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| 00:34.44 | mjordan | pjensen00: 5 & 5. This has good instructions on gathering a lot: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 00:34.51 | mjordan | er, gathering a log |
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| 03:58.58 | tulga | I'm planning to upgrade my server. CPU important or RAM important to handle high usage? |
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| 08:40.26 | m0sphere | using phpagi, i can read the data from the EXTEN variable, and the result is the last pressed key/tone. how can i programatically reset/change/clear the value? |
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| 09:29.14 | Ibrahim22 | Hi everyone! Quick question (I hope). I am trying to play an announcement once and 5 seconds after the caller joins the queue. Is this possible? I have tried every combo with the announce-* and queue-* options in queues.conf but still it doesn't work |
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| 09:47.16 | Stefan27 | Hi, is there any option to make Dial(SIP/A&SIP/B) stop calling B if A reports busy and vice versa? |
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| 10:49.46 | tparcina | What's the diference between http://packages.asterisk.org/centos/centos-asterisk-11.repo and http://packages.asterisk.org/centos/centos-asterisk.repo |
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| 11:26.52 | BakaKuna | tparcina: According to the wiki, the latter serves beta releases. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-ActiveRepositories |
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| 12:48.06 | workingcats | having a bit of a problem here with my asterisk. it's sitting behind a NAT, on the one side it connects to sipgate through that NAT, on the other there's softphones on the same network. now when i try to make a call it connects and works well, except after ~30s it hangs up saying "Retransmission timeout reached on transmission 54bbe2ac-9c06-e411-853f-001e0b25ed8b@ob1 for seqno 3 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST |
| 12:48.06 | workingcats | /SIP+Retransmissions" and "Hanging up call 54bbe2ac-9c06-e411-853f-001e0b25ed8b@ob1 - no reply to our critical packet (see [same link])." |
| 12:48.06 | workingcats | so far so good, the thing is if i make a call from another internal machine but from a different network the calls last as long as i want. any ideas what the cause could be? |
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| 12:48.44 | workingcats | normally i would suspect the NAT between sipgate and my *, but since i can make calls from another network that can't really be it can it? |
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| 13:06.10 | eppigy | hello |
| 13:06.11 | eppigy | i am dave |
| 13:18.26 | workingcats | ok progress, now i get identical behaviour from both networks.. now it doesnt work from either ^^ |
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| 13:40.28 | cusco | hi |
| 13:40.51 | WIMPy | lo |
| 13:41.13 | cusco | is there a way I can turn off 2 PRI spans without affecting other ongoing calls on other spans? |
| 13:41.56 | WIMPy | Didn't removing them from the config work? |
| 13:42.13 | cusco | by config you mean chan_dahdi.conf =? |
| 13:42.38 | WIMPy | I think that might be enough, yes. |
| 13:42.43 | cusco | to reload that config won't it drop ongoing calls on active spans? |
| 13:43.06 | WIMPy | It will. |
| 13:43.20 | cusco | I mean, on other spans that I do not wish to turn off |
| 13:43.25 | WIMPy | One of the many reasons, I prefer not to use dahdi. |
| 13:43.34 | cusco | what do you use instead? |
| 13:43.46 | cusco | (out of curiosity) |
| 13:43.46 | WIMPy | mISDN |
| 13:43.56 | WIMPy | But that's DSS1 only. |
| 13:44.25 | cusco | you lst me... |
| 13:45.03 | cusco | and out of curiosity how would I know if our PRI lines would support it? |
| 13:45.27 | WIMPy | By not being in North America, I guess :-) |
| 13:45.29 | igcewieling | A reload will not drop calls. However, you can't add or remove channels on a reload. |
