| 00:02.24 | *** join/#asterisk ericNL (eric@eyeswideshut.xs4all.nl) |
| 00:11.38 | *** join/#asterisk deweydb (~deweydb@S0106881fa12a6aab.no.shawcable.net) |
| 00:11.43 | deweydb | hi! |
| 00:12.48 | deweydb | i'm setting up a new pbx. and i've got audio working for extension to extension calls, and i can hear audio on outbound calls. but i can't hear anything on inbound calls, just silence. i assume maybe this is a networking issue, and i need to expand my RTP port forward range, but i can't find anywhere online what the range should actually be? right now i've got: 10,000 - 40,000 |
| 00:13.12 | Samot | deweydb: You set the RTP port range. |
| 00:13.30 | deweydb | in my softphone? or in the pbx configuration? |
| 00:14.41 | Samot | In Asterisk. |
| 00:15.00 | Samot | In rtp.conf |
| 00:15.20 | Samot | Is Asterisk behind NAT? |
| 00:15.47 | deweydb | yes |
| 00:16.07 | Samot | And you have all the external IP settings correct? |
| 00:16.15 | Samot | As well as the local networks? |
| 00:19.27 | deweydb | i believe so. |
| 00:19.52 | deweydb | i can provide logs of the issue calls if that helps? |
| 00:23.06 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 00:33.55 | *** join/#asterisk ericNL (eric@eyeswideshut.xs4all.nl) |
| 00:33.59 | WIMPy | Hi [sr]! Hi wyoung! |
| 00:35.58 | *** part/#asterisk kharwell (kharwell@nat/digium/x-xpzechkbejwhshkp) |
| 00:49.24 | deweydb | here's a log file: http://paste.ubuntu.com/23601056/ |
| 00:54.23 | *** join/#asterisk deweydb_ (~deweydb@S0106881fa12a6aab.no.shawcable.net) |
| 01:03.56 | *** join/#asterisk Mango45 (~Graham@d198-166-241-4.abhsia.telus.net) |
| 01:30.41 | kfife | If I'm passing a tech/exten stringto AMI (originate), say SIP/1234, can I specify multiple the way I would do with DIAL()? e.g. SIP/1234&SIP/4567 |
| 01:31.19 | kfife | I've tried but it doesn't seem to work. Do I maybe need to escape the "&" or some such? |
| 01:32.11 | kfife | | maybe? |
| 01:33.25 | kfife | DOC says "Same as you'd specify in the dial command" |
| 01:34.27 | kfife | wonders |
| 01:45.37 | *** join/#asterisk coreyfarrell (~coreyfarr@24-177-86-98.dhcp.nwtn.ct.charter.com) |
| 01:47.02 | *** part/#asterisk coreyfarrell (~coreyfarr@24-177-86-98.dhcp.nwtn.ct.charter.com) |
| 01:47.24 | *** join/#asterisk BeeBuu (~beebuu@116.20.65.54) |
| 01:50.42 | BeeBuu | hi,all. |
| 01:52.39 | BeeBuu | i got a problem , i had send direct_meida=yes in endpoint section,but i found the media still though the server!! Is any thing else need to set ? |
| 01:52.52 | BeeBuu | direct_media |
| 01:55.16 | BeeBuu | can someone help me,please? |
| 02:15.36 | denysonique | I have to softphones, if I dial into an echo test everything works fine. However if I dial one phone to the other both phones cannot hear each other |
| 02:32.44 | kfife | NAT |
| 02:33.33 | kfife | your media is probably not being passed because of NAT |
| 02:39.46 | BeeBuu | kfife,clients and server is in a same net, 192.168.X.X |
| 02:40.58 | BeeBuu | the problem is no NAT at all~~~ |
| 02:41.58 | kfife | Same Ethernet Broadcast Domain? |
| 02:42.07 | kfife | WIFi? |
| 02:43.04 | kfife | Are you on Wi-Fi? |
| 02:44.32 | BeeBuu | one is wifi,and one is eth |
| 02:44.55 | BeeBuu | <PROTECTED> |
| 02:44.56 | BeeBuu | <PROTECTED> |
| 02:44.56 | BeeBuu | <PROTECTED> |
| 02:44.56 | BeeBuu | <PROTECTED> |
| 02:44.58 | BeeBuu | <PROTECTED> |
| 02:45.00 | BeeBuu | <PROTECTED> |
| 02:45.02 | BeeBuu | <PROTECTED> |
| 02:45.24 | BeeBuu | cli show like that |
| 02:46.06 | BeeBuu | clients connected to a same router |
| 02:46.12 | kfife | Both phones registered to AsterisK? |
| 02:46.25 | BeeBuu | sure,both |
| 02:46.38 | kfife | phone calling asterisk, asterisk calls other phone? |
| 02:46.50 | BeeBuu | yeah |
| 02:46.52 | kfife | or calling phone to phone? |
| 02:46.58 | kfife | using DIal? |
| 02:47.15 | BeeBuu | <PROTECTED> |
