| 00:01.20 | *** join/#asterisk linetrace (~linetrace@65.19.81.126) |
| 00:02.12 | linetrace | best to ask compile questions here or #asterisk-dev? |
| 00:02.45 | WIMPy | If it's about the source, probably in -dev, otherwise here. |
| 00:04.08 | linetrace | specifically, getting "Unknown value '' found in build_tools/menuselect-deps for NATIVE_ARCH" when running menuselect, probably related to configure options |
| 00:07.37 | WIMPy | I'd take that to -dev. |
| 00:08.20 | linetrace | thanks |
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| 00:09.02 | Kobaz | WIMPy: i think that'll go on my add-todo list if it's not in 12 |
| 00:11.36 | siraskalot | hello everyone! |
| 00:11.36 | WIMPy | We are on 13 already. So wishes go for 14. |
| 00:12.37 | siraskalot | i'm a newbie here, please spare me the warm welcome.^_^ |
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| 00:13.22 | WIMPy | Everyone has been new at some time. Just |
| 00:13.23 | WIMPy | ~ask |
| 00:13.24 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 00:16.49 | siraskalot | thank you very much. |
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| 00:23.59 | Kobaz | WIMPy: hah, i'm a little out of the loop |
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| 00:28.11 | lnb | got a box running freepbx running Asterisk 12.6.1. Since upgrading the server freepbx to 6.12 (not sure which version of Asterisk was running prior to upgrade) BLF monitored lines no longer display CallerID. Instead the display is monitored extension. It seems to stem from what Asterisk is generating. http://pastebin.ca/2874485 |
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| 00:47.36 | lnb | on another freepbx server that has blf monitored lines that do show callerid/cnam, its running Asterisk 11.13.0 |
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| 00:48.53 | lnb | just did a asterisk-switch-version from 13 -> 12 -> 11 and now it works |
| 00:49.28 | lnb | so something or things have changed in Asterisk from 12+ that are affecting BLF monitored lines |
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| 00:55.52 | [TK]D-Fender | Flawed logic based on evidence |
| 01:04.07 | lnb | really |
| 01:04.33 | lnb | i would say its picture perfect logic based on logical tests |
| 01:04.45 | [TK]D-Fender | You've proved nothign |
| 01:04.59 | [TK]D-Fender | you haven't shown configs between versions, comparitive output or anything |
| 01:05.00 | lnb | that clearly proove a change has occured between ASterisk 11 and 12 |
| 01:05.24 | [TK]D-Fender | in fact today is the first day in the months you been pasting that same useless claim about it that you showed ANYTHING at all |
| 01:05.29 | [TK]D-Fender | Where are the configs? |
| 01:05.32 | lnb | tell ya what, i will take a picture of the phone getting a call on its BLF monitored line so you can compare the two images |
| 01:05.35 | [TK]D-Fender | Where are the CLI settings dumps? |
| 01:05.37 | lnb | what config? |
| 01:05.39 | [TK]D-Fender | You show NOTHING |
| 01:05.44 | lnb | asterisk-switch-version |
| 01:05.47 | [TK]D-Fender | MEANINGLESS |
| 01:05.52 | [TK]D-Fender | GUI garbage |
| 01:05.54 | lnb | that is what I ran |
| 01:05.56 | lnb | gui? |
| 01:06.02 | lnb | what f(@#U&) gui |
| 01:06.03 | [TK]D-Fender | FreePBX scripts |
| 01:06.05 | [TK]D-Fender | Which means nothing |
| 01:06.12 | [TK]D-Fender | The only thing that matters is ASTERIASK configs |
| 01:06.50 | [TK]D-Fender | FreePBX builds your configs and you've never shown them, especially comparitively |
| 01:07.06 | lnb | what i do know is BLF now works right. It shows the CallerID on a BLF monitored line |
