| 00:08.46 | bklang | I think I've figured it out |
| 00:08.46 | bklang | the SDP is so large that the packet is fragmenting |
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| 00:15.25 | bklang | Yep that was it. Packet fragmentation |
| 00:15.34 | bklang | Asterisk was just *too* capable |
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| 06:26.49 | schmidts | good morning |
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| 11:06.06 | grEvenX | hm |
| 11:06.11 | grEvenX | okey, this is pretty bad and serious |
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| 11:07.00 | grEvenX | if I've found something that will cause any asterisk 1.4.x server to "stall" for a while, or even DDOS it pretty simple, would it be of interest to the asterisk community to fix it? |
| 11:07.39 | schmidts | 1.4 is end of life but if you can reproduce this with 1.8 i am sure its interesting |
| 11:07.50 | grEvenX | (I'm not sure if this goes under "security fix" or not, since regular asterisk 1.4 support is eol) |
| 11:08.01 | grEvenX | no, it's luckily fixed in 1.8 it seems |
| 11:08.08 | grEvenX | at least according to my own tests |
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| 11:08.21 | schmidts | how did you make this? |
| 11:08.33 | Aurs | art by accident |
| 11:08.41 | schmidts | :D |
| 11:08.55 | schmidts | and what have you found? |
| 11:09.03 | grEvenX | it's pretty simple, just send a malformed SIP REGISTER to a server |
| 11:09.14 | grEvenX | so I would say it's pretty bad |
| 11:10.06 | schmidts | can you give me an example how this register message is malformed or what do you mean with this |
| 11:10.26 | grEvenX | I'm a bit wary of just giving out all the details since the info could get in the wrong hands |
| 11:10.52 | grEvenX | so i'm not sure what the proper channel for those kind of issues is |
| 11:11.19 | schmidts | grEvenx you are just right here but you can also open a private issue on the asterisk bug tracker, only devs would be able to see it |
| 11:11.39 | grEvenX | ok, that sounds better |
| 11:12.04 | grEvenX | it's very simple to reproduce with a sipp scenario |
| 11:17.32 | Aurs | schmidts: isn't 1.4 EOL date 2012-04-21? (https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions) |
| 11:18.25 | schmidts | Aurs you are right, i forgot ;) |
| 11:19.24 | schmidts | but its even over the "security fix only" state |
| 11:20.09 | Aurs | isn't it security fix only FROM the given date? |
| 11:20.15 | Aurs | until EOL? |
| 11:20.57 | Aurs | 1 year with security fix only, after regular support is expired |
| 11:21.21 | Aurs | (april 21st 2011 - april 21st 2012) |
| 11:21.27 | schmidts | also possible, i have read it the other way until this date but it could also be after this date ;) |
| 11:21.41 | Aurs | so I'd say that it is _in_ security fix only state, and not _over_ it :) |
| 11:22.07 | Aurs | that wouldn't make any sense schmidts |
| 11:22.25 | Aurs | hehe |
| 11:22.30 | schmidts | yes thats true ;) but there havent been any updates latetly to 1.4 so i didnt notice it ;) |
| 11:31.26 | tzafrir | how do I mark part of a comment as "code" or whatever in the Jira? |
| 11:31.42 | schmidts | leifmadsen are there no private issues on jira anymore? i cant find this option in jira |
| 11:31.46 | tzafrir | (so it won't get mangled) |
| 11:32.00 | tzafrir | schmidts, "mark as private"? |
| 11:32.14 | tzafrir | err "viewable by all users" |
| 11:32.54 | schmidts | tzafrir i dont see this option when creating a new issue, is this option after creating it? |
| 11:33.18 | tzafrir | IIRC there is an option at the time of creating an issue. Not sure |
| 11:34.38 | schmidts | tzafrir havent found it, i have just created one issue to test but its not there |