| 02:47.18 | BeeBuu | <PROTECTED> |
| 02:47.24 | BeeBuu | yes,using dial |
| 02:48.05 | BeeBuu | i had no idear about this for 2 days~~~ |
| 02:48.08 | kfife | Never used PJSip |
| 02:48.15 | kfife | Only familiar with CHan_sip |
| 02:49.25 | BeeBuu | anyone can help me please~~~~ |
| 02:49.42 | BeeBuu | it make me crazy~~~ |
| 02:49.44 | kfife | NOt sure what's going on |
| 02:50.11 | kfife | firewall disabled on Asteris? |
| 02:51.19 | BeeBuu | no firewall on it |
| 02:51.49 | kfife | Hmm... Maybe burn it down and start over. :-) |
| 02:51.53 | kfife | I'm stumped |
| 02:52.20 | BeeBuu | :( |
| 02:52.25 | kfife | Try Chan_SIP |
| 02:52.29 | kfife | ? |
| 02:53.16 | BeeBuu | i found chan_sip only can tranfer low video media~~ |
| 02:54.53 | Samot | direct_media in PJSIP is by default "yes" |
| 02:55.15 | BeeBuu | Samot,anything else i need to set? |
| 02:57.07 | Samot | What makes you think the media isn't direct between the two endpoints? |
| 02:57.59 | BeeBuu | cause i monitor the net traffic |
| 02:58.57 | BeeBuu | i make video call,and it took away 2MBits/s |
| 02:58.59 | Samot | So you never see Asterisk drop out of the media path? |
| 03:00.18 | BeeBuu | cli show follow |
| 03:00.18 | BeeBuu | <PROTECTED> |
| 03:00.18 | BeeBuu | <PROTECTED> |
| 03:00.19 | BeeBuu | <PROTECTED> |
| 03:00.21 | BeeBuu | <PROTECTED> |
| 03:00.39 | BeeBuu | so i think it wasn't direct |
| 03:02.33 | Samot | No. |
| 03:02.44 | Samot | What's the IP of Asterisk? |
| 03:02.54 | Samot | Is it .64 or .183? |
| 03:02.59 | BeeBuu | 192.168.2.121 |
| 03:03.05 | BeeBuu | not 64 or 183 |
| 03:03.10 | Samot | So how does that show it's in the path? |
| 03:03.15 | Samot | It shows the IP of the endpoints. |
| 03:03.57 | BeeBuu | u mean it's aleady direct ? |
| 03:04.28 | Samot | Asterisk HAS to be in the media path at the start of the call. It has to bridge to two channels together. Asterisk will drop out of the media path when the call is established and media is flowing.. |
| 03:05.41 | BeeBuu | but i monitor the net,and it really took away 2MBits/s |
| 03:05.49 | Samot | A video call? |
| 03:05.50 | Samot | Of course. |
| 03:05.53 | Samot | It's a VIDEO call. |
| 03:06.14 | BeeBuu | video call is can not be direct? |
| 03:06.40 | Samot | So far you have shown nothing to prove that the media is flowing through Asterisk during the call. |
| 03:06.56 | Samot | Or not flowing through Asterisk. |
| 03:07.15 | Samot | By default direct_media=yes and direct_media_type=invite |
| 03:08.30 | BeeBuu | i think, if not flowing through *, how came that 2MBits/s ? |
| 03:08.42 | BeeBuu | and called over, no traffic |
| 03:09.03 | Samot | So you saw a call and it two 2Mbps? |
| 03:09.13 | Samot | And when the call was over the traffic dropped? |
| 03:09.19 | Samot | And you saw this on your network? |
| 03:09.29 | BeeBuu | yes,i got that |
| 03:09.47 | Samot | s/two/was |
| 03:10.32 | BeeBuu | it was 2M when calling |
| 03:10.46 | BeeBuu | and i saw that on server machine~~ |
| 03:11.00 | Samot | Well the calls have to go through Asterisk. |
| 03:11.14 | Samot | Period. |
| 03:11.43 | BeeBuu | [12351] |
| 03:11.46 | BeeBuu | type=endpoint |
| 03:11.48 | BeeBuu | context=beebuu |
| 03:11.49 | Samot | Don't. |
| 03:11.50 | BeeBuu | disallow=all |
| 03:11.51 | Samot | ?pb |
| 03:11.52 | BeeBuu | allow=g722,ulaw,h264 |
| 03:11.53 | Samot | ~pb |
| 03:11.53 | infobot | extra, extra, read all about it, pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 03:11.54 | BeeBuu | transport=transport-udp |
| 03:11.56 | BeeBuu | auth=12351 |
| 03:11.58 | BeeBuu | aors=12351 |
| 03:11.59 | Samot | Don't flood the channel. |
| 03:12.00 | BeeBuu | direct_media_method=invite ; Direct Media method type (default: "invite") |
| 03:12.01 | Samot | Stop. |
| 03:12.02 | BeeBuu | connected_line_method=invite ; Connected line method type (default: |
| 03:12.04 | BeeBuu | direct_media=yes |
