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| 15:20.41 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Many thanks to kpfleming for 7 years of awesome. |
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| 16:21.41 | file | grins |
| 16:21.47 | Qwell | O.o |
| 16:21.50 | file | I like local_queue_frame's possible return values |
| 16:24.58 | mjordan | sounds like a david vossel patch |
| 16:38.16 | russellb | file: may return 0, or a slightly more positive 0 |
| 16:38.27 | file | russellb, yes! |
| 16:38.30 | file | or a sad 0 |
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| 16:40.05 | mjordan | I'm disappointed in local_hangup. |
| 16:40.47 | russellb | whyfor |
| 16:41.48 | file | mjordan, you shouldn't stray from the yellow brick road |
| 16:42.02 | mjordan | it didn't check the return value. Clearly everyone else was doing it. Non-conformist. |
| 16:42.58 | flujan | hello guys, I am trying to write some code for asterisk and I am looking the function… Do you have some example of the function: ast_request_and_dial? The format, type and data that I need to pass it… I am calling from extension A to extension B. SIP/4000 to SIP/5000 |
| 16:44.14 | mjordan | flujan: that's used in several places. Have you looked at ccss or pbx? |
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| 16:46.25 | flujan | mjordan: no trying to see it in the app_dial. Will check the pbx. By pbx do you mean pbx.c right? |
| 16:48.31 | mjordan | ccss.c and pbx.c, yes. app_dial is a beast. If all you want to know is how to make an outgoing channel, that may not be the best example. But it sounds like you're also trying to bridge the outgoing channel with some other channel, which is a bit more extensive than just ast_request_and_dial. |
| 16:51.21 | flujan | mjordan: yeah i am trying to bridge two channels. Which example do you recommend? I am looking the res_features.c |
| 16:51.55 | mjordan | heh |
| 16:52.34 | mjordan | technically I'd recommend looking at the Dial API for dialing (see app_page and the calendar resources in Asterisk 11) and the Bridging API for bridging (see app_confbridge) |
| 16:52.41 | mjordan | but that might not be the answer you're looking for :-P |
| 16:52.56 | mjordan | is there a reason why you're looking to roll your own dial/bridge? |
| 16:54.18 | flujan | mjordan: yeah, I wanna to try a different approach in atx_transfer |
| 16:54.32 | flujan | I wanna to avoid channel masquerading. |
| 16:54.37 | mjordan | hm. |
| 16:55.07 | mjordan | that is a deep, dark, dangerous hole you're about to go down. You may want to get a firm grasp on how Asterisk does dialing and bridging before jumping into that chasm. |
| 16:55.29 | flujan | mjordan: yeah I know but probably I will learn a lot in the dark hole… lol |
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| 18:55.03 | Kobaz | is it a bug or design feature that IFTIME/GotoIfTime don't respect seconds |
| 18:56.18 | WIMPy | I cannot answer thatn question, but I think I have seen that documented somewhere. Probably voip-info. |
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| 21:09.14 | alecdavis | requesting a shipit for review 2129. I missed you all yesterday. It polishes off the corner cases with available agent hint on queues. |
| 21:19.05 | mjordan | alecdavis: why does a queue having a device state of unavailable imply that an agent is available in the queue? |
| 21:19.09 | mjordan | that seems kind of odd to me |
| 21:23.03 | leifmadsen | aye |
| 21:23.11 | leifmadsen | I would consider unavailable as not available |
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| 21:50.54 | sivadnz | @mjordan: The 'queue_avail' goal is to turn on the BLF (AST_DEVICE_INUSE) when either no agents are logged in, or none are available. |
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