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| 01:21.00 | jeffspeff | anyone familiar with provisioning Yealink phones? I'm trying to understand how the phone is supposed to decrypt the aes_key for common.cfg if the key settings are specified within common.cfg |
| 01:21.14 | IndianaTux | Hi all, I'm trying o get video confbridge to work. extension to extension call I get video but the same extension in the bridge don't have any video. Any pointers ? |
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| 05:26.15 | Haris | hello all |
| 05:26.39 | Haris | I have installed 11.7 on a machine. When I connect eyebeam to it, I get no option in eyebeam to type number to call |
| 05:26.49 | Haris | I'v recently updated settings for videosupport |
| 05:27.21 | Haris | I'v just configured 2 extentions, a realm and now videosupport. other than that everything is out of the box |
| 05:27.29 | Haris | on ubuntu 14.4 LTS |
| 05:27.52 | [TK]D-Fender | There is no such thing as "out of the box" |
| 05:28.04 | [TK]D-Fender | Its your job to configure your SIP peer, general settings, diallpan, etc |
| 05:28.26 | Samot | Did Justin Timberlake do a song about something in a box? |
| 05:28.38 | [TK]D-Fender | If eyebeam doesn't let you call then you aren't properly registered or aren't using it right |
| 05:29.13 | [TK]D-Fender | <PROTECTED> |
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| 05:35.17 | Samot | heh. |
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| 06:14.41 | Lope | to move my asterisk to a non standard port, I've tried this: tcpbindaddr=0.0.0.0:1234 bindaddr=0.0.0.0:1234 and confirmed with netstat -lntup that it's listening on 1234. |
| 06:16.38 | Lope | My existing softphones didn't want to play so nicely though. I edited the account details on the softphones and added the port number. On one of the phones I tried removing the account and re-adding it. That got some functionality where the one phone could call echo, and call the other phone etc, but not the other way around yet. |
| 06:17.09 | Lope | I'm just wondering if it's entirely a phone config issue or maybe there's something else that needs to be done to run SIP on a non-standard port on my * server? |
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| 06:31.21 | KValchev | Hello, I am looking for standalone software for outbound campaigns ("Predictive dialer"). Free or commercial with web gui for administrations and detailed reports. Thanks. |
| 06:34.49 | Lope | looks like my * is configured fine. my android softphones have no troubles. |
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| 06:36.12 | KValchev | aslo Preview |
| 06:38.38 | Lope | it seems like on ALL the softphones, the android ones and SFLphone (linux) I had to create a new account to get the connection on the new port number working. then it works fine. |
| 06:39.08 | Lope | Perhaps there are some uneditable settings. If I cared enough I'd look inside SFLphone |
| 06:39.21 | Lope | 's YML config file. But it's okay. As long as it works. |
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| 08:42.51 | Lope | jeez I get weird issues on Asterisk 13.1. WARNING[195]: chan_sip.c:3778 __sip_xmit: sip_xmit of 0x7f5100000910 (len 954) to softphoneip:port returned -2: No such file or directory |
| 08:46.51 | Lope | I was actually doing some testing. I'm finished now. I'm going to ignore this warning and the previous tcptls warning (which I've never had before) until I'm back on the current version, then see what happens. |
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| 08:57.30 | Greenlight | Am I correct in thinking that if I get Skype for Business then I can use Sykpe with Asterisk over SIP ? |
| 08:58.18 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 08:58.27 | Rasputin3711 | No, you cann't. |
| 08:59.06 | Greenlight | Oh :( Is there any way to connect my Skype account to Asterisk ? |
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| 09:01.47 | Rasputin3711 | http://www.mhspot.com/sts/siptosis.html |