| 03:12.04 | Samot | Stop. |
| 03:12.06 | BeeBuu | sorry, |
| 03:12.18 | Samot | I didn't ask for the endpoint settings. |
| 03:12.46 | Samot | Again, calls have to pass through Asterisk. |
| 03:12.47 | BeeBuu | i make that setting just few lines like above |
| 03:12.55 | Samot | All of them. |
| 03:13.21 | *** join/#asterisk klow (~textual@c-67-161-106-89.hsd1.wa.comcast.net) |
| 03:13.23 | BeeBuu | can't direct? |
| 03:13.38 | Samot | The media, yes. |
| 03:13.47 | Samot | Asterisk will proxy the media to the endpoint. |
| 03:14.05 | BeeBuu | so direct_media=yes for what? |
| 03:14.12 | Samot | But when your device is registered to Asterisk, when you dial 12345 it's going to 12345@AsteriskIP |
| 03:14.32 | Samot | direct_media means "Asterisk please proxy the media and don't touch it." |
| 03:15.12 | Samot | However, when you do things like "Transfer" or "Hold" Asterisk interjects itself for the call. |
| 03:15.29 | BeeBuu | don't touch it -- just that? |
| 03:15.42 | Samot | Why do you need direct media? |
| 03:15.46 | Samot | It's a barely used option. |
| 03:17.37 | BeeBuu | i don't want the * proxy all media flow,,it will take server so many resource~~~ |
| 03:18.03 | Samot | How many calls is the server taking? |
| 03:19.00 | BeeBuu | i want to it be service about 1000 endpoints |
| 03:19.15 | Samot | That will be local or remote to the PBX |
| 03:20.21 | BeeBuu | can i do that with a local asterisk server? |
| 03:20.35 | Samot | That wasn't my question. |
| 03:20.51 | Samot | Yes, you can make 4 different /24's local to the Asterisk server. |
| 03:21.01 | Samot | Or an entire class B or whatever. |
| 03:21.18 | Samot | But are the endpoints going to be on the same local networks as the PBX. |
| 03:21.25 | Samot | IE are they going to be behind NAT? |
| 03:22.12 | BeeBuu | i got you. net work no need to think about~~~ |
| 03:22.21 | Samot | Yes, yes it is. |
| 03:22.32 | Samot | Because Direct Media behind NAT = BAD |
| 03:22.57 | BeeBuu | no NAT,and a bigger net than class C |
| 03:23.20 | Samot | What are the resources on the server? |
| 03:23.24 | Samot | CPU, RAM? |
| 03:23.58 | BeeBuu | 32core 32G |
| 03:24.12 | Samot | How many concurrent calls? |
| 03:25.00 | BeeBuu | about 800~900 |
| 03:25.18 | Samot | 90% of the devices will be on a call at any given point? |
| 03:25.55 | BeeBuu | yeah |
| 03:26.34 | BeeBuu | if fact ,all are video call |
| 03:27.23 | BeeBuu | in face ,all are video call |
| 03:27.23 | BeeBuu | in fact |
| 03:28.26 | BeeBuu | so i think if can be direct between endpoints is necessary... |
| 03:28.35 | Samot | Are these all internal calls? |
| 03:28.47 | Samot | Or are there calls from/to the PSTN? |
| 03:29.19 | BeeBuu | it can be between endpoints is necessary... |
| 03:29.20 | BeeBuu | yes ,just a local using. |
| 03:29.20 | BeeBuu | no f/t outside |
| 03:30.35 | Samot | Well an in progress call is going to take less resources than a call setup/tear down |
| 03:32.04 | BeeBuu | how can i fix my problem? any suggestion? |
| 03:32.18 | Samot | What problem? |
| 03:32.24 | Samot | You have yet to show there IS a problem. |
| 03:32.50 | Samot | If direct_media is not specified in the peer config, it's YES by default. |
| 03:33.14 | Samot | Just liked direct_media_type is INVITE |
| 03:34.05 | Samot | And disable_direct_media_on_nat = no So even behind NAT direct media is able to happen. |
| 03:34.11 | Samot | Those are the DEFAULTS. |
| 03:34.31 | BeeBuu | direct_media=yes , and how came that 2M traffic when calling? |
| 03:34.41 | Samot | On the server? |
| 03:34.54 | BeeBuu | on the server! |
| 03:35.01 | Samot | Because that's where the CALLS GO |
| 03:35.18 | Samot | The SIP Signaling is between the device and Asterisk. |
| 03:35.59 | Samot | If you want 100% direct then dial the ext@PHONEIP in the SIP URI dialing. |
| 03:36.25 | BeeBuu | one call get so big singnal? |
