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| 00:32.47 | cdr1 | hi |
| 00:33.43 | cdr1 | somebody her? |
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| 00:43.27 | cdr1 | ? |
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| 05:36.37 | snuff-work | wedhorn: ping ? |
| 05:40.52 | snuff-work | thought of minor improvement to skinny.. when another call comes in and you select the 2nd one.. should put 1st on hold (like ccm does).. |
| 05:41.16 | snuff-work | also seem to be able to crash * with use of cfwd all button |
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| 16:15.05 | cdr1 | hi |
| 16:15.49 | cdr1 | is their somebody knowing pjsip in asterisk 12? |
| 16:16.59 | pabelanger | cdr1, as your questions is the best approach |
| 16:18.00 | cdr1 | ? ok, I would like to know how to raise vebosity or debug logging for pjsip, as it does not seems to be affected by logger.conf |
| 16:18.53 | pabelanger | pjsip set debug 10 ? |
| 16:19.53 | cdr1 | nop... tels me no such command |
| 16:20.04 | cdr1 | i use pjsip set logger on |
| 16:20.29 | cdr1 | which is a first step but not verbose |
| 16:21.02 | cdr1 | note that is use astersik 12 beta1, should i use the latest svn trunk? |
| 16:25.34 | mjordan | there have not been significant changes in log output since beta1 |
| 16:26.16 | mjordan | the module res_pjsip_logger provide SIP message tracing. The module res_pjsip_log_forwarder is what forwards message from PJSIP to the Asterisk logging facilities |
| 16:27.42 | cdr1 | ok and can i configure or cmd res_pjsip_logger? |
| 16:28.34 | mjordan | if the module is loaded, you should have the CLI command pjsip set logger {on|off|host} |
| 16:28.35 | cdr1 | As far as i understand pjsip = pjproject = external lib |
| 16:29.15 | cdr1 | yes i have pjsip set logger on => that does not show verbose I get notices and errors |
| 16:29.38 | mjordan | As I just said, res_pjsip_logger has nothing to do with the Asterisk logging facilities. It only deals with SIP message tracing. |
| 16:30.29 | cdr1 | So pjsip is almost independant of asterisk (separate lib)? |
| 16:30.33 | mjordan | res_pjsip_log_forwarder forwards messages from PJSIP to the Asterisk log files. It is not configurable. It will produce messages based on the levels provided by PJSIP, which has a static mapping defined in that file. |
| 16:30.39 | mjordan | it is completely independent. |
| 16:30.49 | file | a better question may be... what are you expecting/want to see? |
| 16:31.01 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Installing+pjproject |
| 16:31.15 | cdr1 | ok so I should begin to search on pjsip api and src |
| 16:31.57 | cdr1 | I have 2 problems, first when sip register, it cannot find aor for my endpoint... I reused the exemple from the wiki |
| 16:32.31 | file | on startup if there was a problem with the aor entry it would have mentioned it in the console |
| 16:32.51 | cdr1 | the second is to establish a dtls srtp connection, i want to have more detail than error... |
| 16:33.19 | cdr1 | I will look at startup log so... not though of that |
| 16:34.54 | file | and this is more #asterisk than it is #asterisk-dev |
| 16:35.26 | cdr1 | I agree, sorry, for this one, but my main interest is digging in pjsip for DTLS |
| 16:35.52 | file | DTLS support is the same as chan_sip and unrelated to pjsip |
| 16:37.54 | cdr1 | I don't get it. both chan_sip and pjsip use libsrtp to setup the dtls connection, but chan_sip or pjsip are responsible to find sdp infos to ask libsrtp to setup dtls |
| 16:38.56 | file | it uses the pjmedia SDP parser to get the raw attributes but further parsing and intepretation is done in chan_pjsip, and the code is very similar between both chan_sip and chan_pjsip |
| 16:41.40 | cdr1 | I try pjsip because asterisk 11.6 chan_sip was not finding the sdp fingerprint attribute, so not establishing DTLS |
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| 17:03.26 | cdr1 | ok... so (asterisk12beta1src)res/res_pjsip_sdp_rtp.c is an astersik file right? or pjsip? |
| 17:04.25 | cdr1 | Let me reformulate, is it the right place to ask for pjsip DTLS issue? |
| 17:04.58 | cdr1 | As i have an error in res/res_pjsip_sdp_rtp.c |
| 17:09.15 | file | that is an Asterisk file, yes |
| 17:12.51 | cdr1 | ok so I am in the right place :) |
| 17:13.19 | file | what is the error? |
| 17:14.21 | cdr1 | setup_dtls_srtp() dtls->set_configuration raise error "Attempted to set an invalid DTLS-SRTP configuration on RTP instance" |
| 17:15.27 | file | then that would be coming from res_rtp_asterisk |
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| 17:15.53 | cdr1 | please explain more in details |
| 17:17.17 | file | that code ends up calling into res_rtp_asterisk which sets the DTLS configuration, which has returned an error |
| 17:19.50 | cdr1 | its not clear for me what is the processing flow of the sdp packet? |
| 17:20.20 | cdr1 | res_rtp_asterisk => res_pjsip_sdp_rtc => libsrtp? |
