| 00:02.57 | *** join/#asterisk Spengler (~spengler1@pool-100-16-54-188.bltmmd.fios.verizon.net) |
| 00:03.14 | Spengler | Hello, anyone in here use Vitelity? |
| 00:04.52 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 00:04.57 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 00:12.13 | *** join/#asterisk UnregisteredSnow (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 00:18.39 | *** join/#asterisk MarqueI (~Marquel@social.pc-tiede.de) |
| 00:19.11 | *** join/#asterisk Marquel (~Marquel@fuchsfanclub/allerdings/marquel) |
| 00:19.49 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 00:22.48 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 00:23.16 | *** join/#asterisk chazzam (~chazz@donutokyo.info) |
| 00:53.19 | *** join/#asterisk uhmmm (~uhmmm@host23-236-static.10-188-b.business.telecomitalia.it) |
| 00:59.19 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:03.09 | amonk | Spengler: sort of |
| 01:12.28 | *** join/#asterisk jasonwert (~wert@71.89.137.28) |
| 03:07.12 | *** join/#asterisk overyander (~jeff@209.141.208.197) |
| 03:46.55 | *** join/#asterisk cfx_ (~deserver@unaffiliated/cfx-/x-0311648) |
| 03:58.26 | lvlinux | Spengler: I use 'em |
| 05:13.05 | *** join/#asterisk ericNL (eric@2a01:7e00:e000:69::c0ff:ee) |
| 05:13.22 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 05:21.05 | *** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net) |
| 05:32.42 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 05:33.05 | *** join/#asterisk ericNL (eric@2a01:7e00:e000:69::c0ff:ee) |
| 05:33.55 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 06:24.38 | *** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net) |
| 06:27.36 | *** join/#asterisk evil_gordita (robert@ip70-188-44-118.rn.hr.cox.net) |
| 06:40.11 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 06:40.50 | *** join/#asterisk evil_gordita (robert@ip70-188-44-118.rn.hr.cox.net) |
| 06:45.30 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 07:01.13 | Ellenor | Gn8 |
| 07:24.51 | *** join/#asterisk F2Knight_ (~F2Knight@208.100.173.58) |
| 07:33.44 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 07:38.10 | *** join/#asterisk ericNL (eric@2a01:7e00:e000:69::c0ff:ee) |
| 07:50.15 | *** join/#asterisk F2Knight (~F2Knight@50.139.86.39) |
| 07:52.47 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 07:58.13 | *** join/#asterisk ericNL (eric@2a01:7e00:e000:69::c0ff:ee) |
| 08:25.45 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 08:32.18 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:46.42 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 09:06.49 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 09:09.17 | *** join/#asterisk ericNL (eric@2a01:7e00:e000:69::c0ff:ee) |
| 10:00.48 | *** join/#asterisk Draecos (~Draecos@203-121-194-11.e-wire.net.au) |
| 10:02.10 | *** join/#asterisk Draecos (~Draecos@203-121-194-11.e-wire.net.au) |
| 10:17.47 | *** join/#asterisk ruied (~ruied@bl7-213-65.dsl.telepac.pt) |
| 10:22.29 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 11:08.36 | *** join/#asterisk Milos|Surface (~Milos@pdpc/supporter/student/milos) |
| 11:08.50 | Milos|Surface | How can I resolve this warning, which doesn't impact the call in any way that I've noticed? [Dec 28 00:08:09] WARNING[491][C-00000044]: translate.c:341 framein: no samples for g722tolin16 |
| 11:12.32 | Milos|Surface | apparently something to do with silence suppression |
| 11:12.34 | Milos|Surface | on my client |
| 11:12.35 | Milos|Surface | hmm ok |