| 01:07.24 | [TK]D-Fender | Still meaningless |
| 01:07.29 | [TK]D-Fender | you aren't showing proper backup |
| 01:07.34 | lnb | i dont show them because i don't know what its doing |
| 01:07.43 | [TK]D-Fender | You aren't looking and you aren't showing |
| 01:07.45 | lnb | by running asterisk-switch-version |
| 01:07.49 | [TK]D-Fender | meaningless |
| 01:07.50 | lnb | i watched it |
| 01:07.56 | [TK]D-Fender | Stop mentioning that FreePBX GUI script |
| 01:08.02 | [TK]D-Fender | that has no place here |
| 01:08.04 | lnb | there is no gui |
| 01:08.10 | lnb | its ssh'd to the server |
| 01:08.12 | [TK]D-Fender | talking about it does not prove how Asterisk is configured |
| 01:08.19 | lnb | correct |
| 01:09.02 | [TK]D-Fender | asterisk-switch-version <- this is a FreePBX script |
| 01:09.05 | [TK]D-Fender | it is NOT from Asterisk |
| 01:09.14 | [TK]D-Fender | And means nothing here |
| 01:09.30 | lnb | i will still take a picture of an incoming call and put it and the image of call showing what it does in the p/b i posted earlier |
| 01:09.31 | [TK]D-Fender | it is not a point of comparison in debugging |
| 01:09.39 | lnb | no ? |
| 01:09.41 | lnb | sure it is |
| 01:09.43 | [TK]D-Fender | [20:09]lnbi will still take a picture of an incoming call and put it and the image of call showing what it does in the p/b i posted earlier <- TWO calls |
| 01:09.51 | [TK]D-Fender | ANDW the full backup to match |
| 01:10.10 | [TK]D-Fender | [20:09]lnbsure it is <- no it isn't |
| 01:10.44 | lnb | if you look at the p/b you will see the cli output (partial) of the call that displays 1. 300 <- 300 and it clearly shows why |
| 01:10.48 | lnb | just look at the p/b |
| 01:10.52 | lnb | the cli says it all |
| 01:11.09 | lnb | its the notify statement |
| 01:11.42 | [TK]D-Fender | No, that sdoes not say it all. |
| 01:12.04 | [TK]D-Fender | Configs tell * how to operate and you have shown nothing. |
| 01:12.27 | [TK]D-Fender | You have not shown a proper comparison with full backup for both sides |
| 01:18.52 | lnb | http://barontel.com/images/yealink_t20_blf-working.jpg |
| 01:19.30 | [TK]D-Fender | lnb: nobody cares about what the phone displays |
| 01:19.38 | lnb | really? |
| 01:19.41 | [TK]D-Fender | All we care about is what Asterisk is doing |
| 01:19.52 | [TK]D-Fender | and we're still not seeing 2 sets of configs and debug |
| 01:19.54 | lnb | every company that i have setup so far ABSOLUTELY depends on it |
| 01:20.05 | lnb | for monitored lines |
| 01:20.07 | [TK]D-Fender | And every time you show some other random little thing it wastes everyones time\ |
| 01:20.22 | lnb | i am not wasting anyone's time |
| 01:20.26 | [TK]D-Fender | Stop making excuses and provide the backup |
| 01:20.31 | lnb | what backup? |
| 01:20.35 | lnb | i have no backup |
| 01:20.40 | [TK]D-Fender | show the 2 sets of config. |
| 01:20.41 | [TK]D-Fender | Now |
| 01:20.52 | lnb | what configs? where are they? |
| 01:20.53 | [TK]D-Fender | and matching call debug |
| 01:21.15 | [TK]D-Fender | [20:20]lnbwhat configs? where are they? <- the related Asterisk configs. |
| 01:21.21 | lnb | ok so you want two sets of calls. One with Asterisk 11 and one with 12 |
| 01:21.31 | [TK]D-Fender | EVIDENCE |
| 01:21.39 | [TK]D-Fender | You seem to have no comprehension of what that means |
| 01:21.47 | lnb | i do |
| 01:21.54 | lnb | but i do not know what configs you want |
| 01:22.07 | [TK]D-Fender | I've seen years of proof to the contrary |
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| 06:41.05 | OrNix | morning. tell me please, if I send outgoing call through Dongle/r1, how can I know what dongle was used for the call? |
