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| 07:47.53 | schmidts | good morning |
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| 14:15.03 | pabelanger | anybody else think this looks suspicious? |
| 14:15.05 | pabelanger | WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c315317" |
| 14:15.55 | pabelanger | specifically the tab character within nonce |
| 14:16.49 | jsmith | pabelanger: Yes, that does look interesting |
| 14:16.58 | jsmith | would be more surprised if it were a vertical tab |
| 14:21.52 | pabelanger | Ya, I suspect that character is cause some authentication issue for some clients on 1.8. |
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| 14:28.08 | kpfleming | pabelanger: yes, that is broken |
| 14:29.28 | pabelanger | kpfleming: roger |
| 14:34.12 | jsmith | pabelanger: Is that tab in the nonce sent by Asterisk? |
| 14:34.24 | jsmith | pabelanger: If so, that's totally evil |
| 14:34.28 | kpfleming | yes. there was a report of this on asterisk-users as well. |
| 14:34.31 | kpfleming | it's very broken. |
| 14:34.36 | pabelanger | jsmith: Yar |
| 14:34.37 | kpfleming | however... i bet it's RFC compliant :-)0 |
| 14:34.38 | pabelanger | m18263 |
| 14:34.40 | MuffinMan | [new] [Asterisk] PBX/General 0018263: after update 1.6.2.2 to 1.8.0 registration from peers fails reported by pepesz https://issues.asterisk.org/view.php?id=18263 |
| 14:37.48 | jsmith | kpfleming: "It is recommended that this string be base64 or hexadecimal data." from RFC 2617 section 3.2.1 |
| 14:38.13 | jsmith | kpfleming: So, I think you're right -- the RFC doesn't say it MUST be base64 or hex |
| 14:40.32 | kpfleming | yeah.. but we still shouldn't have whitespace in it (shouldn't have whitesnake in it either) |
| 14:43.17 | sruffell | lol |
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| 21:46.27 | alecdavis | App_voicemail now broken in trunk, asks for box number and password, then abrutly drops out. |
| 21:46.52 | russellb | well that's not ideal behavior :-) |
| 21:48.13 | Kobaz | i concur |
| 21:48.16 | jsmith | Depends on how much you like getting voicemail :-) |
| 21:48.33 | jdoe | (sounds ideal to me) |
| 21:49.24 | alecdavis | M18240 |
| 21:49.26 | MuffinMan | [confirmed] [Asterisk] Applications/app_voicemail 0018240: [patch] VoicemailMain Exits Without Warning reported by leobrown https://issues.asterisk.org/view.php?id=18240 |
| 21:49.28 | russellb | yeah, i just changed my extension the other day to just ring forever and never go to voicemail. |
| 21:50.06 | russellb | so does this bug exist in all versions? |
| 21:50.50 | alecdavis | the existing bug report says yes, 1.4 1.6 1.8 ... |
| 21:51.11 | russellb | k, will assign to the committer of that change |
| 21:51.44 | alecdavis | no fireworks tonight for someone. |
| 21:51.49 | russellb | lol |
| 22:04.49 | alecdavis | I'm working on a callwaiting issue with FXS modules, I need isolate (mute) the FXS port while SAS+CAS signalling is going on, and then rejoin the conference. Otherwise we have the problem describe in M18129. |
| 22:04.51 | MuffinMan | [confirmed] [Asterisk] Channels/General 0018129: [patch] Oneway audio from SIP phone to FXS port after FXS port gets a CallWaiting pip reported by alecdavis https://issues.asterisk.org/view.php?id=18129 |
| 22:07.05 | WIMPy | So, whe a Dial() is executed, the channel is called to create a ne struct ast_channel, also supplying a pointer to the calling cahnnel. |
| 22:07.48 | WIMPy | So the channel calls ast_channel_alloc. But you'd want to copy much of the first channels data over to the new. |
| 22:08.15 | WIMPy | The only thing I found so far is clone_variables, but that actually moves. |
| 22:08.20 | WIMPy | What am I missing? |
| 22:10.15 | kpfleming | you *don't* want to copy very much of the first channel's data over. all the private data is only relevant to the first channel. |
| 22:11.00 | WIMPy | You certainly want to compy all of the caller and forwarding details. |
| 22:11.15 | WIMPy | That isn't that little any more. |
| 22:11.50 | kpfleming | no, you don't :-) |
| 22:12.08 | kpfleming | but some of that stuff is already copied, when appropriate, by the caller of ast_channel_alloc() |
| 22:12.12 | WIMPy | So what DO I want to do? |
| 22:12.29 | WIMPy | That's the part I'm talking about. |
| 22:12.29 | kpfleming | i don't know. you haven't stated a problem you are trying to resolve. |
| 22:13.42 | WIMPy | I just think that I must be missing something when writing more than a screen full of code to copy all that data "by hand". |
| 22:18.21 | alecdavis | SAS+CAS: SAS is the Subcriber Alert signal, 440HZ callwaiting beep, CAS is the CPE alert signal, then if the phone supports callwaiting-callerid it will send back a DTMF 'A' or 'D', mine sends back a 'D'. |
| 22:25.54 | alecdavis | The senario is Phone A (sip) is talking with Phone B (at FXS port), then phone C (sip) calls Phone B. Phone B hears 440hz as expected, so does Phone A (not quiet right), then asterisk sends CAS to Phone B (Phone A hears this), finally Phone B automatically sends back DTMF 'D', Phone A's voice path is now oneway A->B. |
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| 23:54.51 | alecdavis | unrelated: defrag your disk, tried this on a number of Windows PC's now, mydefrag.com, 17 seconds for XP on an old computer to start booting, till the login screen, used to take minutes. |