| 11:19.50 | *** join/#asterisk virtuaposta (~suraj@103.51.139.126) |
| 11:20.10 | virtuaposta | hello guys |
| 11:21.01 | virtuaposta | I have installed asterisk on one of my live boxes, same setup is working for me when I try it within LAN but at live IP, call gets connected but no voice.. any guidance |
| 11:22.00 | WIMPy | From/to where? NAT involved? Which side(s)? |
| 11:22.24 | virtuaposta | no NAT involved |
| 11:22.55 | virtuaposta | asterisk, serving on live IP/box, users can connect and call, but no voice during call |
| 11:29.12 | robmal | By live IP you mean a dynamic one that changes from time to time? Does your ITSP support this? |
| 11:29.19 | virtuaposta | no sound even if I disable iptables |
| 11:29.37 | virtuaposta | no I have static global IP |
| 11:29.54 | robmal | So what is this live IP you are talking about? |
| 11:30.38 | WIMPy | What about the oter side? |
| 11:30.58 | virtuaposta | robmal, forget about "live ip" word, I have a box setup with asterisk, iptables disabled, I am using static IP, users can connect and call each other but no sound during calls |
| 11:31.32 | virtuaposta | what other side WIMPy? |
| 11:32.04 | virtuaposta | sorry I didn't collected your question, anything more in info you need? sorry my bad |
| 11:32.39 | WIMPy | The users. |
| 11:33.13 | virtuaposta | WIMPy, users can connect and call each other, only issue is, there is no sound during call |
| 11:33.16 | WIMPy | You need to look at the whole communication from one phone to the oter including all network components in between. |
| 11:34.30 | virtuaposta | caller and receiver not able able to hear each other, otherwise connectivity and authentication is fine |
| 11:34.59 | virtuaposta | caller and receiver not able to hear each other, otherwise connectivity and authentication is fine |
| 11:42.22 | virtuaposta | also I have not set my global IP anywhere is conf files, in case is required, not sure where to place it. Tried placing externaddr in sip.conf but no help |
| 11:42.47 | WIMPy | Only if you're behind nat. |
| 11:43.01 | WIMPy | So what about the users? No Firewall there? |
| 11:44.09 | virtuaposta | no firewall |
| 11:45.37 | WIMPy | All phones have public IPs? |
| 11:46.19 | *** join/#asterisk lambda79 (~fulcrap@unaffiliated/lambda79) |
| 11:49.11 | virtuaposta | I am using softphones, so yes during connectivity they do |
| 11:50.05 | virtuaposta | I can call and pick the call even, only issue is, cant hear anything |
| 11:51.54 | WIMPy | Then you're in to so fun reading te SIP pakets to find out who's sending something bad. |
| 12:06.00 | *** join/#asterisk chendy (~alexc@50.93.203.50) |
| 12:15.06 | *** join/#asterisk defsdoor (~andy@cpc73037-sutt4-2-0-cust62.19-1.cable.virginm.net) |
| 12:28.53 | *** join/#asterisk dadrc (~quassel@unaffiliated/dadrc) |
| 12:29.56 | *** join/#asterisk chendy (~alexc@183.39.136.115) |
| 12:41.21 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 12:59.25 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:34.33 | *** join/#asterisk u0m3 (~u0m3@188.27.74.65) |
| 13:49.28 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 13:49.28 | *** mode/#asterisk [+o file] by ChanServ |
| 14:06.56 | *** join/#asterisk emk (~emk@unaffiliated/emk) |
| 14:07.56 | *** join/#asterisk Takapa (vegard@junior.svanberg.no) |
| 14:26.11 | *** join/#asterisk tompaw (U2FsdGVkX1@tompaw.xxx) |
| 14:27.18 | *** join/#asterisk jasonwert (~quassel@71.89.137.28) |
| 14:42.13 | *** join/#asterisk steelbrain (~steelbrai@103.255.5.48) |