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| 11:36.13 | schmidts | good morning kpfleming |
| 11:36.25 | kpfleming | morning indeed! |
| 11:37.23 | schmidts | kevin maybe you know how to create a private jira issue? i might be blind and dont see where i can set this ;) |
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| 11:38.19 | schmidts | tzafrir i can only set the viewable state for a comment |
| 11:40.46 | kpfleming | you need to have a bug marshal help you; regular users don't have permission to create private issues |
| 11:41.06 | schmidts | ah ok, thats new ;) |
| 11:41.10 | schmidts | thank you kevin! |
| 11:41.14 | kpfleming | if you have bug marshal privileges, then you can just set the 'security' field on the issue to 'Digium, Bug Marshals and Reporter' |
| 11:41.51 | kpfleming | yes, it's different from the way we had mantis setup |
| 11:43.25 | schmidts | btw i didnt remember which role i had in mantis, if it was bug marshal or something below but it looks like i dont have the same right in jira anymore (moving issues, deleting files..... |
| 11:46.21 | kpfleming | i will check |
| 11:46.45 | schmidts | ok thanks |
| 11:47.36 | kpfleming | i've updated your permissions, but it could take an hour for JIRA to notice the change |
| 11:48.44 | schmidts | ok thank you, russell said once he allready changed it ;) |
| 11:51.03 | kpfleming | it hadn't been done |
| 11:52.55 | schmidts | it looks like ;) |
| 12:07.21 | Aurs | perhaps he did, but JIRA didn't notice it? (ha-ha) :P |
| 12:10.06 | schmidts | maybe :D |
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| 12:28.05 | ahgindia | hello I am using Asterisk 1.6.2.13 version and facing a strange issue...when dialling an outbound call using an outbound trunk, the sip invite is sent after about 10 seconds when dial application is executed, so can anyone guide me what can be wrong here |
| 12:33.45 | kpfleming | ahgindia: please go to #asterisk... this channel is for development discussion |
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| 13:30.58 | JonathanRose | p3nguin: Commented to the issue. I think you must have just not been patching from a clean 1.8.7.0 or maybe you used the wrong -p level or something like that. Anyway, I tested the patch to make sure it applies to clean 1.8.7.0 checkouts and it does. |
| 13:31.34 | JonathanRose | I included detailed instructions in the comment. |
| 13:34.28 | bklang | kpfleming: I figured out my issue last night: the SDP being offered by Asterisk was so large that it was fragmenting the packet, and that's what was getting lost |
| 13:36.24 | kpfleming | oh wow |
| 13:36.44 | kpfleming | so in that situation, a *proper* SIP stack is supposed to switch to TCP, if possible |
| 13:39.16 | anonymouz666 | is there a proper SIP stack? :) |
| 13:40.15 | kpfleming | yes, there are quite a few |
| 13:40.46 | bklang | Yeah, I worked around it by limiting the codecs offered |
| 13:40.53 | kpfleming | sure, that makes sense |
| 13:41.14 | kpfleming | when offer both audio and video, and lots of codecs, and turn on SRTP... the SDP starts to get large pretty fast |
| 13:41.37 | bklang | yeah it does! |
| 13:41.47 | bklang | It was kinda exciting for a minute, until I realized that it was actually the problem |
| 13:47.32 | leifmadsen | :) |
| 13:47.43 | leifmadsen | wonder if we need to document that then... |
| 13:49.01 | pabelanger | who is transnexus? |
| 13:49.11 | russellb | a company? |
| 13:49.17 | pabelanger | re: svn commits for app_osplookup.c |
| 13:49.34 | pabelanger | looks like they are merging new features into 1.8 |
| 13:49.52 | anonymouz666 | they are osp addicted :) |
| 13:49.56 | pabelanger | plus used the merge scripts wrong |
| 13:50.14 | russellb | check repotools/authors for the email address to contact |
