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| 11:02.40 | jmls | morning all |
| 11:02.54 | jmls | is there anyway of delaying variable expansion in global variables ? |
| 11:03.58 | jmls | for example, I want to define CC=activecallguid=${ACTIVECALLGUID} in [globals] and then use ${CC} in my dialplan, and have ${ACTIVECALLGUID} filled in at that point in the dialplan |
| 11:04.56 | jmls | however, the above syntax defines CC=activecallguid= because ${ACTIVECALLGUID} is not yet defined |
| 11:07.06 | jmls | hmm, nevermind, just use a gosub instead. |
| 11:07.30 | jmls | sometimes the solution presents itself just as you make yourself look like a twat. ;) |
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| 13:37.21 | pabelanger | morning! |
| 13:37.35 | pabelanger | Finally made it back to Ottawa |
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| 15:18.52 | kam187 | hey guys |
| 15:20.05 | kam187 | got a wierd issue where asterisk is not always getting voice frames from my channel driver |
| 15:20.13 | kam187 | i can see i'm generating them |
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| 15:22.53 | kam187 | [Jun 1 16:21:05] DEBUG[28254]: rtp.c:2897 ast_rtp_write: Ooh, format changed from unknown to g729 |
| 15:22.53 | kam187 | [Jun 1 16:21:05] DEBUG[28254]: rtp.c:2914 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 |
| 15:23.03 | kam187 | this message appears when its working and isnt there when its not |
| 15:23.08 | kam187 | not sure whats going in |
| 15:23.10 | kam187 | *on |
| 15:23.13 | kam187 | any ideas? |
| 15:23.53 | Corydon76-dig | kam187: is your driver open source? |
| 15:24.00 | kam187 | its based on chan_sebi |
| 15:24.08 | Corydon76-dig | Have no idea what that is |
| 15:24.29 | kam187 | its for GSM usb modem data sticks that support voice |
| 15:25.09 | kam187 | i assume asterisk normally calls the sub in here |
| 15:25.09 | kam187 | static const struct ast_channel_tech modem_tech = { |
| 15:25.17 | kam187 | <PROTECTED> |
| 15:25.17 | kam187 | <PROTECTED> |
| 15:25.29 | kam187 | periodically to check for frames/send frames |
| 15:25.30 | Corydon76-dig | Is it a master channel or a slave channel? |
| 15:25.40 | kam187 | erm whats the difference? |
| 15:26.00 | Corydon76-dig | i.e. is it starting the PBX or is it getting called by something like Dial? |
| 15:26.11 | kam187 | its being called by dial |
| 15:26.54 | kam187 | i stuck some debugs in those two subs (modem_ream/modem_write) and they arent being called when voice doesnt work |
| 15:26.59 | Corydon76-dig | Then your master channel is either not getting bridged or it's hanging out in some routine and not servicing the channel |
| 15:27.10 | kam187 | oh |
| 15:27.24 | kam187 | its just sip -> my channel driver tho |
| 15:27.32 | kam187 | and nothing special going on, just a normal dial |
| 15:27.58 | kam187 | (asterisk-1.4.29) |
| 15:28.26 | kam187 | any ideas on how i can debug it? |
| 15:28.38 | Corydon76-dig | Nope, never written a channel driver before |
| 15:28.47 | kam187 | doh |
| 15:29.55 | kam187 | i would think sip is super stable on 1.4.29 |
| 15:32.46 | moy | kam187: so the call is bridged at some point? |
| 15:33.48 | kam187 | yeah, its very random. say 1 in 10 calls works |
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| 15:36.14 | kam187 | i'm just trying to find the working log/non working log to show u the difference |
| 15:38.57 | moy | also I did not see a straight answer to what you were asked about the driver being open source ( and if it is, where is the source code? ) |
| 15:39.56 | kam187 | its based on this |
| 15:39.57 | kam187 | http://asterisk-es-rsp.irontec.com/websvn/filedetails.php?repname=asterisk-es-rsp&path=%2Fteam%2FOdicha%2Fasterisk-addons-1.4.7%2Fchannels%2Fchan_sebi.c |
