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| 07:47.53 | Justman | hi guys, i make a simple module, which allow to broadcast device states (here is it http://pastebin.org/1142290), but when i'm unloading module, i receive segfault, what can be wrong in this module ? thanks for any help |
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| 07:56.07 | schmidts | good morning |
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| 08:06.25 | *** join/#asterisk-dev lorenzom (~lorenzo@151.8.202.59) |
| 08:06.39 | lorenzom | hi all |
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| 08:09.28 | lorenzom | I'm a 3-year asterisk user and i'm interested in asterisk development |
| 08:09.42 | lorenzom | currently i'm part of a group of developers |
| 08:10.08 | lorenzom | and we |
| 08:10.11 | lorenzom | are interested |
| 08:10.31 | lorenzom | in building a new module (dialplan application) |
| 08:10.40 | lorenzom | for mrcp integration |
| 08:10.51 | lorenzom | MRCPv2 |
| 08:13.43 | lorenzom | does anyone know about any project regarding MRCPv2 modules? |
| 08:41.44 | schmidts | hello lorenzom, |
| 08:42.37 | schmidts | i have taken a short look at the rfc draft to mrcpv2 module and i think there would be much work to bring this into asterisk |
| 08:43.32 | schmidts | mrcpv2 run besides the sip stack, and have to ensure every change of the sip or sdp data in a call is recognized |
| 08:44.39 | schmidts | as far as i have read the rfc draft you want to use asterisk to control the mrcp sessions and also every sip changes or do you want to use asterisk also as the mrcpv media server? |
| 09:13.33 | lorenzom | hi schmidts |
| 09:14.11 | lorenzom | we would like to bring asterisk mrcp client capabilities |
| 09:15.47 | lorenzom | in order to acquire TTS and ASR funcionalities in asterisk |
| 09:16.49 | schmidts | ok i see |
| 09:17.12 | lorenzom | for example |
| 09:17.20 | schmidts | but i think there would be many changes to chan_sip.c to get all the information which are needed vor mrcp support |
| 09:17.27 | lorenzom | mmh |
| 09:19.25 | schmidts | s/vor/for/ |
| 09:19.41 | lorenzom | ;) |
| 09:22.43 | lorenzom | good observation |
| 09:23.13 | lorenzom | do you think that is not possible to write an additional module |
| 09:23.36 | lorenzom | mrcp client |
| 09:24.33 | lorenzom | able to simply control sip session without changing chan_sip? |
| 09:33.07 | schmidts | i didnt have read the rfc draft enough, but i have seen there is a condition that mrcp needs to get any changes on the call, like reinvites sdp changes ... |
| 09:33.14 | oej | what is mrcp? |
| 09:33.16 | oej | curious |
| 09:33.33 | schmidts | http://tools.ietf.org/html/draft-ietf-speechsc-mrcpv2-20 |
| 09:33.53 | schmidts | media resource control protocol |
| 09:34.17 | schmidts | in v2 it is used with sip and sdp |
| 09:36.01 | schmidts | MRCPv2 uses SIP and SDP to create the speech client/server dialog and |
| 09:36.01 | schmidts | <PROTECTED> |
| 09:36.01 | schmidts | <PROTECTED> |
| 09:36.01 | schmidts | <PROTECTED> |
| 09:36.48 | schmidts | IMHO this means mrcp app in asterisk has to control the sip and sdp which is sent out and received |
| 09:38.29 | oej | This is for integration with Nuance or something similar? |
| 09:39.06 | oej | For something like this we need to come up with a pbx core API, then add support in chan_sip. |
| 09:39.22 | schmidts | mrcp v1 was developed for nuance but v2 should be able with "plain" sip |
| 09:39.27 | oej | It propably needs to hook into the speech API we have today |
| 09:40.24 | lorenzom | We would like to use Loquendo MRCP server |
| 09:46.33 | schmidts | as i said not so easy ;) |
| 09:48.30 | lorenzom | :) |
| 09:52.16 | lorenzom | nothing is easy in our life |
| 09:52.20 | lorenzom | :) |
| 09:54.32 | lorenzom | however we are interested to support only MRCPv2 |
| 09:54.44 | lorenzom | so |
| 09:54.47 | lorenzom | as you said |
| 09:55.01 | lorenzom | things could be simpler if |
| 09:55.21 | lorenzom | s/could/should |
| 09:55.34 | lorenzom | s/could/should/ |
| 09:56.24 | lorenzom | MRCPv2 is really able to work with "plain" sip |
| 10:02.45 | schmidts | i dont know if asterisk would be the right place to use as a mrcp client maybe a plain uac would be better |
| 10:05.11 | lorenzom | mmh |
| 10:05.14 | lorenzom | do you mean |
| 10:05.26 | lorenzom | build an uac with mrcp client capabilities |