| 09:04.12 | Lope | Rasputin3711: cool link. |
| 09:05.39 | Greenlight | Ty looks interesting |
| 09:06.18 | Greenlight | So Skype Connect has been discontinued? |
| 09:08.49 | Lope | https://support.skype.com/en/faq/FA10578/what-is-a-sip-profile |
| 09:09.50 | Lope | https://community.skype.com/t5/Using-Skype-in-your-business/SIP-to-Skype-is-it-passible/td-p/556759 |
| 09:10.04 | Greenlight | Yea, thats what I was looking at earlier, hence my initial question |
| 09:10.07 | Lope | https://support.skype.com/en/faq/FA10572/how-do-i-manage-sip-profiles |
| 09:10.08 | Greenlight | Now I'm totally confused |
| 09:10.22 | Lope | https://www.skype.com/en/features/skype-connect/ |
| 09:10.42 | Greenlight | Yea... that says you *can* use SIP with Skype... |
| 09:10.47 | Lope | https://support.skype.com/en/faq/FA10549/what-is-skype-connect-and-how-does-it-work |
| 09:11.15 | Lope | yaeh, it's all over their site. Surely if they discontinued it, there wouldn't be so many pages saying it's available. |
| 09:11.26 | Greenlight | That's what I'd have thought, yea |
| 09:13.29 | Greenlight | We regularly have customers aboard who insist on only being able to call us over Skype, at the moment it's a pain and I wondered if there was a cheap way to have those calls come in to our PBX |
| 09:13.53 | Lope | maybe you should just punch them in the face? |
| 09:14.05 | Greenlight | Then they'd be ex-customers... |
| 09:14.16 | Lope | I suppose. but hopefully also ex skype users? |
| 09:14.32 | Greenlight | Perhaps :S |
| 09:15.12 | Greenlight | And Skype is a pain for the support staff; they can't use their normal headsets etc |
| 09:16.09 | Lope | Has anyone here got experience with google voice? |
| 09:16.22 | Lope | Are there any benefits to using it? |
| 09:17.46 | Lope | I see google voice offers calling USA and canada for free |
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| 10:21.28 | tparcina | How to enable calls from our web page to our Asterisk? |
| 10:23.57 | tparcina | So that our web page visitors can talk with our support staff. |
| 10:26.05 | Rasputin3711 | webrtc |
| 10:29.01 | tparcina | Rasputin3711: TY, I'll check that one. |
| 10:35.14 | tparcina | Rasputin3711: Are you using webrtc? |
| 10:35.20 | tparcina | Is it production ready? |
| 10:36.43 | Rasputin3711 | We don't use webrtc tech. Yes, it's ready. |
| 10:37.21 | tparcina | Rasputin3711: OK, TY. |
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| 11:32.23 | Alblasco1702 | Hello, i have a problem i have a grandstream ht-503 and a asterisk server connected to getter. as i dial my phonenumber after some time asterisk pickup the call and play the ivr menu. When i hangup the phone asterisks leave the line busy till i restart asterisk. |
| 11:33.26 | Rasputin3711 | What is Asterisk version? |
| 11:34.25 | Alblasco1702 | Rasputin3711, Asterisk 11.7.0~dfsg-1ubuntu1-lmce0~precise1 |
| 11:35.07 | Rasputin3711 | Latest Version - 11.22.0 |
| 11:37.25 | Alblasco1702 | Rasputin3711, will that make a big difference? |
| 11:38.07 | Rasputin3711 | Try to update for the latest version and check again. |
| 11:39.05 | nixnothing | the gift that keeps on giving |
| 11:39.12 | nixnothing | 3 day weekend becomes 4 day week |
| 11:40.20 | nixnothing | arent they on 13? |
| 11:40.40 | Greenlight | 11 is LTS i think |
| 11:40.44 | nixnothing | ah |
| 11:42.57 | Samot | 11 and 13 are LTS. |
| 11:50.28 | nixnothing | thanks Samot. looks like 12 is the only one that wasn't |
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| 11:51.39 | Lope | Is there a way to calculate how many RTP ports my asterisk server needs? Lets say for example I want my asterisk server to receive a call, and make an outgoing call simultaneously. Is that 2 RTP ports? |
| 11:51.45 | nixnothing | wait, so is 'certified asterisk' just digium's package? |
| 11:52.46 | nixnothing | looks like it (higher release frequency)... the name was just trowing me on http://www.asterisk.org/downloads/asterisk/all-asterisk-versions |
