| 00:02.57 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 00:19.31 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:19.31 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 00:20.34 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 08:55.48 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 08:55.48 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 09:08.31 | *** join/#asterisk TandyUK (~admin@2a02:13a0:a006:1:9df3:d721:b156:e7e1) |
| 09:10.58 | *** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.213.148) |
| 09:29.36 | td34 | driving me insane!! :P |
| 09:47.44 | *** join/#asterisk samwierema (~samwierem@095-097-255-066.static.chello.nl) |
| 09:48.29 | samwierema | What mailing list is best to track security issues in Asterisk? asterisk-dev? |
| 10:00.34 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 10:11.06 | file | you mean to be notified of them? |
| 10:13.19 | file | we announce on -users, -dev, -announce, Twitter, and the topic here is updated with the security releases as the latest |
| 10:48.39 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 10:53.35 | *** join/#asterisk detha (~detha@unaffiliated/detha) |
| 10:59.04 | samwierema | file: thanks, I missed the "See all lists" link on the website and couldn't find the -announce list |
| 11:12.58 | *** join/#asterisk dnit (673399be@gateway/web/freenode/ip.103.51.153.190) |
| 11:13.15 | dnit | Hi |
| 11:13.33 | dnit | How can I analyze early media in a call |
| 12:11.18 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 12:25.23 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 12:26.26 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 12:30.31 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:38.46 | *** join/#asterisk juvenal (~juvenal@189.38.156.136) |
| 12:47.48 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 12:56.14 | Samot | Analyze? |
| 12:56.16 | Samot | What? |
| 13:14.30 | *** join/#asterisk newtonr (newtonr@nat/digium/x-bwyqauedyhoanwbx) |
| 13:14.31 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:31.26 | *** join/#asterisk sleip (~root@c-76-105-87-143.hsd1.ga.comcast.net) |
| 13:36.07 | *** join/#asterisk juvenal (~juvenal@189.38.156.136) |
| 13:43.37 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:43.37 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:03.52 | *** join/#asterisk bford (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 14:03.53 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:04.51 | *** join/#asterisk kharwell (kharwell@nat/digium/x-vbkvcjfxhshppqyf) |
| 14:04.51 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:05.13 | igcewieling | dnit: wireshark should work. |
| 14:15.31 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-anvohgenqzkjjetz) |
| 14:17.14 | dnit | igcewieling: I mean in asterisk dialplan. Like AMD application does some processing after call starts. I found this app https://github.com/phatjmo/app_cpa but cant figure out how to use it on early media |
| 14:18.46 | dnit | samot: sorry I mean process not analyze |
| 14:18.46 | igcewieling | bad questions lead to bad answers. |
| 14:19.57 | igcewieling | I'm not aware of anything in Asterisk to do that. Good luck. |
| 14:20.35 | dnit | :( |
| 14:20.43 | igcewieling | My former employer charged thousands of dollars per month to customer needing custom AMD. |
| 14:21.35 | igcewieling | What are you trying to accomplish with this crazy idea? |
| 14:22.55 | dnit | We are observing a lot of calls which are transmitting " this number is out of service" in early media instead of proper sip signalling. |
| 14:23.25 | dnit | Our system currently mark these calls as No Answer calls. This is leading to unhappy customers. |
| 14:23.43 | dnit | Our system ( asterisk) |
| 14:23.51 | igcewieling | whjy? The calls were not "answered". |
| 14:24.58 | dnit | But these are out of service numbers right ? |
| 14:25.22 | dnit | So it would be good if we mark them as OOS and not dial them again. |
| 14:25.36 | igcewieling | "out of service" can mean "number does not exist", in the USA it usually does. |
| 14:25.49 | igcewieling | when you call those numbers from other carriers, do they work? |
| 14:26.17 | dnit | No. Same message in early media on all carriers |
| 14:26.27 | dnit | Without SIP signals |
| 14:26.32 | igcewieling | Then the number doesn't exist. |
| 14:26.53 | dnit | Yes I and you know it but how does the system learns that ? |
| 14:26.53 | igcewieling | you are using SIP for connection to PSTN, correct? |
| 14:27.09 | dnit | For asterisk its just a number which has not been answered. |
| 14:27.37 | dnit | We are using SIP trunking to make all the calls via different carriers |
| 14:27.40 | igcewieling | dnit: If your carrier isn't giving OOB signaling there is nothing you can do. |