| 03:36.35 | Samot | Because its VIDEO |
| 03:36.43 | Samot | It takes up bandwidth. |
| 03:37.03 | BeeBuu | Wo..... |
| 03:37.11 | Samot | Just like a ulaw call is about 90Kbps per channel of the call. |
| 03:37.28 | BeeBuu | one call get so many singal,out of my mind |
| 03:37.31 | Samot | So since it's two channels, it would be 180Kbps. |
| 03:37.41 | Samot | What do you mean so many signal? |
| 03:39.09 | BeeBuu | maybe i make a mistake.let me try to say~~ |
| 03:40.38 | BeeBuu | is direct_media=yes can not be tranfer media flow between endpoints when call at asterisk, is that? |
| 03:41.24 | Samot | direct media when enabled will have Asterisk drop out of the MEDIA path. |
| 03:41.54 | BeeBuu | direct_media=yes just make asterisk do not modify ? |
| 03:42.03 | Samot | Yes. |
| 03:42.07 | Samot | Until needed. |
| 03:42.12 | Samot | Like with a "hold" |
| 03:42.29 | Samot | Or a "transfer" |
| 03:42.53 | Samot | Or anything that requires Asterisk to interject media into the RTP stream. |
| 03:43.05 | BeeBuu | Oh.......all mine fault~~~ |
| 03:43.33 | Samot | As I have said multiple times, it is the DEFAULT behaviour with PJSIP. |
| 03:44.01 | BeeBuu | i misunderstanding direct_media=yes can make the endpoints send media each other |
| 03:44.09 | Samot | YES |
| 03:44.26 | Samot | The media IP in the SDP is the ENDPOINTs. |
| 03:44.40 | BeeBuu | how fool i am .... |
| 03:44.41 | BeeBuu | thanks,Samot |
| 03:45.17 | Samot | But all the SIP SIGNALING is going through Asterisk. |
| 03:46.42 | Samot | And a h264 call can be up to 768Kbps |
| 03:47.22 | Samot | So if all your call are internal that makes a video call between two endpoints is going to be about 1.5Mbps or more. |
| 03:47.54 | Samot | So if all your calls are internal that means a video call between two endpoints is going to be about 1.5Mbps or more. |
| 03:58.21 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 04:01.25 | BeeBuu | tons of thanks again |
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| 04:44.26 | foobarbarian_ | I'm having troubles with inbound calls which are dropped after around 5 seconds. There is voice transmitted and I can hear the beginning of the greeting message. Asterisk hangs up because it's not getting a response to a critical packet. There is no NAT involved and it goes over a private line that the vendor insisted we need to buy. Outbound calls work perfectly fine |
| 04:44.43 | foobarbarian_ | This is the error log: http://pastebin.com/diw4k798 |
| 04:44.55 | foobarbarian_ | And this is the trunk configuration: http://pastebin.com/5TGdgEzr |
| 04:45.01 | foobarbarian_ | Any help would be greatly appreciated. |
| 04:47.33 | foobarbarian_ | [X3000 Huawai @10.202.142.36] --- private line --- [cisco router @192.168.242.2] --- lan --- [asterisk @192.168.242.1] --- lan on second if --- [S4B server @10.64.2.2] |
| 05:02.31 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 06:16.54 | Rasputin3711 | Is the option session-timers=refuse dangerous? |
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| 07:40.56 | foobarbarian_ | Can I prevent asterisk from hanging up when it encounters a non response to some critical packet? (Seeing as the connection is up and I can hear the voice and asterisk then hangs up) |
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| 08:04.37 | wdoekes | foobarbarian_: your Contact is bad. if there is no NAT, you should not send 111.221.x.x in your contact to 10.202.x.x |
| 08:05.51 | wdoekes | that SoftX device will attempt to send the ACK there, and apparently it's not getting through to your Asterisk |
| 08:06.28 | wdoekes | preventing Asterisk from hanging up is not the solution |
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| 08:40.08 | foobarbarian_ | wdoekes: wow thanks, I always thought that ip comes from the other side, but turns out it's on /etc/asterisk/pjsip.transports.conf |
