| 00:01.38 | Qwell | jmls_net: ignore it, basically |
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| 03:25.04 | gitguy | hi, i hear too much noise with speex codec, it doesn't happen with other codecs, i filled a bug here: http://bugs.digium.com/view.php?id=12653 |
| 03:25.08 | gitguy | thanks in advance |
| 03:25.35 | gitguy | (attached also a ogg file) |
| 03:26.12 | gitguy | the recorded file was recorded with Monitor() on the server (it has good internet speed) |
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| 16:43.19 | jpeeler | M11973 |
| 16:43.20 | MuffinMan | [assigned] [Asterisk] Applications/app_dial 0011973: The L(XXX) option does not work when the call is bridged. reported by falves11 (Karma: -4.75) http://bugs.digium.com/view.php?id=11973 |
| 16:43.37 | jpeeler | dare i ask if problems are still occurring? |
| 16:43.50 | file | the answer is probably yes |
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| 19:05.12 | jpeeler | if the reporter confirms that the problem is gone, do we just close it? |
| 19:05.22 | file | nods |
| 19:05.38 | jpeeler | yay |
| 19:13.38 | jpeeler | file: is the resolution filed under "fixed" if the problem just went away? |
| 19:13.45 | jpeeler | or no change required perhaps |
| 19:15.13 | file | or suspended |
| 19:15.45 | file | I would suspend with a note saying "I'm suspending this issue since you could not reproduce it with the current version, if this is still an issue in the future feel free to reopen." |
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| 20:14.59 | jpeeler | M12566 |
| 20:15.01 | MuffinMan | [feedback] [Asterisk] Channels/chan_sip/General 0012566: Loss of audio after 2 seconds of P2P RTP bridging reported by falves11 (Karma: -4.75) http://bugs.digium.com/view.php?id=12566 |
| 20:15.30 | jpeeler | it seems i'm having this problem right now |
| 20:15.52 | file | ooh how did you make it happen? |
| 20:15.54 | jpeeler | must go read up on packet2packet bridging |
| 20:16.19 | file | basically the frames never leave rtp.c, they come in via recvfrom, get minimally altered, and then get sent out using sendto |
| 20:16.29 | jpeeler | file: i don't know the network setup here, so i'm not sure |
| 20:16.49 | jpeeler | but it does only happen one way |
| 20:26.15 | jpeeler | file: when you say frames, you mean audio only right? |
| 20:26.41 | jpeeler | the SDP packets would still go through asterisk |
| 20:27.13 | file | yes, audio |
| 20:27.20 | file | SDP would still be handled in chan_sip |
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| 22:29.29 | jpeeler | M11826 |
| 22:29.31 | MuffinMan | [assigned] [Asterisk] Resources/res_features 0011826: Call transfer with 4 and more participants reported by aiker (Karma: neutral) http://bugs.digium.com/view.php?id=11826 |
| 22:29.57 | jpeeler | i need help understanding if the described scenario is proper "telephony" or not |
| 22:31.47 | jpeeler | because one could just tell the user to dial disconnect |
| 22:42.25 | putnopvut | Yeah, that's a situation that qwell and I looked into a while back, and we weren't sure what the "proper" behavior is. |
| 22:42.42 | Qwell | we did determine one thing though. crashing is not the proper behavior. |
| 22:43.42 | jpeeler | oh it's not crashing |
| 22:44.18 | jpeeler | interestingly if nothing is changed in features.conf, the defaults work since * is disconnect and *2 is attended transfer |
| 22:44.21 | putnopvut | I thought it was just leaving a lot of hung channels. |
| 22:45.24 | jpeeler | is it? i mean if it is then it should be fixed |
| 22:45.50 | jpeeler | you can do some interesting things though if the call is not hung up and stays on the line |
| 22:45.57 | jpeeler | i don't know what proper transfer behavior is |
| 22:46.40 | jpeeler | or rather more importantly, is it too much to ask the user if they want to disconnect to dial it rather than it happening automatically |
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