| 15:40.27 | kam187 | its exactly the same (right now) except i added a few better option in show devices - like the reception level etc - but those shouldnt affect anything |
| 15:41.37 | kam187 | http://pastebin.ca/1875443 |
| 15:41.42 | kam187 | ^^ working/not working log |
| 15:42.05 | kam187 | those 'ERROR's are my debug lines printing weather it entered modem_read or modem_write to get/send voice |
| 15:46.43 | kam187 | i just checked modem_devicestate: and it returns '2' in both cases, so its not getting hung up there |
| 15:57.10 | kam187 | very strange |
| 16:01.20 | kam187 | hmm it might be a wierd NAT issue i think |
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| 16:06.12 | kam187 | hmm looks like it is. the way that driver is coded, it needs a frame written before it'll read one |
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| 16:13.25 | kam187 | ugh |
| 16:13.32 | kam187 | rtp port forwarding issue |
| 16:13.34 | kam187 | go figure |
| 16:15.04 | kam187 | i did a make clean at some point and rtp.conf got reset to default |
| 16:17.12 | leifmadsen | only a make samples would do that |
| 16:20.56 | kam187 | yeah i tried 1.6.1 at one point so moved etc/asterisk and then did a make samples with 1.6.1 |
| 16:21.04 | leifmadsen | there ya go :) |
| 16:23.17 | kam187 | hmm still doing the same thing |
| 16:23.22 | kam187 | but direct on local lan it works |
| 16:23.26 | kam187 | so its got to be a routing issue |
| 16:50.12 | pabelanger | m15996 |
| 16:50.15 | MuffinMan | [ready for review] [Asterisk] Resources/res_smdi 0015996: [patch] change the order for ACF registration for SMDI reported by junky https://issues.asterisk.org/view.php?id=15996 |
| 16:50.45 | devmod | Was video-on-hold ever merged into 1.6? |
| 16:50.47 | pabelanger | A trivial patch that looks to be ready to merge. If anybody cares to sign off |
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| 17:40.02 | kam187 | hmm now i got an rtp alignment problem |
| 17:40.05 | kam187 | works fine locally |
| 17:46.25 | *** join/#asterisk-dev ManxPower (~manxpower@216.186.151.147) |
| 17:46.44 | ManxPower | Either I'm missing something in the docs or the "o" option to dial is no longer documented in 1.4 |
| 17:49.42 | kam187 | hmm what does Got RTCP report of 200 bytes mean? |
| 17:49.58 | leifmadsen | ManxPower: you mean this? |
| 17:49.59 | leifmadsen | " o - Specify that the CallerID that was present on the *calling* channel\n" |
| 17:49.59 | leifmadsen | " be set as the CallerID on the *called* channel. This was the\n" |
| 17:49.59 | leifmadsen | " behavior of Asterisk 1.0 and earlier.\n" |
| 17:50.31 | kam187 | oh nvm |
| 17:52.56 | ManxPower | leifmadsen, is that from 1.4.x? |
| 17:53.09 | ManxPower | because it is NOT listed in our 1.4.x |
| 17:53.12 | leifmadsen | ManxPower: yes, directly from the branch code for 1.4, checked out several seconds ago |
| 17:53.26 | leifmadsen | 1.4.x what? |
| 17:53.31 | ManxPower | leifmadsen, maybe the one patch we have to app_dial removed it. |
| 17:53.32 | ManxPower | checking |
| 17:53.39 | leifmadsen | sounds more likely |
| 17:53.58 | ManxPower | Asterisk 1.4.26.2 |
| 17:57.09 | leifmadsen | ManxPower: when I checkout the 1.4.26.2 tag it exists |
| 17:57.15 | leifmadsen | must be a local issue |
| 17:58.02 | leifmadsen | line 179 of apps/app_dial.c in tags/1.4.26.2 |
| 18:01.13 | tzafrir_laptop | http://bugs.debian.org/cgi-bin/433779#105 - should this be considered a bug of asterisk? |
| 18:01.24 | leifmadsen | URL not found? |
| 18:01.47 | tzafrir_laptop | sorry: http://bugs.debian.org/433779#105 |
| 18:02.18 | tzafrir_laptop | "it appears that asterisk is incorrectly treating DNS resolution failure during SIP registration as a "404 not found" SIP response." |