| 13:45.37 | WIMPy | Rest of the world. |
| 13:46.02 | cusco | europe |
| 13:46.31 | cusco | igcewieling: so I can't turn off 2 spans without affecting other ongoing calls (and without going there and unplug the cable) |
| 13:46.37 | igcewieling | I believe there is also "dahdi destroy channel X" |
| 13:46.54 | WIMPy | I'm not sure if Japan still uses their national stuff. But it's mor like most of the world. That's why the E was dropped long time ago. |
| 13:46.59 | igcewieling | cusco: generally people are not crazy enough to expect something like that to work. |
| 13:47.12 | [TK]D-Fender | destroy = bad |
| 13:47.22 | WIMPy | thinks cusco has a good reason. |
| 13:47.26 | [TK]D-Fender | you have to reload chan_dahdi.so to recover. |
| 13:47.36 | [TK]D-Fender | he wants so simulate a pulled plug which that won't do. |
| 13:47.46 | igcewieling | As well as "pri service disable channel X" |
| 13:47.53 | [TK]D-Fender | Which means it's port-based, not channel-based |
| 13:48.21 | igcewieling | Ah, no, if you want to simulate a cable being pulled then you pull the cable and stop messing around with stuff which won't help. |
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| 13:50.29 | cusco | our ISP told us that calls would be forwarded to another PRI line when those became unpluged. So I would like to test it ... I'm not phisically near the equipment tho.. |
| 13:50.55 | [TK]D-Fender | Call someone who is |
| 13:50.55 | igcewieling | cusco: then you can't test. simple as that. |
| 13:52.03 | cusco | can't I set a PRI_CAUSE or something ? :P |
| 13:52.11 | cusco | gah, ok |
| 13:52.25 | WIMPy | If you send a cause, that's not a failure. |
| 13:52.46 | WIMPy | You'd need to ignore the call. |
| 13:52.54 | WIMPy | At least. |
| 13:53.25 | igcewieling | cusco: you are testing a simulation, which is rather pointless. |
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| 13:54.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:55.52 | WIMPy | wonders if really just ignoring a call would be enough. But unfortunately that's somethign Asterisk can't do anyway. |
| 13:57.50 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 13:57.57 | vassilux | I try to get callref from ast_channel, there is ast_channel_tech *tech field. Is it a good way ? |
| 13:58.15 | vassilux | to do it ? |
| 13:58.49 | WIMPy | If you're working on the source, try asking in #asterisk-dev. |
| 13:59.27 | vassilux | ok thank |
| 14:00.04 | tparcina | I'm trying to insall asterisk 11 from asterisk-11 repo, and it says that it depends on asterisk-dahdi. |
| 14:00.13 | tparcina | But I don't have digium hardware. |
| 14:00.30 | tparcina | Is it the error, or I'm doing something wrong? |
| 14:01.10 | WIMPy | I'd call it incorrect. It would work fine without dahdi. |
| 14:01.14 | tparcina | Package is: asterisk.x86_64 0:11.10.0-1_centos6 |
| 14:01.55 | tparcina | WIMPy: But it won't install without dahdi, and I don't won't to install dahdi (since I don't need it). |
| 14:02.15 | WIMPy | And there are some bugfixes in 11.10.2, BTW. |
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| 14:02.33 | WIMPy | I'm sure, rpm has some switch to ignore it. |
| 14:02.56 | WIMPy | But if you don't want to install stuff you don't need, building from soure is the way. |
| 14:02.57 | tparcina | WIMPy: Hopefully it will be on repo soon. |
| 14:03.13 | WIMPy | Thre's a LOT of things that are optional. |
| 14:04.28 | tparcina | WIMPy: Do you know where I can check if this dependency is a bug? |
| 14:04.34 | tparcina | Who maintains the package? |
| 14:04.51 | tparcina | Is there a channel or a mailing list? |
| 14:05.03 | WIMPy | I have no idea. |
| 14:05.13 | WIMPy | But shouldn't the package itself tell you? |
| 14:06.53 | tparcina | WIMPy: That's what package says, but I thing it shouldn't be that way. |
| 14:07.07 | tparcina | And that's what I would like to check. |
| 14:08.49 | WIMPy | But doesn't it tell you who the maintainer is? |
| 14:12.16 | tparcina | WIMPy: I don't know how to get that info. |
| 14:13.53 | WIMPy | Maybe the best idea would be to wait in here for someone knowing who to contact. |
