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| 06:00.18 | schmidts | good morning all |
| 06:05.53 | snuff-work | ello |
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| 06:31.01 | schmidts | snuff still working or atleast again? |
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| 06:46.52 | snuff-work | yer i'm @ work.. |
| 06:51.01 | schmidts | how late is it at you? |
| 06:58.58 | snuff-work | its only 5pm |
| 06:59.27 | schmidts | ok i see ;) |
| 06:59.35 | schmidts | its 9 am over here |
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| 10:00.41 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- IAX2 is now RFC 5456. -=- AstriDevCon 2010, October 29, sign up here: http://bit.ly/astridevcon2010 |
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| 13:28.20 | schmidts | hello all |
| 13:34.05 | leifmadsen | schmidts: yo! |
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| 17:50.55 | jlpedrosa | Hi all, |
| 17:51.04 | jlpedrosa | i'd need some advice in code development... |
| 17:51.36 | jlpedrosa | anyone have a couple of spare minutes? |
| 17:53.12 | Kobaz | some minuteseses |
| 17:54.12 | jlpedrosa | thanks, i'm trying to implement that Dynamic features can by activated on both channels of a bridge... |
| 17:55.03 | jlpedrosa | problem is, it's executed on pbx thread, instead of channel thread, so you got delays... |
| 17:55.06 | Kobaz | that's already possible |
| 17:55.30 | jlpedrosa | how? you gotta choose activateOn self or peer, but can't be both... |
| 17:55.36 | Kobaz | yes it can |
| 17:55.50 | Kobaz | you set the dynamic features variable on a pickup macro |
| 17:55.57 | Kobaz | so then it sets it on the other end as well |
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| 17:56.46 | Kobaz | it should execute on the channel thread, I agree |
| 17:57.30 | jlpedrosa | you mean..? making a macro on channel A that makes a pickup to B channel and then executes Playback(tt-monkeys) ie? |
| 17:57.48 | Kobaz | no... like this: |
| 17:57.58 | Kobaz | Set(DYNAMIC_FEATURES=... |
| 17:58.21 | Kobaz | Dial(sip/foo,,U(pickup)) |
| 17:58.36 | Kobaz | context pickup: Set(DYNAMIC_FEATURES=...) |
| 17:58.59 | Kobaz | now dynamic features is enabled on both legs of the call |
| 17:59.42 | jlpedrosa | ... yep, but I afraid i'm not explaining my self... |
| 18:00.05 | jlpedrosa | SET(DYNA...) |
| 18:00.09 | jlpedrosa | DIAL(....) |
| 18:00.39 | jlpedrosa | then the user executes the dynamic feature (play sound ie) and it sould play on BOTH channels... |
| 18:00.48 | Kobaz | ah yes |
| 18:01.05 | jlpedrosa | did i make myself clear now? |
| 18:01.33 | Kobaz | the problem is... it's either caller, or callee execution |
| 18:01.41 | jlpedrosa | tha's it... |
| 18:01.43 | Kobaz | i've worked around that in dialplan... |
| 18:02.08 | Kobaz | you have a small short bit of dialplan that will do an originate to chanspy whisper, which will then play the track |
| 18:02.24 | Kobaz | the whisper will then play the track to both channels |
| 18:03.07 | jlpedrosa | would need to look originate opts, i need a lot of Dial application... anyway i'd like to put that patch, i think it's useful |
| 18:03.15 | Kobaz | but features should allow a 'both' option... that would be useful |
| 18:03.48 | jlpedrosa | is there any function that makes something like pbx_exec, but instead giving a channel, giving a bridge? |
| 18:03.57 | Kobaz | yes. a patch would be great... you can sign up to issues.asterisk.org... and also the -dev mailing list http://lists.digium.com/mailman/listinfo/asterisk-dev/ |
| 18:04.27 | Kobaz | not that I know of... but that doesn't mean it's not there |
| 18:04.31 | jlpedrosa | i'm already at issues, i've sent some patched in fact |
| 18:04.53 | Kobaz | best place to ask would be the -dev mailing list... or when some of the other people here wake up |