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| 08:30.02 | cHarNe2 | hello guys, i have a dilemma on what codes to send back on invites. If i own the number but dont have it assigned. => 404, If i dont know the number at all. => 604. or sould it be reversed? |
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| 08:35.01 | kaldemar | cHarNe2: reversed |
| 08:35.48 | kaldemar | cHarNe2: see the response code chapter in RFC3261 |
| 08:36.38 | wdoekes | kaldemar: I would say that 604 would be for the "this request will never succeed here" case when he doesn't know the number at all |
| 08:37.27 | wdoekes | although I would send 404 in both cases. I rarely encounter 6xx codes |
| 08:39.57 | cHarNe2 | kaldemar, wdoekes thanks. will take both in consideration |
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| 10:05.36 | hrolf | Hi Asterisk |
| 10:06.17 | hrolf | I'm dialing a number of soft phone. Like 210 and then doing Dial(SIP/210) on that extension |
| 10:06.38 | hrolf | I have another extension 210 in same context |
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| 10:07.09 | hrolf | and in that I run IVR |
| 10:09.30 | hrolf | I want that Dial(SIP/210) to execute extension 210 in context |
| 10:10.15 | hrolf | In short |
| 10:10.33 | hrolf | How can a SIP phone do Dial(XXX) to an extension XXX in same context as the SIP device? |
| 10:10.49 | hrolf | I have, at present, written Dial(SIP/XXX) but it fails |
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| 10:11.36 | skrusty | are you confusing the extension number with the peer name? |
| 10:11.51 | kaldemar | SIP in a Dial is a channel type. that would point to something you have configured in a SIP channel driver. if you want to dial an extension in a context of your dialplan, use a local channel. Dial(Local/exten@context) |
| 10:12.13 | kaldemar | there's also Goto if you really don't have a reason to use a local channel. |
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| 12:06.40 | Stefan27 | On 13.0.0 I still get this weird warning everytime res_monitor.so loads [2014-11-19 09:25:52] WARNING[26207] loader.c: Error loading module 'res_monitor.so': /usr/lib64/asterisk/modules/res_monitor.so: undefined symbol: __ast_beep_stop |
| 12:07.56 | Stefan27 | how to 'debug' it? |
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| 12:31.42 | wdoekes | Stefan27: did you grep the source? |
| 12:32.42 | Stefan27 | i did, only binary matches |
| 12:33.35 | Stefan27 | no text file in (asterisk13.0.0 as unpacked from the tar) contains "__ast_beep_stop" |
| 12:35.15 | Stefan27 | is http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz supposed to contain the symbol __ast_beep_stop somewhere? |
| 12:37.37 | file | what is your system, what is your compiler? |
| 12:38.54 | Stefan27 | Fedora 20, 64 bit, libgcc-4.8.2-1.fc20.x86_64 |
| 12:39.12 | Stefan27 | wait -- |
| 12:39.42 | Stefan27 | gcc-4.8.3-7.fc20.x86_64 |
| 12:41.02 | eirirs | quesition, are chan_skinny and chan_sccp "the same thing" ? |
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| 12:42.09 | Stefan27 | build sequence is: ./configure --libdir=/usr/lib64 ;; make menuselect.makeopts ;; menuselect/menuselect --disable-category MENUSELECT_CORE_SOUNDS --disable-category MENUSELECT_EXTRA_SOUNDS --disable-category MENUSELECT_MOH --enable-category MENUSELECT_ADDONS --enable app_meetme --enable app_page --disable chan_mgcp --disable chan_mobile ;; contrib/scripts/get_mp3_source.sh ;; make %{makeflags} |
| 12:42.10 | file | Stefan27, I haven't heard anything explicitly about those... but it's possible that optional API is unhappy there... |
