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| 02:54.43 | skyroveRR | Hello all. |
| 03:07.30 | WIMPy | hi skyroveRR |
| 03:09.16 | carrar | hi!! |
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| 03:23.36 | skyroveRR | I'm new to asterisk, so I'm reading the asterisk 3rd edition docs :) |
| 03:24.08 | WIMPy | That's a book, not documentation. |
| 03:24.19 | skyroveRR | Oh. |
| 03:27.19 | MaliutaLap | although the book is a good starting point |
| 03:27.25 | carrar | indeed |
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| 03:27.53 | MaliutaLap | explains many of the concepts and gives you a guide to a basic setup, also a small glimpse of what is possible |
| 03:29.08 | carrar | and then it all changes in Asteirsk 13 ;) |
| 03:29.15 | carrar | j/k |
| 03:29.55 | MaliutaLap | carrar: shhh |
| 03:30.10 | MaliutaLap | carrar: it's all about the false sense of security! ;P |
| 03:30.27 | carrar | it's a good book, worth purchasing |
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| 03:33.44 | drmessano | I have a signed copy |
| 03:33.51 | drmessano | Just sayin |
| 03:34.24 | carrar | haha |
| 03:34.51 | carrar | I used to put my names in the first pages too when I bought books |
| 03:35.02 | ChannelZ | I have a digitally signed copy |
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| 07:32.27 | mjt | hello. Not really asterisk-related but: when I can see all network traffic between two SIP endpoints, can I extract SIP account info used (eg, username+password)? |
| 07:32.52 | robmal | Yes. |
| 07:33.21 | mjt | hwell. :( |
| 07:33.22 | mjt | ;) |
| 07:33.44 | mjt | is it something like http basic auth? |
| 07:34.50 | mjt | or more like digest |
| 07:35.38 | mjt | well, it looks like at least they use a random "nounce" |
| 07:35.48 | mjt | *"nonce" |
| 07:39.00 | mjt | so it looks like extracting account info is not at least easy |
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| 15:07.35 | DeeJayTwo | Hi |
| 15:07.50 | DeeJayTwo | Is there a way to have debug level 3 in the console at asterisk startup? |
| 15:08.10 | WIMPy | asterisk.conf |
| 15:08.14 | DeeJayTwo | I know how to enable debug messages in the console with logger.conf but I don't know how to set it at a specified level |
| 15:09.11 | DeeJayTwo | WIMPy: "grep -r -i level" in /etc/asterisk returns nothing... |
| 15:09.47 | DeeJayTwo | a search for debug gives me logger_* files... |
| 15:10.03 | WIMPy | Who said the option would be called "level"? |
| 15:10.49 | DeeJayTwo | how do you replicate a "core set debug 3" ? |
| 15:10.52 | DeeJayTwo | in configuration file? |
| 15:11.10 | WIMPy | Have you tried to look at that file? |
| 15:11.16 | DeeJayTwo | yes |
| 15:11.29 | WIMPy | Then it should be pretty obvious. |
| 15:11.42 | DeeJayTwo | it's a freepbx install |
| 15:11.48 | DeeJayTwo | there's not much in asterisk.conf |
| 15:12.14 | DeeJayTwo | only directories and 3 options... |
| 15:12.22 | WIMPy | Well, if you're usinf FreePBX, you are better of inn #freepbx. |
| 15:12.39 | DeeJayTwo | that's not the freepbx part of it that's interesting me. |
| 15:12.41 | WIMPy | But you can try to set verbose=3 there. |
| 15:12.51 | WIMPy | Not sur how long that would last, however. |
| 15:13.09 | DeeJayTwo | thank you, I'm trying! |
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| 16:12.39 | igcewieling1 | I noticed cel_odbc and cdr_adaptive_odbc do not include the [C-00008ff7] sort of tags. |
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| 16:22.48 | gmalsack | hey all... one of the projects I'm working on, they claim they are using a websocket client to create a channel between xlite and asterisk (ok this seems logical), then when a call comes in from vitelity, the websocket client is bridging the call to the xlite client and there is nothing in extensions.conf related to the call process. is this really possible? |
| 16:23.08 | gmalsack | btw, they are using pjsip |
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| 16:35.00 | linocisco | hi all |
| 16:37.29 | linocisco | hi all I want to provision grandstream GXP1405 phone . with reference to http://www.voip-info.org/wiki/view/Grandstream+provisioning+in+asterisk, I found some config about grandstream.xml. I dont know where that file should be located |
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| 16:39.25 | qakhan | is there any way we can create centralize phone directory in Asterisk |
| 16:39.34 | qakhan | i have Polycom phones |
| 16:39.45 | [TK]D-Fender | qakhan: No. |
| 16:39.59 | [TK]D-Fender | qakhan: Read the Polycom Admin Guide. It tells you what you can do with them. |
| 16:40.35 | qakhan | so asterisk does not support directory, it is phone feature |
| 16:40.37 | qakhan | right? |
| 16:41.33 | [TK]D-Fender | Asterisk does not do what the phone looks for. |
| 16:41.51 | qakhan | ok |
| 16:41.54 | qakhan | thanks |
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| 16:47.07 | [TK]D-Fender | linocisco: That guide is practically worthless. Nobody uses users.conf |
| 16:47.37 | linocisco | [TK]D-Fender, what is updated one? |
| 16:47.45 | [TK]D-Fender | There is none |
| 16:48.07 | [TK]D-Fender | That was the only time Asterisk itself ever had any framework for this concept |
| 16:48.17 | [TK]D-Fender | it isn't *'s job in the first place |
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| 17:53.31 | gmalsack | [TK]D-Fender: why not use users.conf? it's easy! :-) says a dCAP who learned asterisk via the astgui days... sip and voicemail all in one! |
| 17:54.19 | gmalsack | newtonr: you didn't see me say that. lol |
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| 17:57.49 | newtonr | gmalsack, thems fighting words |
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| 18:58.16 | phix | gang! |
| 19:06.06 | TazzNZ | where ? |
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| 20:21.51 | qakhan | is QUEUE_PRIO still available in asterisk 11.10? |
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| 20:57.32 | malcolmd | it's still part of app_queue as far as i can see, even into 13 |
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| 21:05.28 | DeeJayTwo | hi |
| 21:05.36 | DeeJayTwo | I fixed a memory leak into app_voicemail.c |
| 21:05.53 | DeeJayTwo | it wasn't closing sockets for imap backend... |
| 21:06.11 | DeeJayTwo | and a module reload would just add new imap connections without closing the previous ones... |
| 21:06.21 | DeeJayTwo | now, I'm in this situation: |
| 21:06.29 | DeeJayTwo | module load app_voicemail = OK |
| 21:06.39 | DeeJayTwo | module unload_app_voicemail = OK |
| 21:06.47 | DeeJayTwo | module reload app_voicemail = OK |
| 21:06.52 | DeeJayTwo | but.. |
| 21:06.58 | DeeJayTwo | core reload crashes asterisk |
| 21:07.31 | DeeJayTwo | I added debug lines in load_module, unload_module and load_config |
| 21:07.34 | DeeJayTwo | they don't show up |
| 21:07.44 | DeeJayTwo | when doing "core reload" |
| 21:09.34 | DeeJayTwo | not sure what do do next.. :/ |
| 21:09.43 | DeeJayTwo | I'm not very experienced with C and Asterisk code.. |
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| 21:51.34 | mjordan | DeeJayTwo: I would suspect that your solution is worse than the problem. And most likely, a reload is causing you to close connections that don't exist. But without a backtrace, that's just speculation. |
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