| 13:50.34 | russellb | they are indeed the OSP people. |
| 13:51.07 | pabelanger | support@transnexus.com |
| 13:51.27 | russellb | you can start there, it will get routed to the right person ... I can't remember the name of the programmer |
| 13:51.40 | pabelanger | k |
| 13:51.51 | leifmadsen | should the changes be reverted in the meantime? |
| 13:51.52 | russellb | Di-Shi Sun |
| 13:51.53 | russellb | I think |
| 13:52.13 | russellb | di-shi at transnexus dot com |
| 13:52.25 | leifmadsen | they also skipped the Asterisk 10 branch too |
| 13:52.29 | leifmadsen | so changes were only 1.8 and trunk |
| 13:52.38 | kpfleming | yeah... messy |
| 13:52.51 | leifmadsen | I think the changes should be reverted for now until someone can help them :) |
| 13:53.06 | kpfleming | they can be fixed up without having to revert them |
| 13:53.28 | leifmadsen | okie |
| 13:55.09 | pabelanger | Going to CC the -dev mailing list in the email. Basically getting them on asterisk-dev for help :D |
| 13:56.05 | pabelanger | bootstrap.sh was missed too |
| 13:58.30 | bklang | I've found a segfault in Asterisk 10 ast_translator_build_path() that is triggered by app_konference. Is that something that I should report to Asterisk or just to app_konference? |
| 13:58.45 | bklang | Honestly I'm not sure where the fault lies yet, it's a bit over my head |
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| 13:59.53 | kpfleming | well, i wouldn't think that an app should be able to trigger a segfault in an API call unless it's really doing something badly wrong |
| 14:00.17 | kpfleming | if you can determine that the API function is really at fault, it's certainly something we need to fix |
| 14:01.49 | bklang | #0 ast_translator_build_path (dst=0x186b5, src=0x186b5) |
| 14:02.39 | bklang | Yeah, I'll see if I can figure out who is at fault |
| 14:03.20 | kpfleming | whoah... if dst and src are the same, that's definitely not good |
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| 14:04.05 | kpfleming | i would not expect the function to operate properly in that situation, although a segfault is a pretty bad result |
| 14:04.22 | bklang | Here's the full backtrace https://gist.github.com/a506d964b77102a1b463 |
| 14:05.12 | bklang | looks like the immediate caller is app_konference |
| 14:05.34 | kpfleming | yep, sure does |
| 14:05.56 | bklang | There's a comment in member.c that reads "ast_translator_build_path() returns null if formats match" |
| 14:05.59 | bklang | looks like they're relying on that |
| 14:06.42 | kpfleming | heh... and i bet there's nothing documented that it should do that, but it may have done it for years :-) |
| 14:09.10 | bklang | Is that something you want to fix or do I need to pursue app_konference to change their code? |
| 14:10.23 | kpfleming | can't honestly say... it would need some investigation, so probably a JIRA issue is the right thing to do |
| 14:10.30 | bklang | Will do |
| 14:10.47 | kpfleming | but it seems like changing app_konference to not bother calling the function if dst == src would be very easy |
| 14:11.33 | kpfleming | of course... we'll strongly suggest using app_confbridge in asterisk 10 anyway :-) |
| 14:11.41 | bklang | Yeah, I may move to that now |
| 14:11.54 | bklang | I've been using app_konference only because it worked without Dahdi, which wasn't available on Solaris |
| 14:12.03 | file | wonders what the feature diff is between the two these days |
| 14:12.05 | kpfleming | right |
| 14:12.12 | bklang | and I found out just now that confbridge works just fine on Solaris |
| 14:12.12 | kpfleming | file: 800km |
| 14:13.28 | kpfleming | i assume it would have to rely on res_timing_pthread there |
