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| 02:35.49 | snuff-work | file i responded to ur question on cr 2710 but no love from u =) |
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| 09:44.07 | file | snuff-work, can you sanitize the packet so anyone can look at it? |
| 09:47.06 | file | ooh, there's one on the JIRA |
| 09:52.35 | jkroon | file, i note on gerrit there is like only about 20-30 reviews pending - or am I completely missing something? I'm actually somewhat impressed if that is the case. or highly disturbed. either the process is working quite well and there are very few serious issues or it's so bad that nobody cares any more. I'm inclined to think the former. |
| 09:53.15 | file | there's 19 |
| 09:53.41 | file | generally stuff moves fast, unless it's a huge far reaching change |
| 09:54.05 | file | or feedback is given and the submitter does not deal with it, then it falls to someone else which causes it to take longer |
| 09:54.34 | jkroon | eg, the PLC one which I kinda impromptu picked up on even though it doesn't directly affect me? |
| 09:54.54 | file | yes |
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| 09:59.01 | file | not responding can also cause ripples, particularly if you are adding lots of features to PJSIP as the database upgrade scripts are linear - so if one goes in, it then breaks your next change, and so on |
| 09:59.49 | file | which is why 2551 is in merge conflict and I haven't dealt with it - that submitter's changes on 2624/2741 will break 2551 |
| 10:00.00 | file | and 2624/2741 is further along so I updated them |
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| 13:39.10 | acidfu_ | Im trying to find the changelog that says that in Asterisk 11, the SIPTRANSFER_REFERER have now <sip:...> around it |
| 13:39.10 | acidfu_ | anyone knows it ? |
| 13:39.36 | cresl1n | Maybe file knows⦠|
| 13:39.46 | file | um |
| 13:40.02 | file | nope |
| 13:40.05 | acidfu_ | changelog or anything else on the web page would help :) |
| 13:40.29 | file | what's the exact channel variable? |
| 13:41.34 | acidfu_ | this one: SIPTRANSFER_REFERER |
| 13:42.13 | acidfu_ | they talk about it here: https://www.osso.nl/blog/asterisk-dialplan-variable-expansion-security/ |
| 13:43.16 | file | I don't think anything has touched that code in awhile, it could depend on what is received |
| 13:44.43 | acidfu_ | from what they say " ;; drop optional <sip:...> -- changed between asterisk 10 and 11 " maybe that might ring a bell :| |
| 13:44.59 | file | oh, I thought you meant within 11 itself |
| 13:45.52 | acidfu_ | oh sorry for that misunderstanding |
| 13:47.20 | file | it was done 4 years ago or so |
| 13:47.23 | file | commit e5210366e |
| 13:47.37 | file | as a general clean up |
| 13:48.07 | file | don't know why, might have just changed the value by accident |
| 13:55.03 | acidfu_ | @file thanks very much :) |
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| 19:24.28 | file | glues cresl1n to the channel |
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| 19:46.10 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Check out our blog! asterisk-blog.com -=- Follow on Twitter at @AsteriskDev |
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| 19:50.54 | jameswf | irc(cloud) had a moment |
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| 19:53.34 | file | jameswf, when doesn't it? |
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| 22:47.52 | tm1000 | file: so can I not reference the DB while making a hint? |
| 22:47.54 | tm1000 | pbx_functions.c: Dangerous function DB read blocked |
| 22:48.08 | tm1000 | exten => _*27*X.,hint,${DB(AMPUSER/${EXTEN:4}/tc/hint)} |
| 22:48.14 | file | if you enable live dangerously you can |
| 22:48.26 | tm1000 | file: yeah but damn, that enabled shell mode |
| 22:51.25 | *** part/#asterisk-dev kharwell (kharwell@nat/digium/x-cscnumbevykeekru) |
| 22:52.16 | file | has no alternative presently |
| 22:53.13 | tm1000 | file: im finding that sometimes it works and sometimes it doesnt. This is before Iâve even changed live_dangerously |
| 22:53.28 | file | depends on where it comes from |
| 22:54.22 | tm1000 | file: in the dialplan or where its being called from? |
| 22:54.28 | file | where it is being called from |
| 22:55.14 | tm1000 | file: seems to work if I have an extension that calls the dynamic hint through a phone, but if I call extensionStatus from the AMI it is blocked |
| 22:55.18 | tm1000 | does that sound correct? |
| 22:55.28 | file | yes |
| 22:55.43 | file | I suspect it was meant to stop direct access |
| 22:55.49 | tm1000 | file: clarification accepted thank you |
| 22:56.56 | X-Rob | Bah |
| 22:57.07 | X-Rob | I guess we're living dangerously. |
| 22:57.10 | file | but, it's not quite aware that it's indirect... |
| 22:57.32 | file | I'm also not sure of the implications of "fixing" that... |
| 23:02.35 | tm1000 | file: ok so Im trying to generate a âdynamicâ hint from the AMI (using ExtensionState) which unfortuantely has a DB call in it. Is there any other way to get asterisk to âgenerate" a hint |
| 23:03.01 | file | not afaik from that |
| 23:03.38 | file | gtjoseph, . |
| 23:05.40 | file | I shall ask gtjoseph to look into the implications and possibility of allowing it indirectly like that (the use of DB) |
| 23:05.59 | file | I *think* it should be fine |
| 23:06.06 | gtjoseph | I'm trying to follow... |
| 23:06.43 | file | gtjoseph, I'm not sure that the use of dialplan functions within the dialplan in a hint should be restricted from use as a result of a reload over AMI |
| 23:06.47 | file | GetVar and SetVar I can understand |
| 23:07.15 | tm1000 | this isnât only reload |
| 23:07.16 | gtjoseph | hmmmm. |
| 23:07.31 | tm1000 | itâs when I run âExtensionStateâ from the AMI against a dynamic hint that hasnt been generated |
| 23:07.41 | file | tm1000, yes - it's an indirect use |
| 23:07.45 | tm1000 | file: cool |
| 23:09.08 | gtjoseph | I'll look more at this in the morning. |
| 23:09.51 | tm1000 | basically freepbx has thousands of hints. probably tens of thousands. weâve been trying to move these to dyanmic to make reloads faster |
| 23:10.23 | gtjoseph | and agree. if the an access is already in the dialplan then ami shouldn't restrict it |
| 23:15.28 | snuff-work | file: you talk about implementing a no content type INFO for pjsip.. have an skeleton i could work on to make such a thing ? =) |
| 23:15.44 | file | the existing dtmf info one is very skeleton |
| 23:15.52 | snuff-work | k |
| 23:18.38 | snuff-work | i also have issue with pjsip seeming to not update CLID info when sent an UPDATE msg.. |
| 23:18.46 | snuff-work | during call |
| 23:19.30 | file | you could trace res_pjsip_caller_id and see if stuff is being invoked... if not then that'd be why |
| 23:19.37 | snuff-work | or maybe its just that its not pushing it all the way to the phone |
| 23:20.09 | snuff-work | is there something that needs to be turned on at phone endpoint config to allow those msgs? |
| 23:20.51 | file | RPID or PAI support |
| 23:21.24 | snuff-work | mm.. i have trust_id_inbound/outbound on.. and send_rpid=yes |
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| 23:51.17 | tm1000 | gtjoseph: if you are still on ASTERISK-25966 your solution worked 100% |
| 23:51.27 | tm1000 | can I close it or do you need it for something else? |
| 23:52.34 | gtjoseph | leave it open. gerrit will get it |