: din3sh |
21:08.35 | Entomologist | https://issues.asterisk.org/view.php?id=18708 |
21:08.35 | Entomologist | ********************************************************* |
21:11.47 | Entomologist | *** CLOSED (18701) [Channels/chan_sip/CallCompletionSupplementaryServices] no Connected Line Presentation (COLP) transparency for SIP to SIP calls via Asterisk |
21:11.47 | Entomologist | Reported by: roxy |
21:11.48 | Entomologist | https://issues.asterisk.org/view.php?id=18701 |
21:11.48 | Entomologist | ********************************************************* |
21:16.52 | leifmadsen | M18704 |
21:16.54 | MuffinMan | [new] [Asterisk] Channels/chan_sip/Interoperability 0018704: "Require: timer" header still being sent reported by mfrager https://issues.asterisk.org/view.php?id=18704 |
21:17.01 | leifmadsen | I don't understand how that issue gives us anything to go on... |
21:19.47 | leifmadsen | oh he added /* <-- PROBLEM! */ in the code block |
21:20.00 | leifmadsen | The_Boy_Wonder: ping |
21:20.16 | leifmadsen | the issue above refers to 17005 which you closed |
21:20.29 | leifmadsen | mind taking a quick look and telling me if there is a bug there? |
21:20.35 | leifmadsen | (or if there is enough information to go on) |
22:22.44 | The_Boy_Wonder | leifmadsen: hey |
22:41.21 | The_Boy_Wonder | leifmadsen: i commented on that issue |
22:56.58 | Entomologist | *** CLOSED (18371) [Channels/chan_sip/General] [patch] asterisk crash when dialing SIP/${var} where var is empty or not set |
22:56.59 | Entomologist | Assigned to: qwell |
22:56.59 | Entomologist | Reported by: gbour |
22:57.00 | Entomologist | https://issues.asterisk.org/view.php?id=18371 |
22:57.00 | Entomologist | ********************************************************* |