| 14:13.51 | bklang | Haha i love how JIRA shows more issues resolved than opened. That's a killer close rate |
| 14:13.54 | kpfleming | maybe someone could write a solaris specific timing provider that is more efficient than pthread timers... |
| 14:14.45 | bklang | Yeah that would be really nice. However i'm quite sure you don't want me to attempt such a feat |
| 14:14.59 | Qwell | bklang: What's the worst that could happen? |
| 14:15.12 | bklang | Qwell: well, good point, it's already segfaulting :) |
| 14:15.56 | Qwell | what's up with POSIX timers? is that usable? |
| 14:21.00 | bklang | <PROTECTED> |
| 14:21.44 | bklang | ConfBridge is working |
| 14:22.03 | Qwell | no, I mean, POSIX timers as a new timer module |
| 14:22.21 | Qwell | If you're using it on Solaris, it's probably using pthread timers, which suck |
| 14:22.34 | Qwell | (suck compared to the better timer in Linux) |
| 14:22.49 | pabelanger | russellb: heh, just noticed the authors file still has your digium email address, |
| 14:23.44 | russellb | set to russell at russellbryant dot net i guess |
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| 14:25.39 | Qwell | russellb: you totally have the power |
| 14:27.52 | russellb | don't have it set up on the machine i'm using this minute, i'll get it later |
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| 14:28.20 | Qwell | russellb: I'm just messing with you. I'll do it :p |
| 14:28.47 | russellb | heh |
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| 15:01.08 | leifmadsen | Qwell: troll |
| 15:01.13 | leifmadsen | Qwell: <3 |
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| 16:20.17 | Corydon76-work | kpfleming: ping |
| 16:21.01 | kpfleming | Corydon76-work: pong |
| 16:21.25 | Corydon76-work | kpfleming: could you give me permission to commit to repotools, please? |
| 16:22.08 | kpfleming | done |
| 16:22.13 | Corydon76-work | TY |
| 16:27.16 | irroot | im creating a "private" svn server to help speed things up and save bw |
| 16:27.30 | irroot | it will be hosted here in south africa |
| 16:27.58 | irroot | im happy to mirror digium if you intrested |
| 16:28.56 | kpfleming | you can easily mirror repositories from svn.asterisk.org/svn.digium.com by using svnsync |
| 16:29.13 | kpfleming | but of course you can't commit to those repositories, as subversion is not a DVCS |
| 16:29.29 | kpfleming | svk is a way around that restriction, if you want to explore it. |
| 16:29.34 | irroot | yeah 100% read only or local write |
| 16:29.50 | irroot | im talking bout ftp mirror for releases more |
| 16:30.20 | irroot | the svn mirror is for my own purpose and those who use the patchset i maintain |
| 16:30.57 | irroot | we as a company/me are commited to asterisk as a solution and want to expand ofering |
| 16:31.22 | irroot | and are looking at supporting switchvox but the base is too small |
| 16:32.44 | kpfleming | ahh... we can provide you rsync so that you can mirror downloads.asterisk.org if you like |
| 16:32.52 | kpfleming | there is not 'ftp' :-) |
| 16:33.08 | irroot | yeah awesome |
| 16:33.25 | irroot | we will be setting up a hosting platform with the ITSP |
| 16:33.49 | irroot | who is putting there hat in the asterisk ring too supplying asterisk systems |
| 16:36.27 | tzanger | hm... architectural question... |
| 16:36.50 | tzanger | how "granular" should functions be? my gut says that a function should get/set one type of information |
| 16:38.40 | kpfleming | well, that really depends on how it will be used. if calling that function would always be followed by calling two more to do similar things, then it makes sense to combine them |
| 16:38.50 | kpfleming | that's really an 'instinct' thing |
| 16:39.08 | irroot | tzanger things like channel is multiplexed |