| 11:52.53 | nixnothing | need moar coffee |
| 11:53.28 | WIMPy | Lope: Depending on the type of call, you typically need 2-4 ports per call leg. |
| 11:59.04 | Lope | WIMPy: oh, that's surprising. SIP? |
| 12:01.27 | file | nixnothing, certified is released less often and receives much fewer changes |
| 12:03.27 | Lope | WIMPy: is there a diagram or something you can recommend that shows how the data flows? |
| 12:03.38 | Lope | Here it looks like 1 RTP port per call? https://images.duckduckgo.com/iu/?u=http%3A%2F%2F4.bp.blogspot.com%2F_fv-y__irhDM%2FTCeKvSFBYTI%2FAAAAAAAAAF0%2Ft5zc9EQ4kmQ%2Fs1600%2FSIP1.gif&f=1 |
| 12:04.04 | Rasputin3711 | wireshark |
| 12:04.06 | Lope | or per leg I should say |
| 12:04.14 | Lope | Rasputin3711: great idea :) |
| 12:04.38 | file | 2 UDP ports, 1 RTP and 1 RTCP, per call leg - 2 call legs = 4 |
| 12:04.51 | Lope | Oh I see here is another one which shows more ports: https://images.duckduckgo.com/iu/?u=http%3A%2F%2Fblogs.technet.com%2Fblogfiles%2Fisablog%2FWindowsLiveWriter%2FForefrontTMGisSIPaware_7CCA%2Fimage_thumb.png&f=1 |
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| 12:05.44 | Samot | Why would anyone flow audio outside of their PBX? |
| 12:06.52 | Lope | Is the RTCP port unique for each node, or is it shared for all? Because I notice in Asterisk I've defined the RTP port range, but not RTCP? (there's no RTCP file etc) ? |
| 12:07.01 | file | unique. |
| 12:07.06 | file | RTCP is RTP+1 |
| 12:08.26 | Lope | okay so the last diagram I pasted shows 2 UDP ports (1 RTP and 1 RTCP) per leg. But Wimpy said Typically need 2-4 ports per leg? Why would it be more than 2? |
| 12:08.41 | file | if you are carrying multiple media streams. |
| 12:08.43 | Lope | (the last diagram and you) |
| 12:08.58 | file | each stream is a different RTP stream, so gets its own |
| 12:09.08 | Lope | Like audio (2 ports) + video (2 ports) ? |
| 12:09.14 | file | yes. |
| 12:09.29 | Lope | okay. I've not tried video with asterisk. Does anyone use it here? |
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| 12:10.04 | *** part/#asterisk k054 (~k@fsf/staff/k054) |
| 12:14.49 | Lope | What's the default call-limit for sip peers/friends etc? |
| 12:15.16 | WIMPy | none |
| 12:20.29 | Lope | okay. How do you check what your peak number of simultaneous calls is? Is there a logging option for that, when a new peak is reached, it gets logged? |
| 12:22.02 | Samot | That's reporting. |
| 12:22.04 | WIMPy | no |
| 12:22.06 | Samot | That's on you. |
| 12:22.19 | WIMPy | You have to find out yourself. |
| 12:23.32 | Samot | Your peaks will always be changing. |
| 12:23.50 | Samot | Your busy hour is not the same on Monday as it is on Tuesday. |
| 12:23.58 | Lope | What command would you run to see how many calls you have active right now? |
| 12:24.16 | Lope | sip show ... |
| 12:24.31 | WIMPy | core show channels |
| 12:26.45 | Lope | I've JUST started asterisk now. `sip show channels` shows me this: Peer:mysoftphoneIP User/ANR:(None) CallID:534e5a4f354a Format:(nothing) Hold:No LastMsg:Init: OPTIONS Expiry:(empty) Peer:mysoftphonename |
| 12:27.16 | Lope | I'm also getting that tcptls warning I mentioned yesterday, that's probably the lingering call. |
| 12:27.35 | file | sip show channels will show more than voice/video calls |
| 12:27.36 | Lope | still running * 13.1 |
| 12:27.43 | WIMPy | Not all sip channels are calls. |
| 12:27.53 | Samot | ^ That |
| 12:27.57 | Lope | I see |
| 12:28.42 | Lope | `sip show channelstats` looks fun. I'll check this out next time I've got a call going. |
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| 12:30.44 | Samot | Why wait? |
| 12:30.51 | Samot | Make an outbound call, run the command. |
| 12:30.56 | Samot | Make an inbound call, run the command. |
| 12:31.16 | Samot | Learn what the output is so that when you're looking at 25 call stats you understand them. |
| 12:34.28 | Samot | In fact make a call that does a lot of stuff and run the command. |