| 08:40.17 | foobarbarian_ | How do I edit the values in that file via gui? |
| 08:41.16 | foobarbarian_ | found it! Sip settings / admin/config.php?display=sipsettings |
| 08:46.10 | foobarbarian_ | wdoekes: I love you man! Figured it out *facepalm* .. it was wrong settings in the global sip settings |
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| 09:10.56 | wdoekes | glad you could help yourself :) |
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| 12:24.52 | *** join/#asterisk DelphiWorld (~VOIPER@openvpn/user/DelphiWorld) |
| 12:24.54 | DelphiWorld | hello |
| 12:25.05 | DelphiWorld | can i enable / disable a module without using menuselect? |
| 12:31.46 | kkocaerkek | without opening menuselect gui? |
| 12:32.09 | DelphiWorld | kkocaerkek: yes... that gui make me mad using a screen reader :P |
| 12:32.16 | kkocaerkek | yes you can |
| 12:32.22 | kkocaerkek | ./configure |
| 12:32.23 | DelphiWorld | how! |
| 12:32.27 | kkocaerkek | make menuselect.makeopts |
| 12:32.32 | kkocaerkek | menuselect/menuselect --enable func_curl --enable res_config_mysql --enable pbx_lua menuselect.makeopts |
| 12:32.38 | kkocaerkek | like that |
| 12:32.46 | DelphiWorld | ah |
| 12:33.37 | DelphiWorld | kkocaerkek: if i do make menuselect, what to use to enter a category? |
| 12:35.47 | kkocaerkek | menuselect/menuselect --enable CATEGORY_NAME menuselect.makeopts |
| 12:36.25 | DelphiWorld | oh ... i am lost kkocaerkek ;) |
| 12:36.34 | DelphiWorld | let me tel you what i want to enable |
| 12:36.56 | DelphiWorld | i want to activate res_http_websocket, res_crypto and chan_sip |
| 12:37.05 | DelphiWorld | and aditionaly pjsip |
| 12:38.34 | kkocaerkek | add them with --enable parameters |
| 12:38.49 | DelphiWorld | to the file menuselect.makeopt |
| 12:39.44 | kkocaerkek | run menuselect/menuselect command with these parameters |
| 12:40.03 | DelphiWorld | kkocaerkek: i think i got you |
| 12:40.04 | kkocaerkek | You should read this https://wiki.asterisk.org/wiki/display/AST/Using+Menuselect+to+Select+Asterisk+Options |
| 12:40.15 | DelphiWorld | ./menuselect enable res_http_websocket |
| 12:40.17 | DelphiWorld | correct? |
| 12:40.20 | kkocaerkek | nop |
| 12:40.35 | kkocaerkek | the last parameter is "menuselect.makeopts" |
| 12:40.40 | kkocaerkek | menuselect/menuselect --enable res_http_websocket --enable res_crypto menuselect.makeopts |
| 12:41.30 | DelphiWorld | it have been 4 years didnt compile asterisk ;) |
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| 12:47.28 | DelphiWorld | thank you a lot kkocaerkek |
| 12:47.54 | kkocaerkek | you are welcome <DelphiWorld> |
| 12:49.27 | *** join/#asterisk zamba (marius@flage.org) |
| 12:51.24 | DelphiWorld | kkocaerkek: menuselect is asking for ncurses, but its allready installed! |
| 12:51.43 | DelphiWorld | i did this: |
| 12:51.59 | DelphiWorld | menuselect/menuselect enable res_http_websocket menuselect.makeopts |
| 12:52.45 | kkocaerkek | did you forget double dash before enable |
| 12:52.48 | kkocaerkek | --enable |
| 12:53.05 | DelphiWorld | kkocaerkek: yes, this is the price of using tts :( |
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| 13:22.39 | Rasputin3711 | Is the option session-timers=refuse dangerous? |
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| 15:25.33 | DelphiWorld | hey |
| 15:25.49 | DelphiWorld | guys when i use sipml5 with asterisk, i get: |
| 15:26.03 | DelphiWorld | [Dec 9 16:24:52] ERROR[7094]: tcptls.c:736 ast_tcptls_close_session_file: ast_tcptls_close_session_file invoked on session instance without file or file descriptor |
| 15:27.21 | Samot | Are you trying to use TLS? |
| 15:27.43 | DelphiWorld | Samot: using secure web socket (wss) |
| 15:27.53 | Samot | So, yes. |
| 15:27.59 | Samot | And you have all the TLS setup done? |
| 15:28.08 | DelphiWorld | yes certs and everything |