| 14:14.01 | WIMPy | Or use the mailing list. |
| 14:22.02 | igcewieling | tparcina: "rpm -qi therpmname" |
| 14:22.11 | igcewieling | If you want to use packages then learn your package manager. |
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| 14:34.49 | _grim_ | hey all.... using queue,,,,,,,,macro. however when the macro runs none of the variables follow to the macro (as expected), however I don't see a way to pass arguments. does anyone know how to do this? |
| 14:36.01 | cusco | set the var with __ |
| 14:37.09 | _grim_ | cusco, thanks! |
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| 14:46.14 | c|oneman | Are any WiFi VoIP phones reasonably reliable? Or is it Dect or gtfo? |
| 14:46.59 | [TK]D-Fender | many have gotten better. WiFi still isn't great on battery life though |
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| 14:53.48 | WIMPy | And jitter on wifi is rather extreme. You definitely want CAT-iq (or DECT). |
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| 15:01.30 | _grim_ | cusco: didn't work... :-( |
| 15:01.40 | _grim_ | i'll pastebin the dialplan. |
| 15:02.32 | _grim_ | http://pastebin.com/Axiz75DJ |
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| 15:07.33 | _grim_ | I've used the spectralink 8002 with good reliability as long as the re-registration time is really short, like 30 seconds I think is where I set it to. excellent voice quality.... |
| 15:07.40 | Penguin | _grim_: You set a variable with the underscore on it, but you refer to it without the underscore. Set(_MyVar=stuff) and ${MyVar}. |
| 15:08.34 | _grim_ | Penguin: oh damn.... duh... Thanks!!! |
| 15:10.10 | Penguin | Setting a variable with one or two underscore(s) allows for variable inheritance in new channels that get spawned off of a parent channel. I'm not so sure that is what your macro needs anyway. |
| 15:11.45 | _grim_ | Penguin: not sure either.... but it seems to work. carries the call id and queue id through from the queue call to the macro. Always open to other suggestions if there's a better way to do it... :-) |
| 15:12.18 | Penguin | If it works, that the important thing. I was thinking it probably wouldn't help. |
| 15:12.22 | _grim_ | c|oneman: I've used the spectralink 8002 with good reliability as long as the re-registration time is really short, like 30 seconds I think is where I set it to. excellent voice quality.... |
| 15:13.03 | _grim_ | c|oneman: client is using cisco aironet access points in their lawfirm... |
| 15:13.36 | _grim_ | c|oneman: I want to say they are like 1200 series... |
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| 15:21.00 | c|oneman | I've had limited luck with android phones, hard to pinpoint the source of random issues with it |
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| 16:43.42 | lirakis | does asterisk support multiple registration for an "extension" and parallel call forking? |
| 16:47.06 | *** join/#asterisk cmendes0101| (~cmendes01@72.1.46.254) |
| 16:48.40 | file | chan_sip doesn't support multiple registrations per configured peer, but you can have multiple peers and treat them as the same entity around places - and it supports calling multiple things at once, first one to answer gets the call |
| 16:50.23 | lirakis | file, so basically create multople peers and fork multiple calls to them as if they were a "group" |
| 16:50.33 | lirakis | file, how about the new pjsip channel driver? |
| 16:50.36 | file | sure |
| 16:50.59 | file | the new PJSIP channel driver allows multiple registrations to the same AOR, if you configure it as such - and there is a dialplan function to allow calling all registered contacts on an AOR |
| 16:51.25 | lirakis | file, does asterisk actually support AOR's instead of just extensions with the pjsip channel? |
| 16:51.36 | lirakis | if so, how does this work in the dialplan? |
| 16:52.03 | file | it conceptually understands AORs but if you call something it's treated as an extension in the dialplan |
| 16:52.05 | lirakis | for extension matching - are there extensions to the ..."language" to match things with more generality and less "extension" |