| 17:20.51 | file | res_pjsip_sdp_rtp parses the SDP and interprets it, in the case of DTLS this information is then given to res_rtp_asterisk to configure DTLS-SRTP support on the RTP instance |
| 17:21.21 | cdr1 | ok, I begin to figure out the schemas |
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| 17:30.59 | cdr1 | hum... how can I know if ICE completed with success before DTLS starts? |
| 17:32.08 | cdr1 | I guess I have to run asterisk in a debugger... any advice to make this less painfull |
| 17:36.16 | cdr1 | Hum... forget about ICE, this is not the problem, as the DTLS handshake does not begin, this cannot be a connectivity issue |
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| 21:08.13 | n0taBug | Hi Guys, I need some help understanding the un-documented extensions.conf option called 'overrideswitch=yes', essentially what I am trying to do is use Asterisk Realtime to create submenu's dynamically with out needing static proxy contexts in extensions.conf for each submenu containting the switch => realtime line... and ofcourse I am trying to no have to 'dialplan reload' each time I create a new submenu. |
| 21:08.13 | n0taBug | From reading here: http://forums.asterisk.org/viewtopic.php?p=160660 Malcolm Davenport (malcolmd) confirmed that the 'overrideswitch=yes' option was merged, and from my testing on v11 Asterisk does know what it is (I get the warning No such switch 'yes' on the CLI) - can any of you guys help me underestand how to use this feature so I can dynamically create submenu's without reloading Asterisk and without the |
| 21:08.13 | n0taBug | static proxy contexts in extensions.conf? Thanks in advance. |
| 21:09.57 | mjordan | n0taBug: you might get some takers, but not very likely. First, it's not really a development question, so you might be better off asking in #asterisk. Second, not many of us use realtime dialplan. It's not highly encouraged. You might be better off asking the asterisk-users mailing list, as that would reach a wider audience. |
| 21:10.16 | mjordan | ah, and you did ask in #asterisk |
| 21:10.46 | mjordan | looks a bit. |
| 21:11.40 | n0taBug | mjordan: yea I first spent the entire morning trying to figure it out on my own, then asked on #asterisk and then was directed its prolly a good idea to ask here.... I did do my due diligence as much as possible before resorting to asking here, but I do apprecate you looking at this...if it helps, I do see that the option does exist in pbx.c, I just have no idea how to use it |
| 21:12.23 | mjordan | so... first, it's only valid in [general]. |
| 21:12.25 | mjordan | looks some more. |
| 21:13.01 | n0taBug | right, thats what I gather too, and when its in general v11 does try to use it...so I must be using it wrong |
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| 21:17.11 | mjordan | what is the name of your overrideswitch? |
| 21:17.40 | n0taBug | [bypass_switch] |
| 21:17.40 | n0taBug | switch => Realtime/default@extensions |
| 21:17.48 | n0taBug | is what I have at the bottom of extensions.conf |
| 21:17.55 | mjordan | yeah |
| 21:18.03 | mjordan | overrideswitch is not a 'yes'/'no' thing |
| 21:18.16 | mjordan | it is the name of a switch |
| 21:18.46 | n0taBug | yea I thought so too, so I even tried ovverideswitch=bypass_switch to no avail, I just got the warning no such switch 'bypass_switch' |
| 21:19.20 | mjordan | that's the context that the switch is in |
| 21:19.36 | mjordan | it's looking for the actual switch, so, if I'm reading this correctly, Realtime/default@extensions |
| 21:20.04 | mjordan | file: would you believe we have NEED_DEBUG_HERE as a #defined macro in pbx.c? |
| 21:20.06 | mjordan | :-( |
| 21:22.16 | mjordan | n0taBug: if I wasn't clear enough, that means that (I think) it should be overrideswitch=Realtime/default@extensions |
| 21:22.40 | n0taBug | ah, ok let me give that a shot |
| 21:25.48 | n0taBug | mjordan: ok that did get rid of the warning message....but it still does not find the context in the realtime extensions database, sais "Attept to reach a non-existent desination"...I guess thats 'some' progress :) |
| 21:26.06 | mjordan | hrm. |
| 21:27.12 | mjordan | interesting. It attempted to execute a GoSub at that point. |
| 21:27.30 | n0taBug | correct |
| 21:27.56 | n0taBug | Pressing option 5 does a GoTo and trys to send to that newly created submenu which exists in the realtime database |
| 21:29.29 | mjordan | No idea what that means. :-) I'm guessing that's your dialplan. |
| 21:30.05 | n0taBug | Sorry, yes in my dial plan for menu1 pressing 5 does a GoTo and tries to send to menu2 which is a new menu created in the realtime database |
| 21:31.36 | n0taBug | I know that if I make the static context in extensions.conf called menu2 and put switch => realtime then dialplan reload the menu will work, but from what I read this overrideswitch option is supposed to avoid having to create these static proxy context's..... |