| 15:28.35 | Samot | And that error happens when? |
| 15:28.49 | DelphiWorld | when i press login in sipml5 |
| 15:29.10 | *** join/#asterisk rwb (~Thunderbi@204.13.43.166) |
| 15:29.19 | Samot | OK. Show a debug of it. |
| 15:29.22 | Samot | asterisk -rvvvvvvvvv |
| 15:29.30 | DelphiWorld | ok |
| 15:29.33 | Samot | sip set debug on |
| 15:30.43 | igcewieling | for some reason I vaguely remember 'without file or file descriptor' might mean a certificate issue. |
| 15:31.14 | DelphiWorld | strange Samot, all i get in that log is the line i send! |
| 15:33.07 | DelphiWorld | dtlscertfile=/etc/asterisk/keys/asterisk.pem |
| 15:33.23 | DelphiWorld | and that file is present there |
| 15:34.26 | igcewieling | what about tlscertfile= and tlscafile= and dtlscafile= ? |
| 15:35.08 | DelphiWorld | igcewieling, will pastebin |
| 15:35.33 | igcewieling | DelphiWorld: self signed cert? |
| 15:35.56 | DelphiWorld | igcewieling: yes self signed |
| 15:36.15 | igcewieling | ah. you have a lot of work ahead of you. |
| 15:36.25 | DelphiWorld | igcewieling: ;) |
| 15:36.27 | DelphiWorld | igcewieling: http://paste.ubuntu.com/23603838/ |
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| 15:36.41 | igcewieling | I had all sorts of problems with self signed certs. either the phone or asterisk would not like something. |
| 15:36.56 | DelphiWorld | i am using firefox |
| 15:36.59 | Samot | Well if it's self-signed you need to tell sipml5 about the CA's and all that |
| 15:37.16 | igcewieling | yup. |
| 15:37.20 | Samot | So you need to have the cert on the PC running the softclient |
| 15:37.30 | Samot | In order to load it in the client. |
| 15:38.05 | DelphiWorld | Samot: i see |
| 15:38.14 | DelphiWorld | Samot: would sipml5 need to know about self signed??? |
| 15:38.22 | Samot | Yes. |
| 15:38.22 | igcewieling | DelphiWorld: yes! |
| 15:38.30 | Samot | Because you are using a non-public/known CA. |
| 15:38.36 | Samot | It needs to see the CA. |
| 15:38.42 | DelphiWorld | is it pocible to use letsencrypt? |
| 15:38.49 | Samot | Any public CA. |
| 15:38.53 | igcewieling | you could also get a cert from a CA which is already trusted |
| 15:39.02 | Samot | letsencrypt will be one of those. |
| 15:39.03 | DelphiWorld | ok will try letsencrypt! |
| 15:39.52 | igcewieling | verify your browser trusts the CA |
| 15:40.10 | DelphiWorld | yes letsencrypt is trusted |
| 15:40.14 | DelphiWorld | i use it in all my sites |
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| 15:47.43 | DelphiWorld | certbot is installing |
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| 15:58.10 | DelphiWorld | igcewieling: i generated cert, replacing it right now |
| 15:58.54 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 15:58.58 | igcewieling | DelphiWorld: the extent of my help is to point you in the right direction. |
| 15:59.20 | DelphiWorld | igcewieling: yeah. i'm updating you only :P |
| 15:59.39 | DelphiWorld | igcewieling: do i need to specify tlscafile for asterisk? |
| 16:00.20 | DelphiWorld | now i have fullchain.pem |
| 16:01.09 | igcewieling | I had lots of trouble with TLS and SRTP but eventually got it working. However, since we will be fined massive amount of money if we can't handle a lawful intercept order, I decided it was not worth the significant logistical burden. |
| 16:01.46 | igcewieling | $10,000/day is was I was told. |
| 16:01.47 | DelphiWorld | hahaha |
| 16:05.36 | DelphiWorld | strange |
| 16:05.38 | DelphiWorld | same issue guys |
| 16:05.47 | *** join/#asterisk arkadiossi (~Arkadioss@unaffiliated/arkadiossi) |
| 16:16.13 | DelphiWorld | guys: i generated a cert with letsencrypt but asterisk is still telling me that without file or file descriptor |
| 16:21.59 | igcewieling | seems it should be complaining about DTLS? |
| 16:22.18 | DelphiWorld | igcewieling: so what could be wrong? |
