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| 14:33.41 | russellb | mvanbaak: ping |
| 14:33.49 | russellb | mvanbaak: bbryant says that parsing issue should already be fixed |
| 14:33.52 | russellb | mvanbaak: using the latest stuff? |
| 14:34.19 | mvanbaak | yup |
| 14:34.32 | mvanbaak | just did a 'make update' to be sure |
| 14:35.06 | russellb | bbryant: ^^^ |
| 14:35.29 | mvanbaak | and the problem is on line 18371 of chan_sip.c in 1.6.0 |
| 14:35.30 | bbryant | mvanbaak: ok, i'll take a look at that |
| 14:35.58 | mvanbaak | it blindly assumes the input is: transport://hostname:port |
| 14:36.28 | mvanbaak | but with the register line you have transport://username:password@hostname:port |
| 14:36.40 | mvanbaak | so you cannot simply split on : |
| 14:36.52 | jsmith | Or transport://username:password:authuser@hostname:port/exten |
| 14:36.57 | mvanbaak | yeah |
| 14:37.02 | mvanbaak | I even have exten indeed |
| 14:37.33 | jsmith | Just making sure we don't break the authuser part |
| 14:37.36 | bbryant | apparently that fix didn't make it into 1.6 |
| 14:37.41 | jsmith | (since it's yet another colon) |
| 14:39.41 | mvanbaak | yup |
| 14:39.48 | mvanbaak | commit asterisk 127434 |
| 14:39.54 | mvanbaak | that one |
| 14:39.55 | MuffinMan | [127434] Committed by bbryant on 2008-07-02T17:27:36.590817Z: Fix to sip_parse_host so that it passes the correct information to sip_registry. |
| 14:40.30 | mvanbaak | can you merge it to 1.6 please ? |
| 14:40.37 | mvanbaak | my conferencing box is dead now ;) |
| 14:40.45 | bbryant | yep, i'm doing that now |
| 14:41.05 | mvanbaak | and I cant reach my mailbox to grab my svn certificate file from work |
| 14:41.15 | mvanbaak | bbryant: thanks |
| 14:42.06 | bbryant | mvanbaak: sorry for the trouble |
| 14:42.16 | bbryant | that change is merged now |
| 14:42.27 | mvanbaak | \o/ |
| 14:43.13 | mvanbaak | and no worries. it's me running a non-released beta version in production |
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| 14:45.51 | mvanbaak | and confirmed |
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| 16:52.55 | jpeeler | M11017 |
| 16:52.59 | MuffinMan | [assigned] [Asterisk] Channels/chan_zap 0011017: [patch] zap restart fails to generate channels reported by tzafrir (Karma: +98.00) http://bugs.digium.com/view.php?id=11017 |
| 16:53.19 | jpeeler | how far am i supposed to take this if Asterisk can't handle removal of dahdi spans? |
| 16:54.23 | jpeeler | seems like a major problems if reloading chan_dahdi (not restarting) doesn't even bring up the d-channels |
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| 17:01.33 | Seb7 | Hi. Is VLDTMF not working with MixMonitor a bug or feature of Asterisk 1.4? Emphasis on the Variable Length... |
| 17:03.04 | file | that depends... what does VLDTMF have to do with MixMonitor? |
| 17:06.22 | Seb7 | Er, well, if you have MixMonitor, the DTMF gets intercepted, queued, and regenerated as fixed length tones |
| 17:06.34 | Seb7 | Take out MixMonitor and it is variable length |
| 17:07.10 | Seb7 | Oh, and that's only on the Calling Party side. If the Called Party sends DTMF, you just get a blip in the audio, but no tones on 4 out of 5 calls (so far) |
| 17:07.21 | Seb7 | this is Zap on both sides BTW |
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| 17:22.51 | Seb7 | OK, a few ms, exact length seems random, does get through from the calling party. But it is much less than the 100 ms generated in the other direction. |
| 17:24.25 | Seb7 | Er, sorry, I meant called party |
| 17:25.08 | Seb7 | Also, a separate issue, but in zapata.conf, toneduration has no effect on the 100 ms generated by tones from the calling party. Another feature or bug? |
| 17:27.20 | Seb7 | So... basically, I have 3 issues here and I want to know whether just to report the Variable Length issue, because if that were fixed, the other two would be irrelevant or fixed in the process. |
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| 17:41.13 | Seb7 | is off for the night, but will read anything said to me in the morning. |
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