| 16:44.41 | tzanger | as an example, I've got a "behind the scenes" system that has its fingers in a number of different subsystems. would it be "better" to have a single function that interfaces to the backend or a number of them? The type of data to get/set is varied, but among them are available gateways, altitude, present and predicted location, gateway preferences configured by src extension, previous routing information... |
| 16:45.00 | kpfleming | that's really to vague to answer, i think |
| 16:45.01 | tzanger | would I be better off to create a resource that acts as the "access library" for the functions? |
| 16:45.15 | tzanger | sorry for the delay, wife needed printer ink changed |
| 16:47.14 | tzanger | the data from the functions would be used by the dialplan to build up to an eventual Dial() (or Playback/VoiceMail/etc.) |
| 16:58.03 | p3nguin | jonathanrose: Looking good so far! |
| 16:58.20 | JonathanRose | awesome |
| 17:00.54 | irroot | JonathanRose p3nguin nice job folks seen you been at it job well done |
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| 18:08.53 | rdegges | Hey guys, I was wondering if someone could help me confirm a bug I'm having in Asterisk 1.8.7.0 in regards to the usage of ChanSpy's X option? Or should I go to #asterisk with it? |
| 18:09.07 | Qwell | rdegges: #asterisk |
| 18:09.18 | rdegges | will do |
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| 19:27.26 | grEvenX | any devs around with correct role to make a non-public Jira ticket? |
| 19:30.44 | kpfleming | yes. why do you need one? |
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| 19:33.01 | p3nguin | Super secret bugs, I'd guess. |
| 19:35.10 | GreenFlash86 | hi guys, some questions about tls, don't know, if its a bug or no |
| 19:35.12 | GreenFlash86 | i have a connection with asterisk:50000 and client:50001 |
| 19:35.13 | GreenFlash86 | when asterisk qualifies my client over this connection, the via header still points to asterisk:50000. |
| 19:35.15 | GreenFlash86 | is this correct? |
| 19:36.03 | GreenFlash86 | sorry, edit: *still points to asterisk:5061 |
| 19:36.43 | kpfleming | how did a connection get made to asterisk:50000? |
| 19:37.20 | GreenFlash86 | ähm... dunno |
| 19:37.33 | kpfleming | then i'm really confused |
| 19:37.49 | kpfleming | is there a NAT device or something between asterisk and the client endpoint? |
| 19:38.06 | GreenFlash86 | nothing between, only LAN |
| 19:38.27 | GreenFlash86 | perhaps it's only the connection from asterisk to client? |
| 19:38.29 | kpfleming | then how are you seeing this 'connection' using these port numbers? |
| 19:38.38 | GreenFlash86 | networkmonitor |
| 19:39.11 | kpfleming | that's fine... asterisk doesn't listen on high port numbers, and when TCP connections are initiated by asterisk, they will have random port numbers |
| 19:39.37 | GreenFlash86 | sounds like an explanation |
| 19:39.41 | kpfleming | the port number in the Via header when the message is sent over a TCP or TCP/TLS connection is not very important... responses will be sent back over that same connection in most cases |
| 19:40.14 | GreenFlash86 | in most cases? or everytime? |
| 19:40.21 | kpfleming | in most cases |
| 19:40.31 | kpfleming | the SIP message rules for this sort of thing are complex |
| 19:40.54 | GreenFlash86 | aargh... not a very good day for me |
| 19:41.39 | grEvenX | kpfleming: some serious fucked up stuff that can harm any asterisk 1.4 server (DDOS attack with malicious formed SIP header content) |
| 19:41.59 | grEvenX | kpfleming: so I would think it'd be bad to make the details public before a fix is in place |
| 19:42.11 | GreenFlash86 | kpfleming: hmmm, let me think about that, will be back again in some minutes |
| 19:42.25 | kpfleming | grEvenX: security issues should be reported to security@asterisk.org |