| 18:02.41 | tzafrir_laptop | It is labeled as "missing-dependency" |
| 18:05.07 | tzafrir_laptop | But the DNS may be temporarily unavaialble on other times |
| 18:11.48 | ManxPower | leifmadsen, weird. Thank you for taking the trouble to check, I appreciate it. |
| 18:12.18 | leifmadsen | tzafrir_laptop: hmmm, not sure if that is expected behaviour or not -- might be worth it to file an issue in mantis |
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| 20:36.31 | kam187 | anyone here made a channel driver before? |
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| 20:47.14 | Kobaz | mm |
| 20:47.22 | Kobaz | more bugs |
| 20:48.21 | kam187 | yum |
| 20:48.22 | kam187 | :P |
| 20:48.50 | Kobaz | what module gives you the 'dialplan *' commands |
| 20:49.59 | leifmadsen | pbx_config.so I believe |
| 20:53.13 | Kobaz | https://issues.asterisk.org/view.php?id=17440 |
| 20:53.14 | Kobaz | there it be |
| 21:05.56 | Kobaz | it's a shame that 1.6.0 is security fixes only now |
| 21:06.17 | Kobaz | 1.6.2 is just too buggy for me to use in it's current state |
| 21:08.56 | Kobaz | there's some deadlock with tab completion with the dialplan commands that I can't quite narrow down |
| 21:09.01 | Kobaz | it completely kills asterisk |
| 21:09.16 | pabelanger | Kobaz: I can't reproduce your issue with 1.6.2 branch |
| 21:09.26 | Kobaz | hmm |
| 21:09.32 | pabelanger | Kobaz: can you post your extensions.ael file |
| 21:09.52 | Kobaz | it's empty |
| 21:09.56 | Kobaz | well |
| 21:10.01 | Kobaz | right now it's empty |
| 21:10.06 | Kobaz | and it still crashes |
| 21:10.43 | pabelanger | ok, pastebin your s@foo extension |
| 21:11.59 | Kobaz | it doesn't matter what it is |
| 21:12.01 | Kobaz | lemme paste |
| 21:12.17 | Kobaz | http://pastebin.ca/1875626 |
| 21:12.35 | Kobaz | dialplan remove extension s@app_dial_gosub_virtual_context <tab> |
| 21:13.07 | Corydon76-dig | Kobaz: you should NOT be removing that extension |
| 21:13.18 | leifmadsen | that's autogenerated |
| 21:13.34 | Kobaz | i know |
| 21:13.36 | Kobaz | but |
| 21:13.43 | Kobaz | asterisk shouldn't crash when you tab complete |
| 21:13.53 | Kobaz | it doesnt matter what exten@context you use |
| 21:14.05 | Corydon76-dig | Then use a legitimate one |
| 21:14.41 | Corydon76-dig | More specifically, remove one that was ALSO generated by pbx_config.so |
| 21:15.03 | Kobaz | context foo { a => { NoOp(); } } |
| 21:15.12 | Kobaz | Corydon76-dig: i'm not removing one |
| 21:15.15 | Kobaz | i'm using tab completion |
| 21:15.23 | Kobaz | dialplan remove a@foo <tab> |
| 21:15.26 | Kobaz | seg fault |
| 21:15.27 | pabelanger | Kobaz: load the sample dialplans and try 'dialplan remove extension s@demo <tab> <tab> <tab>' |
| 21:15.35 | Kobaz | pabelanger: same thing |
| 21:15.39 | Kobaz | it's the tab completion code |
| 21:15.48 | pabelanger | Kobaz: What OS? |
| 21:16.04 | Kobaz | linux... and kernel 2.6.27.9 |
| 21:17.28 | pabelanger | And your on 1.6.2? |
| 21:17.45 | Kobaz | it happens on 1.6.0.26 and 1.6.2.7 |
| 21:17.54 | Kobaz | i haven't tested 1.6.1.x |
| 21:18.07 | pabelanger | no point, honestly |
| 21:19.49 | pabelanger | Oh well, will try again tomorrow. |
| 21:19.50 | pabelanger | Later all |
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| 21:54.04 | Kobaz | hmm, |
| 21:54.12 | Kobaz | i see ast_add_extension()... how would i create a new context |
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| 22:54.04 | Marquis42 | Kobaz: ast_context_find_or_create() |
| 22:58.37 | Kobaz | yeah, i found it... i was expecting ast_add_context or something like that |
| 22:58.39 | Kobaz | took some digging |
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