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| 13:46.19 | Charrit | Hi |
| 13:47.31 | Charrit | I'm having a Warning message on my fresh compiled asterisk v1.8.9.0 |
| 13:47.34 | Charrit | WARNING[11592]: chan_sip.c:28490 reload_config: No valid transports available, falling back to 'udp'. |
| 13:48.01 | Charrit | Do you know if anything was missing in the menuselect operation? |
| 13:49.53 | leifmadsen | Charrit: per the topic, this is not level 2 support. Please use #asterisk. |
| 13:51.04 | Charrit | Ok. Sorry |
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| 14:58.37 | seanbright | file: did you ever figure out that overlapping AST_AUDIOHOOK_TRIGGER overlap? |
| 14:58.54 | file | ummmm no, I don't think it should overlap |
| 14:59.26 | seanbright | alrighty |
| 14:59.52 | kpfleming | underlap? |
| 15:01.19 | leifmadsen | parallelap |
| 15:02.54 | seanbright | does changing those affect abi? |
| 15:03.02 | seanbright | i would assume yes |
| 15:09.12 | seanbright | http://pastebin.com/raw.php?i=7wraguEG |
| 15:09.30 | seanbright | so that will break existing modules, so trunk only i assume? |
| 15:10.12 | kpfleming | and so it begins... |
| 15:10.30 | seanbright | whats that? |
| 15:11.06 | kpfleming | you must be living under a rock |
| 15:11.18 | seanbright | ok |
| 15:11.28 | kpfleming | http://www.facebook.com/l.php?u=http%3A%2F%2Fwww.youtube.com%2Fwatch%3Fv%3D0SG4e1Y03vI&h=VAQE1_2M- |
| 15:11.33 | kpfleming | darn facebook |
| 15:11.47 | kpfleming | http://www.youtube.com/watch?v=0SG4e1Y03vI |
| 15:11.58 | seanbright | interesting |
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| 15:12.24 | seanbright | not sure how that has anything to do with living under a rock |
| 15:12.39 | seanbright | or how that changes my question about ABI with the audiohooks patch i just linked to |
| 15:12.44 | kpfleming | it doesn't |
| 15:12.48 | kpfleming | i'm just trolling |
| 15:12.51 | seanbright | got it |
| 15:12.58 | leifmadsen | kpfleming: you're a regular russellb |
| 15:13.14 | mjordan | seanbright: I think you could view that as a bug, honestly. Just might not be a bug anyone has experienced (or realized they experienced it) |
| 15:13.32 | kpfleming | mjordan: agreed |
| 15:13.45 | kpfleming | if the ABI is broken, it can't be used as currently defined. correcting it doesn't qualify as an 'ABI change'. |
| 15:14.01 | kpfleming | leifmadsen: i can only aspire to be a regular russellb |
| 15:17.44 | seanbright | but if someone has asterisk modules for 1.8/10 that use any of the flags above AST_AUDIOHOOK_WANTS_DTMF (all of which are getting shifted over an extra bit), won't that break those modules? |
| 15:18.02 | kpfleming | yes, it would. |
| 15:18.13 | kpfleming | until they are recompiled, of course |
| 15:18.17 | seanbright | right |
| 15:18.35 | kpfleming | which bits are currently overlapping? |
| 15:19.03 | seanbright | AST_AUDIOHOOK_WANTS_DTMF and AST_AUDIOHOOK_TRIGGER_WRITE |
| 15:19.20 | kpfleming | hmm |
| 15:19.40 | kpfleming | are either of those used in the asterisk tree now? |
| 15:20.22 | seanbright | AST_AUDIOHOOK_WANTS_DTMF is used by func_volume |
| 15:21.05 | seanbright | _WRITE is not set by anything explicitly, but it is set when AST_AUDIOHOOK_TRIGGER_MODE is set |
| 15:22.05 | seanbright | although AST_AUDIOHOOK_TRIGGER_MODE is never set explicitly either |
| 15:22.39 | kpfleming | then i would say move TRIGGER_WRITE out of the way, and leave all the other ones with the current values |
| 15:22.44 | kpfleming | that minimizes the disruption |
| 15:23.19 | seanbright | alrighty |
| 15:25.57 | seanbright | thoughts? http://pastebin.com/raw.php?i=3mB85icG |
| 15:26.20 | seanbright | in trunk i will just reorganize them all |
| 15:27.31 | kpfleming | yeah, that would work. a comment stating why the bits aren't adjacent would be good for future readers of the code. |
| 15:28.29 | seanbright | will do |
| 15:37.49 | seanbright | do these phones run asterisk? |
| 15:38.07 | file | do you want them to run asterisk? |
| 15:38.14 | mjordan | they probably could |
| 15:38.25 | kpfleming | they could, but would not benefit from doing os |
| 15:38.28 | kpfleming | s/os/so/ |