| 12:42.10 | Stefan27 | 2> err |
| 12:42.26 | Stefan27 | where %ifarch x86_64 |
| 12:42.26 | Stefan27 | %define makeflags OPT=-fPIC |
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| 12:45.19 | wasanzy | hello |
| 12:45.39 | wasanzy | I have a very big challenge. |
| 12:48.45 | wasanzy | I am using confbridge, for conferencing, but I realised confbridge does not exit the user when you enter automatically unless you terminate yourself. Is there a way of doing someting like a loop in confbridge so that for every 10 minutes, an external application is called and base on result of the application, the user is still maintain in the conference otherwise quit the user? |
| 12:49.10 | Eric-K | Hm, trying to setup Asterisk 13 but it does not seem to be able to find pjsip. I built and installed the latest pjsip. |
| 12:49.44 | Eric-K | Configured pjsip with --prefix=/usr/lib --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr CFLAGS='-O2 -DNDEBUG' |
| 12:49.56 | Eric-K | Still with make menuselect, pjsip is not available |
| 12:50.12 | Stefan27 | does OPTIONAL_API have to be enabled explicitly, it's prechecked [*] in the GUI which shows up if i do make menuselect from the unpacked tar asterisk13.0.0-folder, but in my commandline build seequence (rpm-build which starts from just the tar) I don't have menuselect/menuselect --enable OPTIONAL_API explicitly |
| 12:50.37 | file | Stefan27, it should be done explicitly but I don't know what'll happen with manual stuff like that... |
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| 12:56.03 | Stefan27 | Eric, seems to work for me, but i used pjprojects 2.2 (old) and ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --with-external-srtp --libdir=/usr/lib64 |
| 12:56.13 | wasanzy | Any help please? |
| 12:56.30 | Eric-K | Stefan27, S["PKG_CONFIG"]="/usr/bin/pkg-config" in config.status is incorrect. |
| 12:56.41 | Eric-K | On my system it's pkgconfig |
| 12:56.43 | Eric-K | without the minus |
| 12:56.46 | Eric-K | Changing that now |
| 12:56.47 | Stefan27 | what's your OS? |
| 12:57.07 | Eric-K | Debian 7 x64 |
| 12:58.14 | Eric-K | Hm 'checking for PJPROJECT' still no. Might need to set the path as well. |
| 12:58.56 | Stefan27 | <PROTECTED> |
| 12:59.07 | Eric-K | Nothing.... |
| 13:01.50 | *** join/#asterisk infobot (ibot@rikers.org) |
| 13:01.50 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.0.0 (2014/10/25), 11.14.0 (2014/11/10), 1.8.32.0 (2014/11/10); Standard: 12.7.0 (2014/11/10); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 13:02.22 | Eric-K | strange right |
| 13:05.41 | Stefan27 | Yeah -- i remember installing pjprojects was a process of trial and error, but i did it some months ago so i dont remember what i got stuck on |
| 13:06.24 | Stefan27 | shouldn't you install with --libdir=/usr/lib64 anyways? |
| 13:06.40 | Eric-K | I don't have a lib64 dir. |
| 13:06.44 | Stefan27 | oh |
| 13:06.56 | Stefan27 | nvm |
| 13:07.00 | Eric-K | AAAAH!!! |
| 13:07.02 | Eric-K | Found it!! |
| 13:07.07 | Eric-K | How stupid of me. |
| 13:07.23 | Eric-K | When configuring pjproject, ./configure --prefix=/usr/lib should have been ./configure --prefix=/usr |
| 13:07.26 | Eric-K | my god.... :D |
| 13:08.17 | Stefan27 | Oh yeah, didn't notice that in ur configure line either |
| 13:08.33 | Eric-K | Thanks for assisting me. |
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| 13:22.27 | eirirs | have chan_skinny been updated lately? or is it better to use chan_sccp version 4.2 since its newer? |
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| 13:25.46 | eirirs | anyone? |