| 14:29.05 | dnit | Thats when I thought of either making the agents hear the Early media or use https://github.com/phatjmo/app_cpa to process early media |
| 14:30.21 | igcewieling | your idea is more of a realtime speech recognition. |
| 14:31.29 | igcewieling | When we call numbers not in service, our carrier sends back a SIP 404 to us. |
| 14:34.57 | nibbier | I have one problem left with my asterisk installation. it connects via some sip trunk and has internal sip phones. when I connected a specific external incoming call to some other external number asterisk build a simple-bridge, but no sound was transferred. I fixed this by first Playback(somesound) to the caller and then dialing out. This works now. I have the same issue when one of the internal sip phones sens a redirect to an external number when it's |
| 14:34.57 | nibbier | called. Asterisk builds a bridge but no sound. I could unconditionally playback a sound before Dialing the sip phone, but I'd rather only do this in the external-redirect case. any hints? |
| 14:35.11 | dnit | In my case they do give 404 for few but for most of such numbers it just the early media unless the call timesout |
| 14:40.50 | [TK]D-Fender | NIB |
| 14:41.01 | [TK]D-Fender | nibbier, You're allwing re-invites with your peers. Stop allowing them |
| 14:42.55 | nibbier | [TK]D-Fender: the redirect itself is the reinvite? |
| 14:43.55 | [TK]D-Fender | No. What that call goes through allows it |
| 14:44.01 | [TK]D-Fender | fix your peers <- |
| 14:44.32 | nibbier | [TK]D-Fender: I'll read up on those reinvites, disable them for my sip phones and try to understand. thanks for the pointer |
| 14:44.48 | [TK]D-Fender | directmedia=no <-------------- |
| 14:44.52 | [TK]D-Fender | nothing to read up. |
| 14:44.59 | [TK]D-Fender | Put it in all of your peers |
| 14:50.43 | dnit | Can I access early media using prebridge handler ? |
| 14:52.50 | nibbier | [TK]D-Fender: have directmedia=no in the forwarding sip phones extension. still does not work. |
| 14:53.10 | samwierema | Is it possible in PJSIP's endpoint configuration to set two (or more) outbound proxies and load balance between them? |
| 14:54.17 | [TK]D-Fender | it isn't the forwarder |
| 14:54.17 | [TK]D-Fender | the FORWARDER is the the channel that continues |
| 14:54.17 | [TK]D-Fender | [TK]D-Fender> Put it in all of your peers <----I was very clear here |
| 14:54.44 | [TK]D-Fender | dnit, no |
| 14:55.07 | [TK]D-Fender | dnit, Your idea is DOA. * was not built to accomodate this |
| 14:55.19 | [TK]D-Fender | samwierema, No. Load balancing is up to you |
| 14:55.47 | samwierema | [TK]D-Fender: hm, ok. Thanks! |
| 14:57.30 | nibbier | [TK]D-Fender: I put this in the genear section of my sip.conf... replaced any directmedia=yes by directmedia=no.... no success |
| 14:58.15 | [TK]D-Fender | ALL OF YOUR PEERS |
| 15:03.55 | nibbier | sadly I don't even know what a peer is, if you talk about all entities that are defined as type=peer or w/e |
| 15:05.06 | [TK]D-Fender | every SECTION |
| 15:05.46 | [TK]D-Fender | regardless of "type=" |
| 15:05.53 | [TK]D-Fender | AND [general] |
| 15:09.57 | dnit | [TK]D-Fender: Is there any alternative or I just can't acess ealry media in apps like AMD. |
| 15:10.23 | [TK]D-Fender | What happens when you send the call there? |
| 15:10.33 | [TK]D-Fender | What happens when you try to monitor it? |
| 15:10.39 | [TK]D-Fender | You should already have this answer... |
| 15:10.40 | igcewieling | dnit: if you want to write your own asterisk application you may do so. |
| 15:11.47 | *** join/#asterisk mub (~jub@unaffiliated/mub) |
| 15:19.37 | dnit | pre bridge handler is invoked after the call is picked up but before it is bridged ? |
| 15:19.46 | dnit | ah! |
| 15:22.03 | dnit | igcewieling: Looks like ill have to do so. Any hint which part of source code I should study first to accomplish this ? |
| 15:23.42 | igcewieling | dnit: first you'll need to learn things like C, audio processing methods, speech recognition, etc. |
| 15:24.13 | igcewieling | look at the AMD app for a sample |
| 15:24.34 | igcewieling | Personally, I think anyone who tries to do that is an idiot. |
| 15:25.20 | dnit | I wont create my own amd. I just want to use current amd on early media. |
| 15:25.34 | igcewieling | It is like saying that because my cellphone can't do X, then I'll start a cellphone carrier and have feature X. Yes, that might work. No, it isn't a good idea. |
| 15:25.42 | dnit | So simply a wrapper on AMD with access to early media will do my work |
| 15:25.58 | [TK]D-Fender | Again what happens when you TRY the handler NOW? |
| 15:26.27 | dnit | [TK]D-Fender: It is invoked after the call is picked uip |
| 15:27.34 | dnit | \help |
| 15:27.58 | [TK]D-Fender | #asterisk-dev <----- |
| 15:28.57 | igcewieling | goes back to troubleshooting dialplan variable scope. |
| 15:30.11 | [TK]D-Fender | igcewieling, Another 2 degrees up and don't forget to account for windage. |