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| 12:49.48 | Lope | Samot: thanks, I definitely will. I'm a guy who likes to learn and understand everything. I've just got some urgent work to get done now. |
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| 12:59.20 | cervajs2 | somebody using odbc/mysql with 13.9.1 here? |
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| 13:54.30 | Haris | hello all |
| 13:57.45 | Haris | I need to enable ws:// or wss:// access from asterisk. I'v configured in http.conf settings such as [general] \n enabled=yes \n bindaddr=LAN-IP \n bindport=8088 \n enablestatic=yes \n tlsenable=yes \n tlsbindaddr=LAN-IP \n tlscertfile=</path/to/file.pem \n tlsbindaddr=LAN-IP:8089 . Have I missed something ? |
| 13:58.38 | Haris | when I hit ws://LAN-IP:8088 or ws://LAN-IP:8089 or ws://LAN-IP:8088/ws or ws://LAN-IP:8089/ws , Chrome says can't reach this site. it may be down. when I run http://LAN-IP:8088, I get not found page from asterisk |
| 13:59.47 | Haris | I have only installed asterisk from ubuntu 14.4 LTS pkgs and made one or two extensions as per the sip.js page for asterisk ( http://sipjs.com/guides/server-configuration/asterisk/ ) |
| 13:59.54 | Haris | the ws:// part is not working |
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| 14:04.53 | Haris | is there a stun server that we can setup for asterisk on ubuntu 14.4 LTS ? |
| 14:05.08 | Haris | local server |
| 14:11.25 | Haris | coming back in a bit |
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| 14:25.42 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:32.02 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 14:38.02 | cresl1n | waves |
| 14:44.11 | klow | Anyone familiar enough with the SRTP stuff in Asterisk codebase to know how difficult it would be to allow some stronger ciphers to be used? I posted on the mailing list about this. res_srtp.c has a switch/case statement which allows for two 128 bit cipher suites, but several other 256 bit ones are supported in modern endpoints (both physical and soft phones actually) |
| 14:45.06 | cresl1n | Not sure if anybody here has worked on it recently |
| 14:45.07 | cresl1n | file? |
| 14:45.26 | file | I already responded on the -users post |
| 14:46.01 | file | pretty much until you dive in it's hard to know what other places need to be adjusted... |
| 14:46.06 | cresl1n | I'm guessing it wouldn't be *too* hard, mostly gluing in the relevant ciphers from libsrtp |
| 14:46.46 | cresl1n | Last time I poked at that code was a couple of years ago |
| 14:46.54 | file | same |
| 14:47.13 | file | I think when I did DTLS-SRTP was the last time |
| 14:47.17 | file | and that was minimal |
| 14:47.20 | cresl1n | oh |
| 14:47.22 | cresl1n | yeah |
| 14:48.03 | cresl1n | sends file a DTLS-CLIENT-HELLO |
| 14:48.10 | file | ignores cresl1n |
| 14:48.45 | cresl1n | retransmits CLIENT-HELLO |
| 14:49.00 | gtjoseph | does libsrtp have a "supported cipher discovery" function? |
| 14:49.14 | cresl1n | what do you mean? |
| 14:49.32 | gtjoseph | can you call a libsrtp function to get the supportd ciphers? |
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| 14:49.43 | cresl1n | Oh, good question |
| 14:49.45 | cresl1n | not sure :-) |
| 14:49.52 | gtjoseph | looks |
| 14:49.55 | cresl1n | supported ciphers by the remote endpoint or by libsrtp? |
| 14:50.29 | tompaw | Morning |
| 14:51.02 | gtjoseph | morning tompaw... how go your servers? |
| 14:51.05 | tompaw | Check this out guys: http://prntscr.com/bamp9b |
| 14:51.17 | tompaw | left hand side was yesterday with the slin192 bug |
| 14:51.22 | tompaw | right hand side is today :-) |
| 14:51.33 | file | bit better |
| 14:51.50 | tompaw | yeah, like 20-40x better =) |
| 14:51.55 | gtjoseph | yeah, just a bit. :) |
| 14:52.20 | tompaw | we've also set up coredumps, so next time one of them crashes, I'll have sth for you. |
| 14:53.01 | tompaw | btw, I'm thinking of using these: https://www.supermicro.nl/products/system/3U/5039/SYS-5039MS-H12TRF.cfm for our expansion |