| 10:05.32 | lorenzom | and then using it |
| 10:05.38 | lorenzom | as a bridge |
| 10:05.58 | lorenzom | between asterisk and the mrcp server? |
| 10:05.59 | schmidts | it depends on what you want to do? |
| 10:06.10 | lorenzom | mainly TTS and ASR |
| 10:06.55 | schmidts | asterisk would only be the "router" in between which starts the call and handle sip messages from client to server but sdp should be directly between server and client |
| 10:07.49 | schmidts | maybe a real sip proxy like siprouter/kamilio/openser would be the right for this |
| 10:08.45 | schmidts | the problem i see is that a proxy has nothing to do with rtp streams just plain sip, but asterisk normaly stay in the media path for transcoding and other things |
| 10:10.21 | lorenzom | for example |
| 10:12.40 | schmidts | olle do you know the name of this opensource sip client? |
| 10:13.25 | schmidts | ah got it zoiper |
| 10:14.12 | schmidts | lorenzom you should have a look of this client, cause its completly open source so i think it would be easier to build a mrcp client with a real client instead of a uas system like asterisk |
| 10:15.19 | oej | back |
| 10:16.22 | oej | I don't think Zoiper is open source |
| 10:16.25 | oej | Ekiga is |
| 10:27.04 | schmidts | thx olle ;) |
| 10:59.16 | lorenzom | thx |
| 10:59.28 | lorenzom | i'm going to have my launch |
| 11:00.10 | lorenzom | see you later |
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| 11:18.03 | schmidts | cya |
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| 14:19.33 | The_Boy_Wonder | schmidts: ship it! |
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| 14:30.43 | schmidts | thank you, will be done in a moment ;) |
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| 14:43.03 | russellb | which review? |
| 14:45.15 | schmidts | 901 the remove of iteration over all dialogs to remove old subscribes |
| 14:45.52 | russellb | R901 |
| 14:45.55 | MuffinMan | [R901] handle_request_subscribe is too slow cause of iterating through all sip dialogs [Pending] Submitted by schmidts => https://reviewboard.asterisk.org/r/901/ |
| 14:46.09 | russellb | so you got the code in there to remove a subscription when we receive the bad event response? |
| 14:46.23 | schmidts | its very easy as i found ;) |
| 14:46.52 | schmidts | just set the pvt to needdestroy and the dialog_unlink_all function will clear the rest |
| 14:48.20 | russellb | I see handling of 481, but not 489 |
| 14:48.32 | russellb | I think a lot of phones respond with a 489 to a bad NOTIFY |
| 14:49.25 | schmidts | its not a bad notify, just bad call-id |
| 14:49.34 | russellb | yes, i know |
| 14:49.37 | schmidts | ok ;) |
| 14:49.45 | russellb | but that's what was happening before that original code went in |
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| 14:58.38 | schmidts | a 489 would by now only be handled in the default case for incoming packages but i am not sure if a notify response could have a owner, atleast no |
| 14:59.17 | schmidts | if not it would be treaded the same way i would do. |
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| 15:21.59 | schmidts | i have looked at the code in the revision you told me which this was inserted and i think a 489 was handled the same way as now, if there is no p owner the needdestroy flag is set |
| 15:22.30 | schmidts | maybe this was a problem cause there was no check for subscribes for this dialog, but i didnt see this by now |
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| 15:28.35 | schmidts | this check for subscribes on destroying a dialog was never in the 1.2 branch only 1.4 |
| 15:29.37 | schmidts | so i think this way was build into 1.2 and never changed since them, even its not necessary anymore |
| 15:34.34 | schmidts | s/them/then/ |
| 16:42.34 | kpfleming | pabelanger: ping |
| 16:59.55 | pabelanger | kpfleming: pong |
| 17:00.08 | kpfleming | never mind... i fixed the issue already :-) |
| 17:00.37 | pabelanger | Ya, just seen that. So it was spamming you? |
| 17:01.30 | kpfleming | i've just setup cron on all our server to send its output to a mailing list, and i've started watching it for potential problems |
| 17:05.40 | schmidts | russellb still around? |
| 17:22.45 | schmidts | wish you all a nice weekend! |
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| 20:37.49 | schmidts | hello |
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