| 00:00.38 | *** join/#asterisk phunguy (santas@dhcp.i-p.org.uk) |
| 00:09.22 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 00:42.25 | ketas | damn, i seem to have misconfiguration issue |
| 00:44.57 | ketas | i have multiple sip accounts my isp gives out for free so i have all them configured in asterisk |
| 00:45.14 | ketas | calls seem to come in via "wrong channel" |
| 00:46.37 | WIMPy | Did you register them with an extension? |
| 00:55.40 | *** join/#asterisk TysonBrooks (~TysonBroo@c-76-23-24-235.hsd1.ut.comcast.net) |
| 00:55.48 | TysonBrooks | Why would I be getting this? Ignoring audio media offer because port number is zero |
| 00:56.33 | [TK]D-Fender | ketas: probably because you created them all as PEERS and peers match by IP, and they are all coming from the same place so the first one gets all the calls in terms of association |
| 00:57.37 | TysonBrooks | You mean the types right? |
| 00:57.56 | TysonBrooks | if so, they are all friend. |
| 00:58.50 | ketas | [TK]D-Fender: well i use "register => user:pass@host/number" to filter calls later by that number... that part is fine |
| 00:59.33 | ketas | [TK]D-Fender: however the cdr lists that calls come in via wrong path? |
| 00:59.41 | [TK]D-Fender | ketas: I did not talk about the register statement at all. |
| 00:59.44 | ketas | what's correct way to do this i'm wondering |
| 01:00.11 | [TK]D-Fender | ketas: it's the fact you set up the inbound sections as TYPE=PEER |
| 01:00.27 | TysonBrooks | Also something to note, I'm only able to recieve calls 50% of the time. |
| 01:00.30 | ketas | what should i do then? |
| 01:00.30 | [TK]D-Fender | if you made multiple, the FIRST one will get all the calls beacuse the IP is the SAME |
| 01:01.06 | [TK]D-Fender | ketas: You might have no choice on this, especially if they don't auth their calls which very few do |
| 01:01.07 | ketas | indeed it is |
| 01:01.23 | [TK]D-Fender | ketas: Try setting up a "type=user" entry for their inbound and see if it'll hit. |
| 01:03.31 | ketas | i don't get the "don't auth" part |
| 01:04.39 | *** join/#asterisk suporte85 (~guardadig@201-43-142-229.dsl.telesp.net.br) |
| 01:04.40 | [TK]D-Fender | ketas: If they just throw the call at your server without credentials attached |
| 01:05.33 | ketas | i guess so |
| 01:05.52 | ketas | well still confusing |
| 01:06.24 | *** join/#asterisk Lachesis (~ems@unaffiliated/lachesis) |
| 01:07.00 | Lachesis | what's a good web interface I can install alongside asterisk ala FreePBX? my box is used for other things and can't run VMs. |
| 01:08.21 | ketas | [TK]D-Fender: is that required to get calls in via right "user"? |
| 01:09.00 | [TK]D-Fender | ketas: Assuming they send their calls in that mamer |
| 01:09.04 | [TK]D-Fender | manner* |
| 01:09.05 | ketas | [TK]D-Fender: i'm sure they don't think anyone would want to configure multiple account to asterisk |
| 01:09.12 | ketas | accounts |
| 01:10.41 | [TK]D-Fender | Lachesis: FreePBX |
| 01:10.43 | ketas | however original use case for this is consumer gateway with two ata's... this has single ip too |
| 01:11.28 | Lachesis | [TK]D-Fender, thanks, for some reason i thought freepbx was just a distro, but looking into it more i see i was wrong |
| 01:11.59 | [TK]D-Fender | Lachesis: The fact they rolled a distro is comparatively new. |
| 01:15.23 | *** join/#asterisk vlad_starkov (~vlad_star@213.152.135.29) |
| 01:17.23 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:22.00 | ketas | [TK]D-Fender: it's still damn confusing |