| 13:26.10 | wasanzy | I need help |
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| 14:31.02 | cervajs2 | hi, can you recommend any tool which monitor AMI events ? ( + sorting, searching,.. ) |
| 14:32.28 | mjordan | sorting/searching wouldn't really make any sense with AMI. Events inform you about the live call state of the system. |
| 14:32.37 | mjordan | what are you trying to build? |
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| 14:35.51 | cervajs2 | i'm trying encode recordings based on hangup event. i have problem with agents in queue with local channel. i need visualize events to understand why i'm receiving hangup when the call is answered in the queue. |
| 14:36.41 | cervajs2 | i.e. i want sort events by uniqueid |
| 14:37.04 | Stefan27 | preload => func_periodic_hook seemed to remove my warning __ast_beep_stop not found when loading res_monitor.so ... |
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| 14:37.52 | cervajs2 | mjordan: ahh. misunderstanding. i want store events and then do some analysis |
| 14:39.14 | Stefan27 | is func_period_hook supposed to be preloaded? |
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| 14:45.28 | mjordan | Stefan27: no. Generally, that WARNING is spurious due to some of the oddities in the Asterisk module loader. |
| 14:45.51 | mjordan | cervajs2: You would need to write that application yourself. There are AMI libraries available in a large variety of languages. |
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| 15:16.09 | cervajs2 | mjordan: ok. maybe i will try push ami events to elasticsearch. |
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| 15:59.30 | lnb | [TK]D-Fender: http://pastebin.ca/2874897 |
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| 16:02.09 | matthew-moretalk | having some issues with our astersik/kamailio deployment for a few days I thought the issue was with kamailio but iv realised its not |
| 16:03.17 | [TK]D-Fender | is still waiting for the other half |
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| 16:04.01 | lnb | what half>? |
| 16:04.13 | lnb | there are two calls |
| 16:04.23 | lnb | one with asterisk11 and one with asterisk12 |
| 16:04.24 | [TK]D-Fender | What did I ask for yesterday about a dozen times? |
| 16:04.30 | lnb | some configs |
| 16:04.40 | lnb | but i do not know what configs you want |
| 16:04.44 | Eric-K | So I'm working with pjsip now, just testing of course. Can I add headers to my dialplan like P-Asserted-Identity? Like I used to do with sipaddheader? |
| 16:05.04 | Eric-K | Or does pjsip have a whole other approach. |
| 16:05.26 | lnb | you saw from the pb what was done ... how it was upgraded to 12 |
| 16:05.52 | [TK]D-Fender | And I told you repeatedly your FreePBX distro Asterisk swithing script means precisely nothing |
| 16:05.58 | [TK]D-Fender | And you keep talking about it. |
| 16:06.05 | [TK]D-Fender | Stop. |
| 16:07.00 | lnb | if it means nothing, why then when using it to go to asterisk 11 does the monitored line display the callers NAME |
| 16:07.26 | Eric-K | Ah, PJSIP_HEADER |
| 16:07.33 | lnb | i did not touch any configs and i still don't know what configs you are talking about |
| 16:07.39 | [TK]D-Fender | If you're not looking at configs along with your debug then are you aren't looking at how the car was built |
| 16:07.55 | lnb | what configs are you talking about? |
| 16:08.03 | [TK]D-Fender | What do YOU think? |
| 16:08.03 | lnb | give me file names |
| 16:08.11 | lnb | i dont think anything |
| 16:08.24 | [TK]D-Fender | Well aware |
| 16:08.25 | lnb | just say the freeking names already |