| 14:43.02 | steelbrain | so I switched my stack from chan_sip to pjsip and I'm gonna cry, there's a one-way audio weird issue that I have no idea why is happening |
| 14:43.51 | steelbrain | Is there any list of all valid pjsip options? I haven't found it so far, also icesupport was renamed to ice_support but stunaddr or stun_addr seem to be unrecognized, Any ideas why? |
| 14:48.30 | *** join/#asterisk emk (~emk@unaffiliated/emk) |
| 14:49.20 | *** join/#asterisk JonathanD (~JonathanD@freenode/staff-emeritus/jonathand) |
| 15:00.05 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 15:06.19 | steelbrain | Any ideas where there's a 502 at the end? |
| 15:06.19 | steelbrain | http://pastebin.com/raw/v3eEBj74 |
| 15:20.58 | [TK]D-Fender | There is no 502 in that PB at all |
| 15:49.00 | *** join/#asterisk boris_t (~boris_t@212.119.243.183) |
| 16:03.57 | strata | i wouldn't want any 502's in my PB either. just peanuts to make it crunchy |
| 16:08.47 | *** join/#asterisk steelbrain (~steelbrai@103.255.5.48) |
| 16:08.56 | steelbrain | [TK]D-Fender, Sorry I meant 501 |
| 16:09.00 | steelbrain | It's the last response |
| 16:09.24 | steelbrain | btw, I've got PJSIP to work for me **magically** |
| 16:09.41 | steelbrain | I added direct_media=yes and it worked, I'd assumed it was the default but I guess not |
| 16:10.40 | [TK]D-Fender | Where did you configure the mailbox for it? |
| 16:11.17 | steelbrain | I don't use a mailbox so I didn't fill anything regarding the mailbox |
| 16:11.29 | [TK]D-Fender | Well taht's all it's asking about for that |
| 16:11.40 | [TK]D-Fender | So if you didn't set it up, why should it succeed? |
| 16:11.49 | steelbrain | Oh, thanks for the info! |
| 16:12.36 | [TK]D-Fender | "Gimme my mailbox info!" "Mailbox? What's that?" |
| 16:12.43 | steelbrain | Makes sense! |
| 16:12.47 | steelbrain | Another question I have for you guys is that I have a GoIP-8, it has 8 gsm slots, is there any way I could connect it to asterisk as a trunk and call out via a specific sim card? |
| 16:13.01 | steelbrain | This is the extension configuration for it atm, |
| 16:13.07 | steelbrain | [to-goip] |
| 16:13.22 | steelbrain | exten => _X.,1,Dial(PJSIP/goip/${EXTEN}) |
| 16:14.21 | strata | is it better to use sip or pjsip configuration? |
| 16:14.42 | steelbrain | strata: I **think** chan_sip is going to be removed sooner or later in favor of pjsip |
| 16:15.01 | strata | ok so does pjsip work with realtime arch? |
| 16:15.19 | steelbrain | Let me send you the link to wiki about PJSIP RealTime |
| 16:15.25 | steelbrain | https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime |
| 16:15.26 | strata | i'm on chan_sip right now because i had some trouble initially with pjsip and TLS+SRTP |
| 16:15.48 | [TK]D-Fender | <steelbrain> Another question I have for you guys is that I have a GoIP-8, it has 8 gsm slots, is there any way I could connect it to asterisk as a trunk and call out via a specific sim card? <- depends how it is designed to work. |
| 16:16.00 | [TK]D-Fender | Do it LET you target a specific one? |
| 16:16.03 | [TK]D-Fender | does* |
| 16:16.13 | [TK]D-Fender | Read the manual for it |
| 16:16.39 | steelbrain | I don't think it's manual has any info about how it works :P It's china-made, but I'll dig the docs |
| 16:16.51 | [TK]D-Fender | Everything is made in China |
| 16:16.58 | [TK]D-Fender | That doesn't excuse poor documentation |
| 16:17.15 | steelbrain | Agreed |
| 16:33.04 | strata | it's just a language barrier. usually the documentation is correct - just not grammatically sensible. |