| 16:22.24 | DelphiWorld | where's dtls in this case? |
| 16:22.32 | DelphiWorld | i am not in the phase of rtp yet |
| 16:23.20 | DelphiWorld | all i get is |
| 16:23.20 | DelphiWorld | tcptls_close_session_file invoked on session instance without file or file descriptor |
| 16:24.39 | igcewieling | I believe DTLS is signaling. |
| 16:25.01 | igcewieling | try removing tcp and ws from the transports and set tlsenable=no and see what happens. |
| 16:25.05 | DelphiWorld | i think dtls is only for rtp but not sure |
| 16:25.18 | DelphiWorld | hold on |
| 16:26.48 | DelphiWorld | igcewieling: same exact message |
| 16:26.57 | Samot | You may also want to have tlsbindaddr= set. |
| 16:27.08 | Samot | Because from your pastebin, that's completely missing. |
| 16:27.14 | igcewieling | Samot: the tls* settings also apply to DTLS? |
| 16:27.52 | Samot | You know what the TLS in DTLS is for right? |
| 16:28.09 | DelphiWorld | Samot: but i'm doing wss, what have to do with tlsbindaddr? |
| 16:28.14 | DelphiWorld | i'm not using pur sip in tls |
| 16:30.51 | Samot | You're trying to use WebRTC, right? |
| 16:31.09 | DelphiWorld | Samot: yep |
| 16:31.25 | Samot | https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 16:31.32 | Samot | You followed that ^? |
| 16:31.43 | DelphiWorld | Samot: yep lol |
| 16:31.49 | Samot | What browser are you using? |
| 16:32.05 | Samot | In newer versions of Chrome and Firefox (and others) DTLS-SRTP is mandatory. This means you must generate certificates for Asterisk to be able to use WebRTC. See: Secure Calling Tutorial |
| 16:32.21 | Samot | Because you did click on the link in "See: Secure Calling Tutorial"? |
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| 16:32.39 | DelphiWorld | Samot: i'm using firefox |
| 16:33.40 | Samot | OK, so you did follow the "Secure Calling Tutorial"? |
| 16:33.47 | DelphiWorld | yes |
| 16:34.03 | Samot | That states that the tlsbindaddr has to be set? |
| 16:34.15 | DelphiWorld | i didnt verify it |
| 16:34.24 | DelphiWorld | but this is if you want to use the sip tls transport! |
| 16:34.42 | Samot | WebRTC is still using SIP. |
| 16:34.47 | DelphiWorld | yes |
| 16:34.51 | Samot | WS/WSS is a transport. |
| 16:34.52 | DelphiWorld | but not the sip tls transport |
| 16:34.59 | Samot | WSS requires TLS |
| 16:35.07 | DelphiWorld | its using the sip web socket (WS) and secure sip web socket (WSS) transport |
| 16:35.35 | Samot | So the Asterisk Wiki says "You need to follow this tutorial in order for WSS transports to work for WebRTC in sipml5" |
| 16:35.50 | Samot | And you didn't follow all of the instructions. |
| 16:36.13 | DelphiWorld | Samot: will refollow |
| 16:38.00 | Samot | Also, if you're allowing non-TLS connections you need to have TLS listening on a different port. |
| 16:38.26 | DelphiWorld | Samot: the 8088 is for both ws/wss? |
| 16:38.48 | Samot | No, in Chan_SIP in general. |
| 16:38.49 | DelphiWorld | what port should i give to sipml5. the res_http_websocket port or the tls port |
| 16:39.03 | Samot | If you're UDP/TCP traffic is 5060, the TLS traffic needs to be 5061 |
| 16:39.14 | Samot | Or something unique. |
| 16:39.22 | DelphiWorld | forget the normal tls traffic |
| 16:39.29 | DelphiWorld | this is logic that tls need diferent port |
| 16:39.33 | DelphiWorld | what about the wss port? |
| 16:42.20 | DelphiWorld | Samot: you know what |
| 16:42.25 | DelphiWorld | will move to pjsip |
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| 16:44.23 | igcewieling | DelphiWorld: have you considered using a standalone client for testing? |
| 16:44.45 | DelphiWorld | igcewieling: yeah but my main purpose is sipml5 |
| 16:44.49 | DelphiWorld | i have my aastra phon |
| 16:44.51 | DelphiWorld | i have my aastra phone |
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| 17:02.58 | DelphiWorld | is the pjsip module auto loaded? |