| 15:33.05 | igcewieling | [TK]D-Fender: and take off the lens cap. 8-) |
| 15:33.53 | igcewieling | [TK]D-Fender: some sort of race condition between 2 legs of a .call |
| 15:34.07 | [TK]D-Fender | WHY THE RUSH?! |
| 15:34.28 | igcewieling | [TK]D-Fender: users! they always want it yesterday. |
| 15:36.24 | igcewieling | I few months ago I re-wrote part of my dialplan to eliminated the need for chan_local, but since nobody actually *uses* the feature, nobody realized for a few months. |
| 15:36.44 | igcewieling | nobody actually uses the .call files feature I'm troubleshooting that is. |
| 15:46.28 | *** join/#asterisk jkroon (~jkroon@vc-nat-gp-s-41-13-8-228.umts.vodacom.co.za) |
| 15:48.04 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 15:59.32 | *** join/#asterisk dakudos (~dakudos@c-73-243-248-133.hsd1.co.comcast.net) |
| 16:22.52 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
| 16:23.46 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 16:24.26 | *** join/#asterisk woose (~root@unaffiliated/woose) |
| 16:31.51 | *** join/#asterisk juvenal (~juvenal@189.38.156.136) |
| 16:40.31 | *** join/#asterisk miralin (~Thunderbi@91.237.94.1) |
| 16:47.00 | *** join/#asterisk juvenal (~juvenal@189.38.156.136) |
| 17:02.03 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 17:02.29 | *** part/#asterisk panatrator69 (~panatrato@cpe-97-99-68-237.tx.res.rr.com) |
| 17:02.46 | *** join/#asterisk panatrator69 (~panatrato@cpe-97-99-68-237.tx.res.rr.com) |
| 17:05.52 | wabbits | Who do you recommend vm hosting? |
| 17:11.09 | drmessano | Linode |
| 17:13.13 | wabbits | thanks drmessano |
| 17:16.54 | lorsungcu | wabbits: vultr.com |
| 17:19.03 | wabbits | thanks lorsungcu |
| 17:50.11 | drmessano | wrong |
| 17:50.21 | drmessano | He asked for recommendations |
| 17:59.10 | *** join/#asterisk juvenal (~juvenal@189.38.156.136) |
| 18:27.08 | *** join/#asterisk juvenal (~juvenal@152.250.188.25) |
| 18:50.16 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
| 19:09.12 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 19:10.23 | *** join/#asterisk samwierema (~samwierem@82.169.225.211) |
| 19:12.46 | *** join/#asterisk sakhi (~sakhilouw@vc-nat-gp-s-41-13-2-192.umts.vodacom.co.za) |
| 19:18.05 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-170.cust.bezeqint.net) |
| 19:19.11 | *** join/#asterisk jkroon (~jkroon@165.16.204.170) |
| 19:19.51 | *** join/#asterisk w9sh (~dad@c-73-43-86-234.hsd1.ga.comcast.net) |
| 19:21.30 | *** join/#asterisk Jack17 (~jackkk@136-144-133-251.colo.transip.net) |
| 19:23.16 | Jack17 | Hello guys i have asterisk 11.23.1 im trying to add a trunk that has two ips a signaling ip and media ip any idea guys ? |
| 19:24.07 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::1:3) |
| 19:25.02 | file | Asterisk isn't configured in regards to the remote media IP, that is part of the negotiation process when doing a call |
| 19:36.41 | *** join/#asterisk juvenal (~juvenal@152.250.188.25) |
| 19:40.53 | *** join/#asterisk jkroon (~jkroon@165.16.204.170) |
| 19:51.50 | igcewieling | Jack17: ignore the media ip when setting up the peer |
| 20:06.04 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 20:16.29 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 20:57.45 | Samot | He didn't have any of his NAT/external IP stuff setup |
| 20:57.56 | Samot | He asked in #freepbx too since he has FreePBX |
| 21:20.34 | *** join/#asterisk iheartlinux (~jwpierce3@mail.trunkmasters.com) |
| 21:28.47 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 21:28.47 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 21:40.57 | *** join/#asterisk newtonr (~newtonr@99-104-129-136.lightspeed.brhmal.sbcglobal.net) |
| 21:40.57 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 21:46.24 | *** join/#asterisk FuriousGeorge (4970460a@gateway/web/freenode/ip.73.112.70.10) |
| 21:46.24 | FuriousGeorge | hey all |
| 21:58.14 | FuriousGeorge | could someone kindly take a look at this sip debug output for me. im trying to figure out why im not getting video , but i dont see anything obvious |
| 21:58.18 | FuriousGeorge | https://pastebin.ca/3842624 |
| 21:58.42 | FuriousGeorge | unfortunately, i can only debug it on site, and so ium stuck here |
| 22:54.18 | *** join/#asterisk juvenal (~juvenal@152.250.188.25) |
| 23:09.39 | *** join/#asterisk juvenal (~juvenal@152.250.188.25) |
| 23:19.39 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 23:20.29 | *** join/#asterisk iheartlinux (~jwpierce3@mail.trunkmasters.com) |
| 23:20.54 | *** join/#asterisk Bhakimi (~textual@rrcs-69-75-121-202.west.biz.rr.com) |
| 23:21.23 | Bhakimi | hi guys, how can i adjust a volume of a channel using asterisk AMI ? |
| 23:24.12 | *** join/#asterisk juvenal (~juvenal@152.250.188.25) |
| 23:24.35 | *** part/#asterisk kharwell (kharwell@nat/digium/x-vbkvcjfxhshppqyf) |