| 16:52.32 | lirakis | file, ok - well intersting to hear "AOR" mentioned in the context of asterisk at all |
| 16:52.59 | file | the new PJSIP channel driver uses SIP concepts - AORs, contacts, URIs |
| 16:53.06 | lirakis | sure - yeah i am familiar with PJSIP |
| 16:53.27 | lirakis | i just wasnt sure how much of that made it through the API boundry into asterisk itself |
| 16:53.40 | lirakis | since - historically it had no concept of an "AOR" |
| 16:54.05 | lirakis | last question which you may or may not know |
| 16:54.26 | lirakis | do you have any clue whether the grandstream PBX appliances use the chan_sip driver, or the newer pjsip driver? |
| 16:54.33 | lirakis | i would have to guess they are running chan_sip |
| 16:54.39 | lirakis | just ... curious |
| 16:54.45 | file | I think they're based off of 1.8 |
| 16:55.00 | file | you could go grab the source code and look |
| 16:55.04 | lirakis | true |
| 16:55.13 | lirakis | or .. if they modified it at least |
| 16:55.26 | lirakis | ill check their gnu sources page |
| 16:55.31 | lirakis | see if it has a version |
| 16:55.35 | lirakis | file, thanks |
| 16:56.19 | lirakis | Asterisk_UCM6100_R1_0_5_19.zip |
| 16:56.19 | file | briefly remembers an interview with them and smiles |
| 16:56.20 | lirakis | heh |
| 16:56.41 | lirakis | yeah - ive been getting beta stuff from their new beta program, and they are doing a new PBX. |
| 16:56.59 | lirakis | ive never used one of their pbx devices before - so just curious about it |
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| 16:58.29 | lirakis | file, yeah 1.8.1.23-1 |
| 16:58.40 | lirakis | or at least - thats the summary file |
| 16:58.42 | lirakis | ok - thanks again |
| 16:59.14 | file | wonders if they've put security patches in |
| 16:59.20 | lirakis | probably not |
| 16:59.20 | lirakis | heh |
| 16:59.24 | lirakis | idk really |
| 16:59.40 | lirakis | like i said - never used any of their pbx stuff |
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| 16:59.51 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 17:11.57 | *** join/#asterisk qakhan (~qakhan@50-204-254-12-static.hfc.comcastbusiness.net) |
| 17:13.10 | qakhan | i am trying to connect to mssql using odbc to get record in dialplan |
| 17:13.31 | qakhan | but i am getting following error in cli. |
| 17:13.32 | qakhan | res_odbc.c:1562 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Can't open lib '/usr/lib/libtdsodbc.so' : file not found |
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| 17:16.13 | mbowie | qakhan: Sounds like libtds isn't installed. |
| 17:18.33 | qakhan | mbowie i tried to install but No package libtds available. |
| 17:18.41 | qakhan | i am using CentOS 6.5 |
| 17:18.46 | mbowie | qakhan: Try freetds |
| 17:18.52 | qakhan | nop |
| 17:19.27 | mbowie | Well, you have to have some kind of client library to go between Asterisk and your server, so you'll need something. |
| 17:19.39 | qakhan | it is also not available i installed freetds through ftp |
| 17:20.41 | mbowie | qakhan: I just Googled "centos 6.5 freetds" and the first result is instructions on installing it. |
| 17:21.05 | qakhan | http://www.freetds.org/software.html |
| 17:22.10 | qakhan | i followed http://voidtech.wordpress.com/2013/11/06/centos-6-configure-freetds/ |
| 17:22.36 | [TK]D-Fender | So go find your files |
| 17:23.35 | eppigy | hello [TK]D-Fender |
| 17:23.42 | [TK]D-Fender | epphello dave |
| 17:24.15 | Katty | DAVE |
| 17:24.19 | Katty | hugs eppigy |
| 17:24.29 | Katty | hi fenderbender. |
| 17:24.43 | [TK]D-Fender | 's autocomplete is actually broken.... |
| 17:25.32 | Katty | you gotta stop breaking things, dear. |
| 17:25.38 | Katty | this is why we can't have nice things. |
| 17:27.49 | qakhan | [TK]D-Fender libtdsodbc.so file is there |
| 17:28.07 | [TK]D-Fender | kattand did you run ldconfig? |
| 17:28.26 | [TK]D-Fender | qakhan: and did you run ldconfig? |
| 17:28.31 | Katty | i sure didn't! |
| 17:28.44 | Katty | but i did perform an in place upgrade of an eset agent from labtech |