| 15:38.51 | Qwell | seanbright: They do have space invaders though! |
| 15:38.53 | Qwell | not really. |
| 15:38.55 | Qwell | but they could! |
| 15:39.39 | seanbright | just curious |
| 15:39.45 | *** join/#asterisk-dev dms (~dms@nat/digium/x-avharpcnyycvsooo) |
| 15:39.51 | seanbright | do they speak IAX or SIP or both or neither? |
| 15:39.56 | seanbright | or Ice |
| 15:39.57 | russellb | SIP |
| 15:39.57 | seanbright | heh |
| 15:39.58 | kpfleming | SIP only |
| 15:40.08 | seanbright | cool |
| 15:40.13 | kpfleming | no RTP, just SIP. we do all the media over SIP. wait, no, that's not right. |
| 15:40.25 | russellb | that was my understanding |
| 15:40.35 | kpfleming | media over ICMP, yeah, that's it. |
| 15:40.35 | russellb | SIP session per media frame |
| 15:40.38 | seanbright | makes sense to me |
| 15:40.47 | jsmith | russellb: SIP over RTP? |
| 15:40.53 | jsmith | gasps |
| 15:41.11 | pabelanger | waits for the tin can integration |
| 15:41.13 | jsmith | Why not just move to SCTP?!? |
| 15:41.16 | jsmith | stirs the pot |
| 15:41.18 | file | everything is compressed using bzip2 and sent over a single TCP socket |
| 15:41.29 | russellb | file: all phones use the same tcp socket? |
| 15:41.35 | file | russellb, yes! |
| 15:41.41 | file | we couldn't afford to pay for more than one ;( |
| 15:41.43 | russellb | amazing technology |
| 15:41.49 | file | tcp sockets cost $$$$$$$ |
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| 15:42.48 | russellb | how about support for sending a sha256 of the whole conversation instead of RTP |
| 15:43.10 | Qwell | russellb: yes, but it sends it as an md5 of the sha256 |
| 15:43.19 | russellb | that's critical for bandwidth constrained but CPU and storage plentiful environments |
| 15:43.45 | russellb | Qwell: well that's clearly an improvement, i like the innovation i'm seeing here |
| 15:44.08 | russellb | but seriously, the phones are really cool :) |
| 15:44.39 | kpfleming | i have a patent on the sha-256 method of store-and-forward conversations. you now owe me US$1 for mentioning it in public. |
| 15:44.50 | russellb | FFFFF |
| 15:45.51 | file | kpfleming, I have a patent on mentioning you have a patent in #asterisk-dev - you now owe me US$2 |
| 15:45.57 | kpfleming | crap |
| 15:46.11 | kpfleming | russellb: pay my US$1 to file |
| 15:46.13 | russellb | kpfleming: you should tweet that Digium has introduced the most bandwidth efficient HD Voice ever with sha-256 based store-and-forward technology |
| 15:46.21 | kpfleming | russellb: in April :-) |
| 15:46.31 | russellb | s/introduced/announced/ |
| 15:46.32 | Qwell | file: I have a patent on accepting payment from a third party over a patent. You owe me $3. |
| 15:46.36 | russellb | whatevs |
| 15:47.04 | kpfleming | welcome to the craziness that is #asterisk-dev |
| 15:47.41 | russellb | song and dance time |
| 15:47.49 | russellb | take it away, file ! |
| 15:47.58 | file | russellb, that'll be $19.95 |
| 15:48.06 | kpfleming | i wonder if malcolm is planning to bring any phones to FOSDEM |
| 15:48.25 | russellb | i'm bring a 12 pound box of stickers |
| 15:48.28 | russellb | so .. many .. stickers |
| 15:48.40 | Qwell | russellb: You should replace the stickers with chocolate. |
| 15:48.51 | russellb | Qwell: that's for the return trip |
| 15:48.57 | kpfleming | Qwell: it's belgium, you don't bring chocolate, you take chocolate. |
| 15:49.11 | russellb | just wants waffles |
| 15:49.13 | Qwell | I bet they would love the american chocolate there. |
| 15:49.30 | Qwell | we love theirs, so the reverse most also be true |
| 15:50.44 | kpfleming | right, just like american beer. they love it there. |
| 15:51.42 | russellb | on a recent trip to Ireland, everyone was drinking Guiness except one of the guys who lives there, who was drinking Budweiser |
| 15:52.09 | kpfleming | real Budweiser from the Czech Republic, i bet |
| 15:57.20 | Qwell | seanbright: 1<<6? |
| 15:57.51 | Qwell | oh, you moved write not wants_dtmf. |
| 15:58.01 | kpfleming | yep |
| 16:02.09 | seanbright | Qwell: yes. it's confusing. |