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| 17:52.06 | WIMPy | lnb: If you want to talk baout the effects of using som FreePBX script, you should do so in #freepbx. It doesn't mean anythign to us here. |
| 17:54.36 | lnb | WIMPy: i wasn't trying to talk about freepbx scripts. I was asking what the cause might be for monitored lines going from Asterisk 11 to 12 or 13. |
| 17:54.49 | WIMPy | But you are. |
| 17:55.20 | lnb | because of using the script to upgrade Asterisk? |
| 17:55.39 | WIMPy | Yes. Noone knows what it's doing. |
| 17:55.51 | lnb | you're right |
| 17:56.47 | WIMPy | Not here anyway. So ask those who should know. |
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| 19:17.42 | Eric-K | Switching to pjsip, I am trying to add a P-Asserted-Identity header in my dialplan. |
| 19:18.01 | Eric-K | So I tried working with https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER but no effect. |
| 19:18.27 | Eric-K | Even if I simply add a test header like described at the wiki, it's missing. |
| 19:18.46 | Eric-K | I normally worked with sipaddheader. |
| 19:19.08 | malcolmd | did you do it using a pre-dial handler? |
| 19:19.25 | Eric-K | No, I don't think I tried that. |
| 19:19.33 | Eric-K | With a handler context? |
| 19:20.02 | malcolmd | yop. there's a bit about that further down on that wiki page |
| 19:20.15 | Eric-K | My dial currently is: exten => _XXXXXXX.,n,Dial(PJSIP/${EXTEN}@mysipprovider) |
| 19:20.22 | Eric-K | let's see |
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| 19:24.08 | Eric-K | Works! Thanks malcolmd! |
| 19:24.46 | malcolmd | huzzah! :D |
| 19:25.32 | Eric-K | :D |
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| 19:59.35 | Eric-K | malcolmd I also noticed that there is no user agent header on my outgoing calls. |
| 19:59.51 | Eric-K | Adding a [global] context and setting user agent does not make a difference (pjsip). |
| 20:00.05 | Eric-K | Should I define that somewhere else? |
| 20:00.24 | Eric-K | I'd like it to display my current Asterisk version. |
| 20:02.44 | Penguin | [general] ? |
| 20:03.22 | Eric-K | In pjsip? |
| 20:04.08 | Eric-K | The whole user agent header is missing, usually it should default to the current version or something. |
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| 21:02.33 | malcolmd | that's weird. i'm pjsip and i see outbound invites from asterisk displaying the user-agent string, e.g. [Nov 19 15:01:39] User-Agent: Asterisk PBX SVN--r427306 and i'm not defining anything in the globals |
| 21:03.40 | [TK]D-Fender | Chan_sip always defaulted to the * version... |
| 21:03.50 | [TK]D-Fender | I would expect pjsip to do the same... |
| 21:04.11 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 21:04.31 | Katty | hello my asterisk does not work at all how to fix pls is urgent thx |
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| 21:06.42 | Katty | Nugget: ping |
| 21:07.17 | *** join/#asterisk mdhas (~mdhas@206.223.170.186) |
| 21:07.22 | Katty | i'm collecting names for my xmas card sendout |
| 21:07.36 | Nugget | hi hi |
| 21:07.50 | Katty | Qwell: so you, and drmessano |
| 21:07.59 | Katty | Penguin: you |
| 21:08.06 | Katty | Nugget: you want a card this year? |
| 21:08.24 | Nugget | I will cherish it! |
| 21:08.39 | Katty | excellent. i don't have an address for you from last year |
| 21:08.44 | drmessano | I'm looking forward to mine |
| 21:08.53 | drmessano | I still have last years on the wall in my office |
| 21:08.53 | Katty | so send me your address. |
| 21:09.00 | drmessano | Why you no still have it? |
| 21:09.08 | drmessano | Du Hast? |