| 17:28.53 | Katty | which might include something like ldconfig |
| 17:29.38 | qakhan | i didnt |
| 17:29.41 | qakhan | i try it |
| 17:30.59 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-adynhblrbnfbcrmd) |
| 17:33.38 | qakhan | same message |
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| 17:34.27 | [TK]D-Fender | show us the files |
| 17:36.22 | qakhan | which files? |
| 17:43.15 | [TK]D-Fender | your so's |
| 17:43.19 | [TK]D-Fender | and the configs |
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| 18:01.22 | qakhan | http://pastebin.com/xPPa5aYX |
| 18:01.29 | qakhan | http://pastebin.com/vg9LaehR |
| 18:01.36 | qakhan | http://pastebin.com/z8swyYyK |
| 18:03.28 | mbowie | qakhan: And /usr/lib/libtdsodbc.so exists? |
| 18:03.42 | [TK]D-Fender | dump that whole folder's file list |
| 18:05.20 | qakhan | http://pastebin.com/DavjMwhY |
| 18:05.28 | qakhan | mbowie yes |
| 18:07.00 | [TK]D-Fender | ls -la /usr/lib |
| 18:07.02 | [TK]D-Fender | ^^^^ |
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| 18:08.23 | qakhan | http://pastebin.com/3s5PYH3m |
| 18:09.34 | [TK]D-Fender | 1 thing that your permissions don't match the other non-symlinked files |
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| 18:11.15 | qakhan | ok, so libtdsodbc.so does not has permission? |
| 18:11.29 | [TK]D-Fender | look at the files there |
| 18:11.39 | [TK]D-Fender | I just told you somehting you don't have to go beyond that last pastebin to see |
| 18:12.02 | [TK]D-Fender | Then go test ODBC separate from Asterisk |
| 18:12.58 | *** join/#asterisk izbushka (~mirage@176.120.113.118) |
| 18:13.03 | izbushka | hi |
| 18:14.51 | izbushka | how could I exec function from agi script with argument contening comma? |
| 18:15.42 | izbushka | bash agi script: echo EXEC DongleSendSMS GSM,$PHONE,\"$text\" |
| 18:16.05 | izbushka | it ignores backslashes for some reason.. |
| 18:21.47 | pjensen00 | mjordan: I submitted the bug report, ASTERISK-24020 |
| 18:24.28 | mbowie | izbushka: Just a guess, but try wrapping the echo string in non-eval marks, so: echo 'EXEC DongleSendSMS GSM,'$PHONE',"'$text'"' |
| 18:24.54 | mbowie | (Totally making that up on the fly... it could use a sanity check. |
| 18:25.11 | izbushka | mbowie, thanks, trying.. |
| 18:27.39 | izbushka | command: echo 'EXEC DongleSendSMS MTS-GSM,+38'$PHONE',"'$TEXT'"' where $TEXT is "example,text" |
| 18:28.02 | izbushka | got AGI Script Executing Application: (DongleSendSMS) Options: (GSM,+38xxx,text,example) |
| 18:28.16 | izbushka | it ignores quotes |
| 18:30.20 | mbowie | Again, I'm shooting from the hip, but with that echo syntax, I'd try adding escaping now and see if that makes any difference. |
| 18:34.57 | mjordan | pjensen00: thanks |
| 18:36.31 | izbushka | well i've got this result running agi script from shell: EXEC DongleSendSMS MTS-GSM,+380xxxx,"text,example" |
| 18:36.56 | izbushka | looks good, but asterisk still ignores quotes |
| 18:37.19 | izbushka | what the output of AGI should look like? |
| 18:37.40 | igcewieling | You may need to escape the quotes |
| 18:39.18 | *** join/#asterisk nicknam1232 (02798f28@gateway/web/freenode/ip.2.121.143.40) |
| 18:40.06 | izbushka | well, EXEC DongleSendSMS MTS-GSM,+380xxxx,\"text,example\" works.. Thanks igcewieling, mbowie |
| 18:40.09 | nicknam1232 | What base os you like run asterisk ? |
| 18:45.15 | [TK]D-Fender | GNU/Linux |
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| 18:55.22 | savinger | How can I automate adding users or extensions to asterisk? Through say the AMI? |
| 18:57.10 | qakhan | [TK]D-Fender i uninstalled freetds and installed again |
| 18:57.41 | qakhan | now there is no libtds***.so fiels |
| 18:58.11 | [TK]D-Fender | savinger: Edit the config file or use realtime databases for it. |
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| 19:51.10 | TazzNZ | morning all |
| 19:52.04 | mbowie | o/ |
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| 20:04.44 | nicknam1232 | bye [bi] |
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| 20:11.59 | lirakis | any known issues with new browser changes and webrtc ? |