| 21:32.37 | mjordan | No idea. Since this feature isn't really documented and/or used, I think you have to approach anything regarding it cautiously |
| 21:33.09 | mjordan | The GoTo will bomb out however if it can't find the context of the thing you're looking up. That happens before the overrideswitch |
| 21:33.56 | mjordan | my guess is that the context at least has to be a matchable thing. |
| 21:35.12 | n0taBug | sure wish there was more info on this thats forsure....overrideswitch=Realtime in general without the "/defaul@extensions" part segfaults asterisk ....whoops! |
| 21:35.24 | mjordan | sighs. |
| 21:35.39 | mjordan | really, if this entire 'feature' surfaced today, it would never go in. |
| 21:36.32 | n0taBug | I guess if the purpose of realtime asterisk is to avoid doing reloads, yet you cant create new contexts in realtime wihtout a realod...whats the purpose of even supporting the dialplan in the realtime db...know what I mean? |
| 21:37.15 | mjordan | heh |
| 21:37.29 | mjordan | dialplan in a realtime database is not a concept we typically advocate. |
| 21:37.41 | mjordan | It doesn't really map terribly well, and it has other known issues. |
| 21:37.50 | n0taBug | yea I saw you mentioned that earlier... I was wondering why you said that.. |
| 21:37.59 | mjordan | well |
| 21:38.13 | mjordan | (1) programmatic execution doesn't map well to relational databases |
| 21:38.35 | mjordan | (2) There are other tools today (chef, puppet, git) that allows for remote version control management/ pushing of configuration information to multiple systems |
| 21:38.54 | mjordan | (3) There are known deadlocks with pbx_realtime |
| 21:39.10 | mjordan | but really, the big one is #2. |
| 21:39.17 | mjordan | For what it attempted to solve, there are better tools today. |
| 21:39.29 | mjordan | If I have a n Asterisk systems and I want them all using the same dialplan, I'd just use pupet. |
| 21:39.31 | mjordan | puppet. |
| 21:39.39 | mjordan | And Git to manage the version control |
| 21:40.07 | n0taBug | I see I see...so if you were to do a configuration where you had ever changing menu's, new menus being added, old ones removed, you would still do it in the dial plan in static files, then reload each time? |
| 21:40.33 | mjordan | Nope |
| 21:40.34 | mjordan | I'd use AGI. |
| 21:40.42 | mjordan | Particularly if I was changing them a lot. |
| 21:40.53 | pabelanger | +1 |
| 21:40.54 | mjordan | leifmadsen can flame me for that |
| 21:40.59 | mjordan | or pabelanger, but yayz! |
| 21:41.01 | mjordan | I got it right |
| 21:41.03 | Marquis42 | +1, although there are ways of making that kind of thing database-driven without using realtime. :) |
| 21:41.09 | n0taBug | yea that was suggested to me earlier too.......I hadnt thought of using AGI becase really I didnt want to re-invent whats already there in realtime |
| 21:41.42 | pabelanger | Puppet is smart enough to know that extensions.conf has changed then you can invoke dialplan reload to update asterisk |
| 21:43.02 | n0taBug | yea my whole thing was avoiding having to do the dialplan reload under production systems which is why I was trying the realtime approach |
| 21:43.18 | mjordan | What is the rate of change for these parts of your dialplan? |
| 21:43.22 | mjordan | 20 times a day? 10? |
| 21:43.31 | n0taBug | easily |
| 21:43.41 | pabelanger | well, that is more a deployment / configuration issue then asterisk |
| 21:43.56 | pabelanger | If you test everything before changing it production, it won't break |
| 21:44.16 | pabelanger | Or schedule maintenance windows |
| 21:44.44 | n0taBug | not sure I follow.....im not worried about the changes breaking the dialplan... I was more worried about the constant dialplan reloads so many times per day on a production system |
| 21:44.55 | pabelanger | then use AGI |
| 21:45.06 | pabelanger | no dialplan reloads |
| 21:45.38 | n0taBug | yea, seems like thats going to be the way I need to go |
| 21:46.19 | n0taBug | thanks for taking the time on this one guys.... apprecate it |
| 21:46.21 | mjordan | n0taBug: np |
| 21:46.27 | mjordan | sorry it's not the answer that you hoped for |
| 21:46.37 | mjordan | but I think in the long run, you'll be much more stable if you avoid realtime dialplan |
| 21:46.38 | n0taBug | its more then I had when I first came here right? :) |
| 21:47.00 | n0taBug | thats good feedback, thanks for that |
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| 21:58.23 | leifmadsen | mjordan: using puppet is for people who haven't learned of the magical and better ways of chef |
| 21:58.26 | leifmadsen | .trololo |
| 22:00.34 | pabelanger | lies |
| 22:01.31 | leifmadsen | heh |
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| 22:53.21 | mjordan | and here I thought ansible was the new hotness |
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| 23:58.31 | *** join/#asterisk-dev Jamuel (~Adium@70-90-173-246-California.hfc.comcastbusiness.net) |