| 19:42.47 | grEvenX | kpfleming: ah, thanks. I'll just send an e-mail to that address then? |
| 19:43.04 | kpfleming | yes please |
| 19:43.56 | grEvenX | will do right away, it should be pretty easy to reproduce, I'll even include my sipp scenario |
| 19:44.11 | kpfleming | thanks |
| 19:56.46 | GreenFlash86 | kpfleming: bak... very confusing |
| 19:57.38 | GreenFlash86 | i guess i will handle it like you told me, a response has to go back over the same connection |
| 19:58.03 | kpfleming | well, that depends on the situation |
| 19:58.18 | kpfleming | if you receive an INVITE, the response has to back to where the Contact header tells you it must go |
| 19:58.47 | kpfleming | SIP message processing and response rules do not lend themselves to simple statements that cover all cases :-) |
| 19:59.14 | GreenFlash86 | yes ok, but the response for the "option" request has to go back over the received connection? |
| 19:59.25 | kpfleming | no. OPTIONS is processed just like an INVITE. |
| 20:00.01 | GreenFlash86 | argh :(, ok, good to know then everything is right i guess |
| 20:00.19 | kpfleming | and SIP-over-TLS is astonishingly consfusing, to the point that even at SIPit events people can't get their implementations to interoperate very reliably |
| 20:00.25 | GreenFlash86 | so i need to find the problem at another place |
| 20:00.54 | mickecarlsson | asterisk 1.8.7.0, voicemail tagged as urgent (menu option 4) are not saved, any takers? |
| 20:03.11 | GreenFlash86 | kpfleming: yes, i know that :( and our mission is to use an older program and built in tls support |
| 20:03.51 | mickecarlsson | To duplicate: dial an extension that goes to voicemail, press # to start recoring a message, press # when finished, press 4 to mark it as urgent, press 1 to accept |
| 20:03.52 | GreenFlash86 | i will be here for some minutes, maybe i need your help again |
| 20:03.58 | mickecarlsson | voicemail is not saved |
| 20:04.25 | mickecarlsson | Onlcy clue: app_voicemail.c: -- Saving message as is |
| 20:04.28 | kpfleming | GreenFlash86: this is not an appropriate place to ask about SIP implementation.. this is for discussion of asterisk code. head to the sip-implementors mailing list. |
| 20:04.31 | mickecarlsson | Checking code now |
| 20:05.06 | GreenFlash86 | kpfleming: ok |
| 20:20.34 | mickecarlsson | False alarm re Urgent Voicemail |
| 20:20.52 | mickecarlsson | New folder created called Urgent, voicemails are stored there. |
| 20:20.57 | mickecarlsson | It pays to RTFM |
| 20:21.03 | mickecarlsson | blushes |
| 20:26.12 | grEvenX | kpfleming: sent |
| 20:30.53 | *** join/#asterisk-dev Jamuel (~Adium@c-67-180-156-186.hsd1.ca.comcast.net) |
| 20:33.19 | *** join/#asterisk-dev grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 20:33.28 | Jamuel | Hi. Was wondering if anyone knows of anyone taking a crack at backporting Reviewboard-1313 (Deadlocks dealing with dialplan hints during reload) https://issues.asterisk.org/jira/browse/ASTERISK-17666, https://issues.asterisk.org/jira/browse/ASTERISK-17760 for 1.6.2.20? |
| 20:33.29 | grEvenX | ran out of power, did I miss anything |
| 20:33.31 | DocAwesome | [ASTERISK-17760] [Status: Closed] [Assigned: rmudgett] [patch] deadlock in chan_sip - https://issues.asterisk.org/jira/browse/ASTERISK-17760 |
| 20:33.31 | DocAwesome | [ASTERISK-17666] [Status: Closed] [Assigned: rmudgett] [patch] Deadlock in ast_remove_hint Race with ao2_callback - https://issues.asterisk.org/jira/browse/ASTERISK-17666 |
| 20:35.05 | kpfleming | grEvenX: nope. your email has been received. |
| 20:36.43 | grEvenX | kpfleming: ok, good |
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| 23:38.55 | grEvenX | bedtime |
| 23:38.58 | grEvenX | gnight |