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| 14:53.16 | tompaw | asterisk doesn't have any beef with E3-1200 v5, does itg? |
| 14:53.26 | file | Asterisk doesn't care |
| 14:54.48 | tompaw | It feels like audio processing is low-level enough to have favorites. |
| 14:55.37 | gtjoseph | what do you compile on? |
| 14:57.25 | tompaw | as in cpu? E5 |
| 14:57.52 | gtjoseph | yeah, good. same instruction set. |
| 14:58.10 | gtjoseph | that's the only thing I can think of that's make any difference. |
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| 15:04.55 | tompaw | cool, off to see the landlord for the construction plans, looks like we'll have > 5T of hardware in our new server room and want to make sure it won't collapse. |
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| 15:13.08 | xarmiex | anyone familiar with queues/ringing peers that already have a ringing call on them even though ringinuse=no ? |
| 15:13.47 | xarmiex | i just see people with workarounds / call limits etc, but no real fixes |
| 15:14.54 | klow | Do devs accept bribes? re: srtp ciphers . This is something I could talk to my mgmt about, because we want it for our production systems .. |
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| 15:36.40 | tompaw | I think they're called "Bounties" |
| 15:39.23 | jkroon | ok, managed to fix the asterisk issues. the pjproject ones are a little extensive for my current time frames. |
| 15:40.33 | jkroon | wrong # sorry |
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| 15:59.56 | SoBlindWolf | Hey would anyone be able to help me setup my Google Voice number to work with Asterisk and FreePBX? |
| 16:00.52 | SoBlindWolf | I have to go fro now but I will be back |
| 16:00.58 | SoBlindWolf | Switching classes |
| 16:01.03 | gtjoseph | tompaw: you ultimately had to convert one of your local channels to PJSIP to get the performance benefit, correct? |
| 16:01.35 | tompaw | that's right, otherwise the originator made no sense, as the referencing channel was slin192 already. |
| 16:01.51 | gtjoseph | yeah, just double checking |
| 16:05.02 | SoBlindWolf | Okay I am back for a few minutes |
| 16:05.48 | SoBlindWolf | Is GV able to work with Asterisk and FreeBPX? |
| 16:08.24 | *** join/#asterisk jeffspeff (~Jeff@12.49.160.131) |
| 16:08.45 | SoBlindWolf | Welcome! :D |
| 16:10.33 | SoBlindWolf | Okay guys I have to go I will be back after lunch US Central so I can ask questions :D Sorry If I am a pain in advance thank you for your time and have a nice day. |
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| 16:20.06 | cervajs2 | xarmiex: it will be in ast 13.10.0. check mailing list after 13.9.0 was published |
| 16:22.14 | cervajs2 | howto create segfault: load res_config_mysql.so & queue_log => odbc,db in extconfig.conf (tough experience) |
| 16:42.34 | xarmiex | cervajs2: any chance this fix will hit the 11.x tree ? (havent found it in the 13.x changelog yet, still looking) |
| 16:46.01 | cervajs2 | xarmiex: its in 11 too |
| 16:47.32 | jwpierce3 | can I use sip.conf for my trunk and pjsip.conf for my devices? |
| 16:48.05 | file | yes. |
| 16:48.18 | file | provided they are bound to different ports |
| 16:51.43 | xarmiex | cervajs2: do you have bug number im running 11.16.0 on that box, trying to confirm thats the issue im seeing |
| 16:53.00 | cervajs2 | xarmiex: https://github.com/asterisk/asterisk/commit/83dadc4683abcd10f0c4566abef541d997dcf5b8 |
| 16:53.17 | xarmiex | thx |
| 16:53.17 | cervajs2 | xarmiex: https://issues.asterisk.org/jira/browse/ASTERISK-16115 |
| 16:54.41 | xarmiex | i cant believe that bug has been sitting there for that long |
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| 16:57.31 | cervajs2 | app_queue is something like lord voldemort |
| 17:01.46 | xarmiex | yeah i vaguely remember it not being so great, i thought they were re-writing the entire thing a few years ago |
| 17:02.21 | file | nope |