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| 16:19.12 | tzafrir_laptop | Sadly still no IP phone with a free firmware |
| 16:19.56 | tzafrir_laptop | (and yes, this applies to everyone. Also to each and every phone endorsed by us, which has very clear GPL violations) |
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| 16:28.23 | CarelessChaser | Hi All! How I can through AMI take information about current calls with uniqueid that will be write in cdr? |
| 16:33.48 | CarelessChaser | with "Action CoreShowChannels" I can take info about each channel, but there only fields Uniqueid and BridgedUniqueID, and I do not know which of them will be placed in cdr |
| 16:34.45 | pabelanger | CarelessChaser: #asterisk |
| 16:35.06 | pabelanger | tzafrir_laptop: Didn't your company just announce an IP phone too? |
| 16:36.36 | tzafrir_laptop | I wasn't sure if it's "endorse", "promote" or "provide" |
| 16:36.47 | CarelessChaser | thanx |
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| 19:56.51 | otherwiseguy | grr. looks like trunk's optional API stuff is broken on Lion: dyld: Symbol not found: ___ast_adsi_available |
| 19:57.13 | kpfleming | no surprise. |
| 19:57.19 | kpfleming | ______ is broken on Lion. |
| 19:57.21 | otherwiseguy | In addition to the libasteriskssl stuff being broken: install: libasteriskssl.dylib.1: No such file or directory |
| 19:57.38 | otherwiseguy | It *used* to work. :-) |
| 19:57.53 | otherwiseguy | blah blah, unsupported, blah blah. :-) |
| 19:58.33 | *** part/#asterisk-dev CarelessChaser (~chaser@46.63.53.135) |
| 20:17.07 | pabelanger | otherwiseguy: heh, funny you say that. I think I'm seeing the same problems under ubuntu precise (12.04) and 1.8 branch |
| 20:18.37 | pabelanger | undefined symbol: ast_aes_set_encrypt_key because res_crypto was not compiled |
| 20:20.01 | pabelanger | http://pastebin.com/9Q9hRR0A |
| 20:20.28 | otherwiseguy | pabelanger: In this example, res_adsi was actually compiled and installed. |
| 20:42.03 | *** join/#asterisk-dev dvossel (~dvossel@user-69-73-31-221.knology.net) |
| 21:00.44 | putnopvut | All right everyone, it's time to play: "Is this a regression?" Today's candidate comes from issues ASTERISK-16959 and ASTERISK-19201. The raised issue is that for manager.conf and http.conf, there once was an option called "tlsbindport" that would allow for one to specify the port to bind to for TLS connections. My research shows that this option was removed by the IPv6 merge prior to the creation of Asterisk 1.8.0. Howeve |
| 21:00.44 | putnopvut | r, sample config files were not updated. In 1.6.2, there is no "tlsbindport" option, but there is an "sslbindport" option. So my question is, is this a regression that should be fixed by reintroducing the tlsbindport option in 1.8, or should we simply remove the option from sample config files and indicate that the port should be indicated as part of the tlsbindaddr option? |
| 21:04.05 | *** join/#asterisk-dev moy (~moy@173.239.155.74) |
| 21:05.38 | leifmadsen | putnopvut: is functionality removed, or just the way in which the functionality is applied? |
| 21:06.27 | leifmadsen | if the latter, then just update the sample files, UPGRADE.txt and CHANGES. (Perhaps a notice/warning be implemented indicating the functionality is now done via <foo>) |
| 21:06.33 | leifmadsen | IMHO |
| 21:07.20 | otherwiseguy | yeah, isn't it now tlsbindaddr=host:port or something? |
| 21:08.15 | leifmadsen | like tcpbindaddr and udpbindaddr in sip.conf right? |
| 21:08.26 | putnopvut | yes, the functionality is there, it's just configured differently. |
| 21:08.40 | otherwiseguy | I say just update the documentation. |
| 21:08.44 | leifmadsen | putnopvut: then ya, I suggest documentation changes is appropriate |
| 21:08.48 | putnopvut | Okay, sounds good to me. |
| 21:08.55 | leifmadsen | don't forget UPGRADE.txt and CHANGES |
| 21:09.02 | leifmadsen | basically it was missed unfortunately |
| 21:09.10 | otherwiseguy | blames leifmadsen |
| 21:09.20 | leifmadsen | everyone needs a scapegoat |
| 21:09.26 | otherwiseguy | especially me! |
| 21:09.40 | otherwiseguy | Of course, I also need cake. |
| 21:09.49 | leifmadsen | agreed |
| 21:09.52 | leifmadsen | I have cake in the kitchen |