| 17:04.30 | [TK]D-Fender | modules.conf <- |
| 17:11.21 | DelphiWorld | Samot: how to make sure res_http_websocket is loaded |
| 17:11.37 | igcewieling | core show modules like X |
| 17:12.21 | DelphiWorld | hey igcewieling no such command core show modules ! |
| 17:12.55 | igcewieling | maybe "core show module like X" . try "help" |
| 17:14.04 | DelphiWorld | http show status |
| 17:14.58 | igcewieling | isn't that for the regular HTTP server in Asterisk? |
| 17:15.27 | DelphiWorld | yes but websocket res is required |
| 17:15.29 | DelphiWorld | and i saw it |
| 17:15.44 | DelphiWorld | http show status give a uri called "ws", but not wss |
| 17:20.44 | DelphiWorld | igcewieling: LOL... i think i found it! |
| 17:21.55 | DelphiWorld | igcewieling: i think http.conf need tls too! |
| 17:22.17 | igcewieling | I wish you the best of luck |
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| 17:37.42 | DelphiWorld | igcewieling: works now |
| 17:37.59 | DelphiWorld | i must declare a tls connection in http.conf |
| 17:38.05 | DelphiWorld | mozilla dont accept ws anymore |
| 17:38.06 | igcewieling | bizarre |
| 17:38.26 | DelphiWorld | firefox or (SIPML5), dont accept ws:// |
| 17:38.28 | DelphiWorld | only wss |
| 17:38.39 | DelphiWorld | but, now calling failed and i must fix... ;( |
| 17:38.54 | DelphiWorld | same error as befaure |
| 17:39.10 | igcewieling | does it work in any other browser? |
| 17:39.23 | DelphiWorld | will try crome |
| 17:41.09 | igcewieling | When working on websockets recently I saw a note about Firefox handling certs differently. I use chrome so didn't pay much attention to it. |
| 17:41.38 | igcewieling | I'm doing non-voip stuff with them. |
| 17:42.01 | igcewieling | a browser based ssh terminal 8-) |
| 17:42.34 | DelphiWorld | hahaha |
| 17:47.21 | DelphiWorld | websocket unresponsive, disconnecting |
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| 18:30.25 | dwheeler | Good day all! |
| 18:31.54 | dwheeler | Is there a way to play audio and record with silence detection on the input channel (like Record()) so that a caller can verbally interrupt a prompt for voice recognition? |
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| 18:39.40 | dwheeler | Is there a way to split a call into the caller's input and output, sending the input stream down one macro, and the output stream down a different macro? |
| 18:41.50 | WIMPy | Not directly. But you can play around with ChanSpy. |
| 18:42.03 | *** part/#asterisk daemon (~daemon@hostname.racing) |
| 18:46.00 | dwheeler | WIMPy, Can ChanSpy listen to only half of a conversation (one way)? |
| 18:46.34 | rrittgarn | core show application ChanSpy |
| 18:46.37 | rrittgarn | look at option o |
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| 18:46.57 | dwheeler | rrittgarn, I'll play with it and see if I can make it do what we need. |
| 18:46.58 | dwheeler | Thank you! |
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| 22:10.48 | rolledgold | quick question, if I get nothing out of the CLI with verbose on is there somewhere else to look why nothing shows up? |
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| 22:11.12 | WIMPy | logger.conf |
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| 22:15.05 | rolledgold | checked that messages => notice,warning,error,verbose |
| 22:15.16 | rolledgold | console => notice,warning,error,debug |
| 22:16.41 | WIMPy | So where is verbose on your console? |
| 22:21.54 | rolledgold | Console Enabled - DEBUG NOTICE WARNING ERROR VERBOSE |
| 22:22.43 | rolledgold | no output |
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| 22:29.05 | rolledgold | any other ideas why output is quiet on the CLI? |
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| 23:03.43 | WIMPy | Nothing happening? No verbose enabled? |
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