| 17:02.47 | file | we did do ARI, which can be used to implement queueing using outside logic - which has done by quite a few people |
| 17:07.59 | cervajs2 | file: do you know if there is something public available? |
| 17:08.15 | file | I'm not aware of anything generic and actively maintained |
| 17:08.17 | cervajs2 | preferably in node.js |
| 17:08.38 | file | generally it is rather business/environment specific centric |
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| 17:46.58 | SoBlindWolf | hey guys |
| 17:48.05 | SoBlindWolf | I was wondering if Google Voice number would work for Asterisk and FreeBPX |
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| 19:16.27 | jrun | file: should one always work with ast_channel_ref(p->owner) ? i'm woking on ast_start_mixmonitor() |
| 19:16.44 | jrun | i guess my question is, is it recommended to work with the refs to channel? |
| 19:16.58 | jrun | or should working with channels be fine? |
| 19:17.07 | jrun | *working directly |
| 19:17.13 | file | something has to hold a ref, or else it could go away |
| 19:17.23 | jrun | owner_ref |
| 19:17.31 | jrun | oh i see |
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| 20:16.44 | tompaw | file: hi, I got a sample coredump |
| 20:16.54 | file | k |
| 20:16.56 | file | k |
| 20:18.27 | tompaw | can I give you a URL to my owncloud, or email it to you guys somehow? it's 5 megs gzipped |
| 20:18.54 | file | a coredump is useless to anyone else, you need to get a backtrace from it against the binary |
| 20:19.11 | file | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationAfterACrash |
| 20:19.18 | tompaw | on it. |
| 20:27.53 | tompaw | got it, do you want to have a quick look at a pastebin or should I submit it to your JIRA? |
| 20:29.15 | file | JIRA would be best, it would end up there regardless |
| 20:30.13 | tompaw | ok |
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| 20:36.41 | tompaw | your guildline says to ask at IRC first ;) |
| 20:37.33 | file | noone reads those |
| 20:46.38 | tompaw | https://issues.asterisk.org/jira/browse/ASTERISK-26079 |
| 20:50.00 | file | will be fixed in next release |
| 20:50.04 | file | https://issues.asterisk.org/jira/browse/ASTERISK-25941 |
| 20:50.39 | file | same thing |
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| 20:52.32 | tompaw | how did you manage to find this race condition in my backtrace? I was looking at this issue before, but ot didn't occur to me they could be related. |
| 20:52.45 | file | same backtrace for both |
| 20:56.12 | tompaw | I guess I'll have to read up on gdb then cause I still don't see it, but great news! thanks! |
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| 20:58.27 | file | Thread 1 is the problem |
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| 21:01.42 | tompaw | hm... my function calls and source file names are obscured, they all show up as ?? @ libpjsip.so |
| 21:01.59 | file | your PJSIP isn't built with symbols, but the general flow is the same |
| 21:02.08 | tompaw | I can see it now |
| 21:03.25 | tompaw | that is a huge relief, we've been pulling our hair off because of this. |
| 21:04.26 | tompaw | I'll be looking forward to next release then, cheers. |
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| 21:05.04 | file | the future is friendly |
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| 21:23.32 | drmessano | LinkedIn makes trolling difficult |
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| 21:57.35 | dan_j | Hi. Is there any way to have multiple email addresses for voicemail email notification? |
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| 22:04.50 | dan_j | Without using aliases. Use case, two people want to receive voicemail emails for one mailbox. One is using gmail, other is using yahoo, so no features for them to set up a distribution address. |
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| 23:22.26 | lvlinux | anybody know how to get systemd to auto-restart Asterisk after a crash? |
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