| 21:09.14 | Katty | no i have your drmessano |
| 21:09.18 | Katty | last year nugget didn't want a card |
| 21:09.18 | drmessano | Oh |
| 21:09.22 | drmessano | Ohhhh |
| 21:09.30 | drmessano | Well screw him, I want mine |
| 21:09.35 | Katty | did you move since you and kathy split up? |
| 21:09.44 | Katty | or are you still in apt c |
| 21:09.45 | drmessano | No, I made that bitch move out |
| 21:09.55 | drmessano | lol |
| 21:09.56 | drmessano | Yes |
| 21:10.16 | Katty | you still have my address on woodland? |
| 21:10.40 | drmessano | I do |
| 21:10.53 | Katty | excellent |
| 21:11.19 | drmessano | Do you still dance naked in front of that window? |
| 21:11.24 | drmessano | I mean umm |
| 21:11.28 | drmessano | ./clear |
| 21:11.42 | Katty | rolls eyes |
| 21:11.50 | drmessano | :) |
| 21:12.16 | drmessano | You will be one of the very few cards I send out this year |
| 21:12.27 | Katty | yay |
| 21:13.55 | drmessano | I'm sending a card to you, Jesus, Santa, Julian Assange, Tom Baker, and Danny Cooksey |
| 21:14.22 | Katty | Penguin: PING. |
| 21:17.03 | *** join/#asterisk iggi (~iggi@ip69.50-31-21.static.steadfastdns.net) |
| 21:18.24 | Katty | file: do you want an xmas card this year? |
| 21:19.32 | glaz | Why is no one sending me xmas cards :( |
| 21:20.27 | Katty | glaz: well the answer to that is simple |
| 21:20.37 | Katty | glaz: because you have asked your friends if they woudl like a christmas card from you this year |
| 21:20.48 | Katty | glaz: don't wait for life to send you a card... |
| 21:20.52 | Katty | glaz: go out and YOU do the sending! |
| 21:21.07 | Katty | glaz: life is much more rewarding that way |
| 21:21.23 | glaz | That's a way for seeing things |
| 21:21.25 | glaz | a good one |
| 21:22.01 | *** join/#asterisk u0m3_ (~u0m3@92.80.86.86) |
| 21:23.08 | Katty | Kobaz: ping |
| 21:24.00 | glaz | But all I want for xmas is to find someone to replace me at work, that's not going to be easy! |
| 21:24.51 | *** join/#asterisk iggi (~iggi@ip69.50-31-21.static.steadfastdns.net) |
| 21:24.56 | [TK]D-Fender | packs up to head home |
| 21:25.04 | glaz | oh crap, 4:30pm, time to go home! |
| 21:25.04 | Katty | goshdarnitfender |
| 21:25.33 | *** join/#asterisk u0m3__ (~u0m3@92.80.86.86) |
| 21:25.35 | Penguin | katty: Hiiiiiiiii |
| 21:25.53 | Katty | Penguin: ohai! |
| 21:25.56 | Katty | Penguin: xmas card? |
| 21:26.15 | Penguin | katty: You may send me a Christmas card every year that you'd like. |
| 21:26.35 | Katty | Penguin: still have the same po box as last year? |
| 21:26.35 | Penguin | I can't promise to return one -- I'm not good with sending cards to anyone. |
| 21:26.38 | Penguin | Yep! |
| 21:26.42 | Katty | excellent |
| 21:26.46 | Katty | do you still have my address? |
| 21:26.52 | Katty | if you get around to sending one, of course. |
| 21:26.58 | Penguin | Let me check. |
| 21:27.35 | Penguin | W* Drive? |
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| 21:28.51 | Katty | Penguin: yes'r |
| 21:28.58 | Penguin | Yep, got it. |
| 21:29.17 | Katty | excellent |
| 21:29.35 | Penguin | Got to run... see you later. |
| 21:29.39 | Katty | buhbye |
| 21:30.07 | *** join/#asterisk iggi (~iggi@ip69.50-31-21.static.steadfastdns.net) |
| 21:42.24 | Qwell | Katty: always |
| 21:43.09 | Katty | perfect. |
| 21:43.28 | Katty | still on weatherstone? |
| 21:43.35 | Qwell | mmh |
| 21:44.06 | Katty | Qwell: (is that a yes?) |
| 21:44.16 | Qwell | yar! |
| 21:44.18 | Katty | excellent |
| 21:44.20 | Katty | still have mine? |
| 21:44.31 | Qwell | uhh |
| 21:45.25 | Eric-K | malcolmd do you also have a sip.conf in place? |
| 21:45.31 | Eric-K | because i don't have anything sip running |