| 16:34.09 | strata | my favorite is when the wrong words are used and somehow it turns into a rule 34 situation |
| 16:35.06 | WIMPy | Blame google |
| 16:38.06 | *** join/#asterisk bof22 (~Thunderbi@ARennes-650-1-18-19.w90-32.abo.wanadoo.fr) |
| 17:04.09 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 18:04.00 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 18:06.17 | *** join/#asterisk jasonwert (~quassel@71.89.137.28) |
| 18:13.22 | steelbrain | This PJSIP is so powerful, and therefore confusing |
| 18:14.20 | steelbrain | Anther question I have to ask is that max_contacts=1 was being respected when I defined a user in pjsip.conf, but now that I've converted the base into a wizard and placed it in pjsip_wizard.conf and renamed config param to aor/max_contacts=1, it's not working anymore |
| 18:15.54 | steelbrain | here's my pjsip_wizard.conf http://hastebin.com/delocucudi.txt |
| 18:16.27 | steelbrain | and this is my pjsip.conf http://hastebin.com/vafajugoka.txt |
| 18:17.27 | steelbrain | It used to allow only one connection to one account, but now it allows as many as possible |
| 18:22.39 | *** join/#asterisk jjrh (~jjrh@2607:f0b0:1:6e12:110b:3f5d:e57a:c29a) |
| 18:27.00 | steelbrain | Found the solution, aor/remove_existing=false should be set to disallow new connections |
| 18:32.30 | [TK]D-Fender | wizard = garbage |
| 18:32.47 | [TK]D-Fender | it's a lazy attempt to not make the separate sections like you should |
| 18:33.39 | Ellenor | [TK]D-Fender: maybe pjsip is too powerful for them |
| 18:34.05 | [TK]D-Fender | No, simply lazy. |
| 18:34.11 | steelbrain | I love it so far |
| 18:34.26 | steelbrain | It has simplified my user generation script |
| 18:34.32 | steelbrain | I only have to write 3 lines to create a user now |
| 18:35.01 | [TK]D-Fender | Anyone who is capable of scripting things like that wouldn't need "wizard" for anything and as soon as the needs exapand it's time to reinvent the wheel |
| 18:35.08 | steelbrain | I guess everything has it's pros and cons, the con of PJSIP at the moment is not-so-good-documentation |
| 18:35.59 | steelbrain | [TK]D-Fender, Agreed but for the most basic operations, it reduces the lines of code in your config generator script, it's awesome while it works, I'll deal with it when it oesn't :P |
| 18:37.38 | [TK]D-Fender | tosses "wizard" in the same trash bin he used for AEL and users.conf |
| 18:39.05 | steelbrain | :D |
| 18:51.46 | file | can you elaborate on how the documentation is not so good, and how it could be better? |
| 18:52.09 | Ellenor | file: Needs absolute newbs to understand it. |
| 18:55.59 | file | or cake |
| 19:01.59 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 19:06.07 | *** join/#asterisk steelbrain (~steelbrai@103.255.7.12) |
| 19:31.05 | *** join/#asterisk steelbrain_ (~steelbrai@103.255.7.12) |
| 19:31.11 | steelbrain_ | Does asterisk support USSD? |
| 19:31.18 | steelbrain_ | I wanna capture and reply to *111# |
| 19:32.11 | steelbrain_ | Oops, logged in twice :D |
| 19:37.09 | *** join/#asterisk mac_ified (~mac_ified@67-9-150-210.res.bhn.net) |
| 19:37.53 | [TK]D-Fender | Directly no, perhaps there is a channel driver for some device that provides it itself |
| 19:59.00 | *** join/#asterisk ruied (~ruied@bl7-213-65.dsl.telepac.pt) |
| 20:08.43 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 23:14.20 | *** join/#asterisk war9407 (war@static-72-73-18-14.clppva.fios.verizon.net) |
| 23:43.33 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |