| 00:08.53 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 00:17.00 | *** join/#asterisk-dev infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net) |
| 00:17.00 | *** topic/#asterisk-dev is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Check out our blog! blogs.asterisk.org -=- Follow on Twitter at @AsteriskDev |
| 00:30.49 | *** join/#asterisk-dev spatel (~spatel@pool-98-118-124-216.bstnma.fios.verizon.net) |
| 01:08.55 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 03:59.44 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 04:20.48 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 05:04.45 | *** join/#asterisk-dev jkroon (~jkroon@165.16.203.105) |
| 05:10.46 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 05:31.31 | spatel | Hello! |
| 05:32.13 | tsearle | Hello |
| 05:32.58 | tsearle | Way to early to be cheerful (at least in Europe) |
| 05:35.00 | spatel | hey!! |
| 05:35.03 | spatel | morning |
| 05:35.09 | spatel | I am from US :) |
| 05:35.41 | spatel | tsearle: hey i am working on opensips + asterisk integration and have some question if you help me out? |
| 05:36.25 | spatel | I want opensips my registra and asterisk for media traffic. |
| 05:36.41 | tsearle | I use kamailio instead of opensips, but I guess i could try |
| 05:36.55 | spatel | is it possible opensips use asterisk for RTP media? |
| 05:38.06 | spatel | why kamailio but not opensips? they are pretty much same function vise.. |
| 05:38.35 | spatel | I have deployed using this doc - https://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration |
| 05:39.30 | spatel | But this doc doesn't saying SIP client traffic will flow via Asterisk :( so i am trying to understand, does asterisk can be in media path in this document? |
| 05:41.11 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 05:48.45 | tsearle | Well by default opensips wouldn't do rtp. And asterisk will by default |
| 05:49.08 | *** join/#asterisk-dev CELYA_ (~Thunderbi@2a01cb0583d31d00f4d7ddd1bc0987bc.ipv6.abo.wanadoo.fr) |
| 05:49.28 | tsearle | So just use opensips without any rtp proxy and things will be fine |
| 05:50.47 | spatel | You are saying flow will be like following |
| 05:52.20 | tsearle | If you have phone - sips - ast - sips - phone |
| 05:52.21 | spatel | sip_client_A------REGISTER---->[Opensips]-----INVITE----->[ASTERISK]---it will send call back to opensips----[Opensips]---opensip look into lookup table----> sip_client_B |
| 05:52.31 | tsearle | The rtp would flow phone - ast - phone |
| 05:53.03 | spatel | Reason i need opensips because it can handle large registration and want to put multiple asterisk behind opensips for media |
| 05:54.20 | tsearle | Yeah, well as I said, opensips (unless you enable rtpproxy) won't touch sdp so won't be in the media path |
| 05:55.02 | spatel | yes, i am aware of that. without rtpptroxy it can't handle media |
| 05:55.20 | spatel | I am trying to replace rtpproxy with asterisk :) |
| 05:55.45 | spatel | asterisk has much better media handling. |
| 05:56.05 | spatel | all i am trying to find how will be call flow and RTP flow |
| 05:57.35 | spatel | I need to spend little more time to poke around and try out something... |
| 05:58.20 | spatel | I think i need to get some sleep first :) |
| 05:58.48 | spatel | tsearle: thanks and have a good day.. i will catch you guys tomorrow evening :) |
| 05:58.50 | spatel | good nite |
| 06:13.08 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 06:17.25 | *** join/#asterisk-dev pchero_work (~pchero@87.213.247.82) |
| 06:18.05 | *** join/#asterisk-dev pchero_work (~pchero@87.213.247.82) |
| 06:35.08 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 06:40.22 | *** join/#asterisk-dev CELYA_ (~Thunderbi@2a01:e0a:d:1111:b749:768d:ed77:7c9a) |
| 07:21.15 | *** join/#asterisk-dev mvanbaak (~mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 07:21.15 | *** mode/#asterisk-dev [+o mvanbaak] by ChanServ |
| 07:27.14 | *** join/#asterisk-dev CELYA_ (~Thunderbi@2a01:e0a:d:1111:b028:63cb:4a88:ef99) |
| 07:28.09 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 09:39.32 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 10:00.18 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 10:22.13 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 11:36.01 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 11:51.45 | *** join/#asterisk-dev tzafrir (~tzafrir@lemon.tzafrir.org.il) |
| 11:51.45 | *** mode/#asterisk-dev [+o tzafrir] by ChanServ |
| 11:52.35 | *** join/#asterisk-dev spatel (~spatel@pool-98-118-124-216.bstnma.fios.verizon.net) |
| 12:36.25 | *** join/#asterisk-dev spatel (~spatel@pool-98-118-124-216.bstnma.fios.verizon.net) |
| 12:45.21 | *** join/#asterisk-dev hehol (~hehol@gatekeeper.loca.net) |
| 12:55.31 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-52-147-153.dialup.adsl.anteldata.net.uy) |
| 13:36.58 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 14:02.26 | *** join/#asterisk-dev bford (uid283514@gateway/web/irccloud.com/x-twqqzgwwaiwvdrjk) |
| 14:02.26 | *** mode/#asterisk-dev [+o bford] by ChanServ |
| 14:03.09 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-52-147-153.dialup.adsl.anteldata.net.uy) |
| 14:15.59 | *** join/#asterisk-dev spatel (~spatel@static-71-174-102-210.bstnma.fios.verizon.net) |
| 14:28.33 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 14:43.16 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-52-147-153.dialup.adsl.anteldata.net.uy) |
| 14:47.27 | *** join/#asterisk-dev kharwell (uid358942@gateway/web/irccloud.com/x-gicdelzmjxrksthi) |
| 14:47.27 | *** mode/#asterisk-dev [+o kharwell] by ChanServ |
| 14:49.13 | *** join/#asterisk-dev scgm11_ (~scgm11@87.44.202.35.bc.googleusercontent.com) |
| 14:55.29 | *** join/#asterisk-dev scgm11__ (~scgm11@147.204.224.35.bc.googleusercontent.com) |
| 14:59.49 | *** join/#asterisk-dev scgm11_ (~scgm11@87.44.202.35.bc.googleusercontent.com) |
| 15:00.34 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 15:04.51 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-52-147-153.dialup.adsl.anteldata.net.uy) |
| 15:32.23 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 16:06.27 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 16:09.06 | *** join/#asterisk-dev pchero_work (~pchero@dhcp-077-249-058-090.chello.nl) |
| 16:15.08 | seanbright | how would you bind rtp to ipv4 only? |
| 16:15.13 | seanbright | re: ice candidates |
| 16:15.48 | gtjoseph | in rtp.conf? |
| 16:15.59 | gtjoseph | [ice_host_candidates] |
| 16:16.04 | seanbright | hmm |
| 16:16.15 | seanbright | 20:52 <@kharwell> file: is that automatic or do you have to configure it to do so? |
| 16:16.18 | seanbright | 20:52 <@file> automatic |
| 16:16.21 | seanbright | 20:53 <@file> provided IPv6 is enabled on the system, and you aren't binding RTP to only IPv4 or IPv6 |
| 16:16.30 | file | bind_rtp_to_media_address methinks |
| 16:17.08 | gtjoseph | yeah what's the goal? you can also blacklist in rtp.conf as well., |
| 16:17.25 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-50-33-14.dialup.adsl.anteldata.net.uy) |
| 16:18.01 | seanbright | i have ipv6 enabled (because it's the default), but i don't want to send IPv6 candidates |
| 16:18.09 | seanbright | because they are link local addresses anyway |
| 16:18.35 | gtjoseph | ice_blacklist = fe80::/64 |
| 16:22.28 | seanbright | seems to work |
| 16:22.29 | seanbright | thanks |
| 16:26.07 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-50-33-14.dialup.adsl.anteldata.net.uy) |
| 16:38.46 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-50-33-14.dialup.adsl.anteldata.net.uy) |
| 16:45.56 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-50-33-14.dialup.adsl.anteldata.net.uy) |
| 16:47.42 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 16:50.30 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-50-33-14.dialup.adsl.anteldata.net.uy) |
| 16:57.50 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-50-33-14.dialup.adsl.anteldata.net.uy) |
| 17:02.06 | *** join/#asterisk-dev scgm11__ (~scgm11@87.44.202.35.bc.googleusercontent.com) |
| 17:13.51 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-50-33-14.dialup.adsl.anteldata.net.uy) |
| 17:17.46 | *** join/#asterisk-dev CELYA_ (~Thunderbi@2a01cb0583d31d00f4d7ddd1bc0987bc.ipv6.abo.wanadoo.fr) |
| 17:56.02 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-50-33-14.dialup.adsl.anteldata.net.uy) |
| 18:25.25 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-50-33-14.dialup.adsl.anteldata.net.uy) |
| 18:43.22 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-50-33-14.dialup.adsl.anteldata.net.uy) |
| 19:03.53 | seanbright | so... turns out that broken headsets cause audio problems |
| 19:08.26 | *** join/#asterisk-dev scgm11_ (~scgm11@87.44.202.35.bc.googleusercontent.com) |
| 19:09.59 | file | seanbright: wha |
| 19:19.39 | *** join/#asterisk-dev scgm11_ (~scgm11@87.44.202.35.bc.googleusercontent.com) |
| 19:23.46 | seanbright | I KNOW |
| 19:29.06 | *** join/#asterisk-dev scgm11_ (~scgm11@87.44.202.35.bc.googleusercontent.com) |
| 19:30.20 | *** join/#asterisk-dev scgm11_ (~scgm11@r186-50-33-14.dialup.adsl.anteldata.net.uy) |
| 20:00.15 | *** join/#asterisk-dev CELYA_ (~Thunderbi@lfbn-1-5192-96.w90-105.abo.wanadoo.fr) |
| 20:47.03 | seanbright | iiiiiiiiiiinteresting: https://tools.ietf.org/html/rfc6464 |
| 20:47.15 | file | yes |
| 20:50.29 | seanbright | we don't do rtp extension negotiation, correct? |
| 20:50.40 | seanbright | or we do it in a hard-coded kinda way |
| 20:50.57 | file | we do negotiation for it |
| 20:51.03 | seanbright | huh |
| 20:51.11 | seanbright | my mistake then |
| 20:51.14 | file | you have to, as values aren't fixed |
| 20:51.52 | *** join/#asterisk-dev pchero_work (~pchero@dhcp-077-249-058-090.chello.nl) |
| 20:52.03 | file | you're thinking of remb |
| 20:52.37 | file | that's hardcoded |
| 20:55.03 | seanbright | ahh |
| 20:55.34 | seanbright | so in theory, we could negotiation this ssrc-audio-level stuff, but we have no way of passing that information to, say, app_confbridge |
| 20:55.40 | seanbright | could negotiate* |
| 20:55.41 | file | correct |
| 20:56.00 | seanbright | word |
| 20:56.05 | seanbright | get on that, woulda? |
| 20:56.20 | seanbright | hopefully told ya right. |
| 20:56.43 | file | please allow 5-10 years for delivery |
| 21:08.47 | kharwell | seanbright: haha glad I gave you a new phrase to use |
| 21:37.35 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 22:18.48 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 22:40.55 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |
| 23:42.10 | *** join/#asterisk-dev afournier (~admin@189.153.118.12) |