| 21:45.33 | Eric-K | just pjsip |
| 21:45.44 | malcolmd | nope, chan_sip is explicitly disabled |
| 21:46.13 | Eric-K | might have to do with my current pjsip config |
| 21:46.20 | Eric-K | it's basic basic, taken from the wiki |
| 21:46.52 | Eric-K | using "A SIP trunk to your service provider, including outbound registration" from https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples |
| 21:47.01 | Eric-K | replaced with credentials of my sip trunk |
| 21:49.28 | malcolmd | i've used a very similar example where i'm dialing from a digium phone out *-trunk (revision mentioned previously) using a very similar configuration example to dial out a digium sip trunk and the outbound invite correctly shows the User-Agent string w/ my version of Asterisk. |
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| 21:56.22 | Eric-K | I really wonder what I'm doing wrong. |
| 21:56.23 | Eric-K | hah |
| 22:01.07 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 22:02.16 | malcolmd | i dunno. if i want to manipulate it i can make an object, e.g. [global] type=global user_agent=Whee and that's the User-Agent string on outbound requests. if i don't define it, it's my Asterisk version. *shrug* |
| 22:03.07 | Eric-K | hehehe |
| 22:03.13 | Eric-K | let me quickly try the global one |
| 22:04.12 | Eric-K | That works. Adding [global] and including type=global. |
| 22:04.20 | Eric-K | I tried that before without type=global |
| 22:05.02 | malcolmd | huzzah |
| 22:06.51 | Eric-K | From: "sipvicious"<sip:100@1.1.1.1>;tag=62636532653163353133633401333438303432343032 |
| 22:06.54 | Eric-K | lol |
| 22:07.22 | Eric-K | just scrolling though my wireshark |
| 22:08.43 | Eric-K | people abusing poorly secured servers |
| 22:09.55 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 22:20.20 | Eric-K | next up, dns srv |
| 22:20.46 | Eric-K | sip trunk provider has a domain with dns srv, pjsip does not recognize that so i guess i need to add some setting |
| 22:21.04 | Eric-K | maybe no port |
| 22:25.42 | Eric-K | yes, works without a port |
| 22:25.53 | Eric-K | needs the : at the end of the domain tho |
| 22:27.58 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 22:30.56 | *** join/#asterisk MadHatter42 (~tuwid@217.73.143.43) |
| 22:34.25 | smkelly | glares at file |
| 22:42.35 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-rqjdalbzzltcfxff) |
| 22:42.35 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:53.39 | *** join/#asterisk jhlavacek (~jirka@jix.nextradsl.cz) |
| 23:01.35 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 23:01.43 | *** part/#asterisk alexw (~textual@unaffiliated/alexw) |
| 23:35.45 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 23:44.27 | ChannelZ | Printing out ${RTPAUDIOQOS} in my h exten it runs on the 'local side' channel.. how do I look at the 'remote channel'? IE SIP/Foo dials a number which goes to SIP/Provider, and I want to look at the stats for SIP/Provider when the call ends |
| 23:48.41 | mjordan | ChannelZ: Hangup handler |
| 23:48.55 | mjordan | ChannelZ: attach a hangup handler to the channel using a pre-dial routine. |
| 23:48.57 | rmudgett | ChannelZ: Use a pre-dial handler to setup a hangup handler on the SIP/Provider |
| 23:49.02 | mjordan | rmudgett: jinx |
| 23:49.44 | rmudgett | You have to be on v11+ for pre-dial and hangup handlers |
| 23:53.36 | ChannelZ | ..reading on the Wiki.. thanks |
| 23:53.55 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 23:54.13 | ChannelZ | and is there no way yet with PJSIP channels to get the RTCP status? Everything I've tried comes up blank |