| 20:12.12 | lirakis | for example, chrome's longer icepwd frag requirement? |
| 20:12.18 | TazzNZ | lirakis: which browser? |
| 20:12.19 | lirakis | is that patched in a release? |
| 20:12.29 | lirakis | TazzNZ, chrome beta |
| 20:12.39 | lirakis | possibly chrome as well |
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| 20:30.55 | lirakis | i ask b/c we have been getting recent reports of the following on the cli for webrtc calls |
| 20:30.59 | lirakis | chan_sip.c:10512 process_sdp: Rejecting secure audio stream without encryption details: audio 52559 RTP/SAVPF 111 103 104 0 8 106 105 13 126 |
| 20:31.36 | lirakis | we were not aware of any SDP changes by browsers |
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| 20:37.30 | mjordan | lirakis: which version of Asterisk are you using? |
| 20:39.35 | lirakis | i believe the issue was reported with chrome against asterisk 11.11 |
| 20:39.38 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.11.0 (2014/07/10), 1.8.29.0 (2014/07/10); Standard: Asterisk 12.4.0 (2014/07/10); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 20:39.48 | lirakis | rc1 |
| 20:41.04 | mjordan | lirakis: if you get that message, it's because something offered us SAVP/SAVPF and didn't provide encryption credentials, which should be (IIRC) SDES-SRTP. DTLS should be using a media type of UDP/TLS/RTP/SAVP |
| 20:41.15 | mjordan | in your snippet, the browser is offering the media type incorrectly. |
| 20:41.19 | mjordan | (shocker) |
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| 20:41.56 | lirakis | honestly - im not the person working the issue, just popped in to see if there were any known recent issues. |
| 20:41.57 | mjordan | however, we added a compatibility option to sip.conf to work around the browser's poor compliance with the RFC in this case, avpf=yes |
| 20:42.06 | lirakis | we develop sip.js, and a user reported this |
| 20:42.20 | lirakis | we are striving for compatability (and have tested) against asterisk with chrome |
| 20:42.26 | mjordan | lirakis: gotchya. I'd suspect firefox, since that was the one that we knew of that offered things incorrectly |
| 20:42.37 | lirakis | and are having trouble tracking down the problem |
| 20:42.40 | mjordan | you may want to see what the sip.conf entries are for force_avp and avpf |
| 20:42.49 | lirakis | yeah i mean i know firefox did offer rtp incorrectly |
| 20:43.03 | lirakis | i cant find the issue on github or our google group |
| 20:43.08 | mjordan | hm |
| 20:43.08 | lirakis | otherwise id have more info he he |
| 20:43.16 | mjordan | ah, mystery bugs :-) |
| 20:43.18 | mjordan | the best kind |
| 20:43.40 | mjordan | so yeah, typically Asterisk will expect to see SDES-SRTP encryption keys in the offer when it gets SAVP/SAVPF |
| 20:43.49 | lirakis | yeah we have tested against 11.9.0 in our guides |
| 20:43.51 | lirakis | http://sipjs.com/guides/server-configuration/asterisk/ |
| 20:44.11 | mjordan | the force_avp option will make it look for DTLS encryption even when offered RTP/AVP, RTP/AVPF, RTP/SAVP, or RTP?SAVPF |
| 20:44.20 | lirakis | ill dig more into it tomorrow when the other dev comes back on line - but i guess his laptop was dying at a conference he he |
| 20:44.36 | mjordan | lirakis: for this kind of question, you may want to jump in #asterisk-dev |
| 20:44.59 | lirakis | mjordan, ok will do for tomorrow |
| 20:45.00 | lirakis | thanks |
| 20:45.24 | file | that option doesn't exist in 11.9.0, just in 11.11.0 that was just released |
| 20:46.00 | lirakis | yeah in the guide we have avpf=yes and encryption=yes |
| 20:46.01 | mjordan | file: <lirakis> i believe the issue was reported with chrome against asterisk 11.11 |
| 20:46.31 | lirakis | yeah ill check it out tomorrow - just wrapping up for the day now |
| 20:47.31 | file | offers mjordan SDES with a transport of UDP/TLS/RTP/SAVP |
| 20:47.39 | mjordan | heh |
| 20:47.46 | mjordan | All we're proving is that media types are useless |
| 20:47.55 | mjordan | it should just be "here's stuff I want, you go figure it out" |
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