| 18:05.22 | jlpedrosa | any person in particular? |
| 18:05.32 | Kobaz | nothing comes to mind |
| 18:05.57 | jlpedrosa | ok, thanks for your time Kobaz |
| 18:05.58 | file | there is no such thing. |
| 18:06.00 | Kobaz | jpeeler, kpfleming, Qwell, jsmith, seanbright are all good |
| 18:06.29 | file | you can't execute an application on a bridge, the best you could do is front end it with a Local channel and use Meetme to do the bridge |
| 18:06.51 | jlpedrosa | well... I'd patch the code... |
| 18:08.08 | jlpedrosa | well, gotta go it's quite late here, thanks Kobaz |
| 18:08.12 | Kobaz | k |
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| 18:29.33 | russellb | Kobaz: i feel left out :-p |
| 18:30.22 | Kobaz | aw |
| 18:30.24 | Kobaz | whoops |
| 18:33.51 | russellb | everyone in here is good! |
| 18:34.08 | Qwell | I'm here. |
| 18:34.13 | Qwell | so, clearly that isn't the case |
| 18:37.13 | seanbright | well said |
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| 20:16.17 | philipp64|laptop | a few weeks ago there were some patches proposed to allow a dialplan to monitor call progress.... don't remember the details or if was eventually committed... but it got me thinking. if I forward a call out to a GSM provider (in this case AT&T) via my telco's SIP/SS7 gateway, how much information gets passed along through protocol conversion? |
| 20:17.16 | philipp64|laptop | specifically, will I get a separate indication that tells me when the call rolls over from trying to ring the handset, to getting forwarded to the Octel device? |
| 20:17.31 | kpfleming | nope |
| 20:17.46 | kpfleming | not unless the forwarding happens by redirection |
| 20:17.56 | kpfleming | but it's unlikely they'll communicate that over SIP anyway |
| 20:18.28 | philipp64|laptop | so I need to count out the interval between the RINGING indication and the earlier PRORGRESSING and stop the call before the forward implicitly happens? |
| 20:18.36 | kpfleming | i have no idea |
| 20:18.44 | kpfleming | that's not a development question |
| 20:19.20 | philipp64|laptop | well, I was trying to figure out what the new patches gets me in terms of functionality. |
| 20:20.05 | philipp64|laptop | I'll ask the protocol wonks on #asterisk what the implications are... |
| 20:20.23 | philipp64|laptop | anyone remember what the defect number was for that patch? |
| 20:24.21 | philipp64|laptop | didn't see it under Channels/chan_sip/* so wondering if it could be filed under something else? |
| 20:24.52 | leifmadsen | it could |
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| 20:36.41 | philipp64|laptop | what covers dialplan stuff? |
| 20:42.57 | leifmadsen | PBX/pbx_config |
| 20:43.33 | leifmadsen | the possibility exists for something to have been filed into the wrong location, so I'd just search the whole tracker, including closed items |
| 20:43.41 | russellb | leifmadsen: if by dialplan stuff, you mean the module that just parses extensions.conf ... |
| 20:44.13 | leifmadsen | I'm not sure what the issue really is as I came in only part was through |
| 20:51.04 | philipp64|laptop | leifmadsen: as I remember someone wanted to be able to add a flag/option to Dial() to allow them to have DIALSTATUS also reflect additional states, such as "RINGING" and "PROGRESSING". |
| 20:51.26 | philipp64|laptop | it might have been limited to SIP, however. not sure there was Q.931 support. |
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| 20:52.01 | leifmadsen | ya, not sure, I don't remember that issue |
| 20:53.38 | philipp64|laptop | I remember scratching my head and wondering how the transition would be handled, i.e. ${DIALSTATUS} changing from RINGING to ANSWERED, etc. |
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