| 21:19.50 | kpfleming | putnopvut: i agree with the consensus here |
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| 22:06.43 | seanbright | pabelanger: you are responsible for the dahdi debs for ubuntu? |
| 22:07.15 | seanbright | if so: any thoughts about a dahdi-dkms package? |
| 22:15.51 | leifmadsen | seanbright: I know he has thought about it, but hasn't gotten that far |
| 22:16.02 | leifmadsen | I'm looking at doing the same thing for DAHDI RPMs (locally) |
| 22:17.39 | *** join/#asterisk-dev otherwiseguy (~otherwise@rrcs-173-197-0-67.west.biz.rr.com) |
| 22:22.34 | seanbright | leifmadsen: cooool |
| 22:22.48 | leifmadsen | if I figure it out I'll pass it back upstream |
| 22:27.44 | WIMPy | Anyone here from Digium feeling responsible for product manuals? |
| 22:29.14 | leifmadsen | WIMPy: I'm going to guess you likely want http://www.digium.com/en/company/contact/ |
| 22:32.34 | otherwiseguy | So, somebody help me out here. Reading the sample config, it looks like the callbackextension option (or adding one via a register= line) should force any call that is matched by that peer/registration to land at the aforementioned extension. From any tests I have ever done, it looks like really just send calls to the extension specified in the RURI whether the peer we match has a callbackextension or not. What are we suppos |
| 22:32.34 | otherwiseguy | to be doing? |
| 22:33.32 | WIMPy | Is "technical support" bidirectional? :-) |
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| 22:36.45 | jsmith | otherwiseguy: My understanding (based on the sample configs) matches yours. |
| 22:38.13 | WIMPy | callbackextension is a replacement for a register =>... |
| 22:38.22 | pabelanger | seanbright: no, but ubuntu already has DKMS support for DAHDI packages |
| 22:38.51 | putnopvut | otherwiseguy: I'll have to read up on it a bit. |
| 22:39.56 | otherwiseguy | It seems like callbackextension sets the Contact header to include the extension, so by default if you are registering to another Asterisk box and they do Dial(SIP/asteriskboxwithcallbackextension) it will default to making the RURI the callbackextension, but you can override that with Dial(SIP/asteriskboxwith…/${other_extension}) |
| 22:41.48 | putnopvut | Yeah, I would have expected the Contact header setting behavior like you mentioned. jeez... |
| 22:41.50 | otherwiseguy | WIMPy: right. and callbackextension should just match whatever you would do when you do register => user:pass@peer/callbackextension. So I would assume that both just do something that seems different from what I expected--send the Registration with a contact header of the callbackextension so that the other side (might) dial that particular extension. |
| 22:42.46 | otherwiseguy | is looking at ASTERISK-13456 |
| 22:43.55 | otherwiseguy | ASTERISK-13456 basically tries to add peer matching by callbackextension if you define multiple peers with the same host/port but different callbackextensions…which seems kind of weird. |
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| 22:47.29 | otherwiseguy | But, if callbackextension is registering to a peer and changing the Contact header, I can kind of see why someone would want it. But I tend to think of peers as unique hosts and handle calls from the host as matching the single peer and differentiating solely in the dialplan for different extensions called from that peer. |
| 22:56.13 | dvossel | omg callbackextension |
| 22:56.19 | dvossel | do not want |
| 22:56.22 | *** part/#asterisk-dev dvossel (~dvossel@user-69-73-31-221.knology.net) |
| 23:00.43 | russellb | otherwiseguy: you scared him off |
| 23:04.38 | kpfleming | boo |
| 23:05.03 | kpfleming | tomorrow.. flying! |
| 23:05.29 | file | on a bird? |
| 23:05.36 | kpfleming | platypus |
| 23:05.37 | otherwiseguy | russellb: I am a bad man. |
| 23:05.44 | file | Perry! |
| 23:06.00 | kpfleming | file: are you now addicted? |
| 23:06.03 | file | yes. |
| 23:06.14 | kpfleming | that's bad, because you work from home |
| 23:06.41 | file | Phineas and Ferb know no boundaries! |
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