| 01:29.10 | ketas | how does type=user make sense here |
| 01:35.35 | ketas | ok, trying someting |
| 01:35.38 | ketas | thing |
| 01:44.23 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 01:44.24 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:47.16 | ketas | [TK]D-Fender: is that different "To:" i need? |
| 01:49.27 | ketas | or was that part i need somewhere else |
| 01:51.11 | ketas | it feels like "telephony" stuff is too complex compared with all the else that runs over ip |
| 01:53.02 | WIMPy | Depends on the protocoll used. |
| 01:53.30 | ketas | indeed i match for wrong things here |
| 01:54.11 | [TK]D-Fender | ketas: No |
| 01:54.26 | [TK]D-Fender | ketas: You need an entire ned ENTRY in sip.conf as TYPE=USER |
| 01:54.27 | ketas | was it in the invite sip:... then? |
| 01:54.29 | ketas | hmm |
| 01:54.36 | [TK]D-Fender | that will auth by the USERNAME |
| 01:54.36 | *** part/#asterisk cusco (~tralala@2001:41d0:1:6caf::3) |
| 01:58.52 | ketas | hell, that didn't match at all |
| 02:02.42 | *** join/#asterisk TysonBrooks (~TysonBroo@c-76-23-24-235.hsd1.ut.comcast.net) |
| 02:04.28 | ketas | with type=user |
| 02:04.38 | ketas | but where should the user name go? |
| 02:05.20 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 02:09.31 | [TK]D-Fender | [] |
| 02:09.33 | [TK]D-Fender | username= |
| 02:09.40 | [TK]D-Fender | the usual |
| 02:12.40 | ketas | i think i got it fixed |
| 02:13.02 | ketas | however i don't think matching by user works |
| 02:14.53 | *** join/#asterisk vlad_starkov (~vlad_star@213.152.135.29) |
| 02:19.07 | ketas | so i just created properly named matching section for host |
| 02:20.52 | *** join/#asterisk Thazza (~AndChat50@14-200-214-174.static.tpgi.com.au) |
| 02:21.29 | *** join/#asterisk Free99 (~Free99@cpe-66-108-105-10.nyc.res.rr.com) |
| 02:22.05 | Free99 | hey guys, i just tried moving from a working mysql setup to an odbc setup. for some reason I can't read my sippeers from the odbc connection |
| 02:22.30 | Free99 | I checked odbc is configured correctly by using isql |
| 02:22.57 | Free99 | the error I keep getting is that "config.c:2256 find_engine: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available" |
| 02:23.33 | Free99 | I tried uncommenting the preload lines on modules.conf for odbc too |
| 02:36.11 | *** join/#asterisk tadspoles (~tadspoles@nc-184-3-103-49.dhcp.embarqhsd.net) |
| 02:39.54 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |
| 02:54.48 | TysonBrooks | Any one know of a full featured open source soft phone? |
| 02:55.41 | *** join/#asterisk youjelly (~youjelly@39.47.49.81) |
| 02:56.22 | ketas | TysonBrooks: only half featured i'm afraid... |
| 02:56.43 | TysonBrooks | whats missing from the one(s) you're thinking of? |
| 02:56.55 | Lachesis | what's wrong with my (freepbx generated) sip.conf section? |
| 02:56.55 | ketas | i use linphone |
| 02:56.56 | Lachesis | http://pastebin.com/raw.php?i=LPt6b1RM |
| 02:57.42 | tm1000 | Lachesis: I dont know…whats wrong? why dont you tell us |
| 02:57.46 | Lachesis | whenever i try to register with the username 501 and the password testpass123, i get the message: [May 18 02:53:51] NOTICE[30397] chan_sip.c: Registration from '<sip:501@my.domain.name>' failed for '192.168.60.25:5060' - Wrong password |
| 02:59.15 | Lachesis | i've tried it both with and without that username=501 line |
| 03:00.29 | tm1000 | Lachesis: did you reload |
| 03:00.44 | Lachesis | i restarted asterisk |
| 03:01.30 | tm1000 | sorry to tell you but that is correct |
| 03:01.35 | tm1000 | and so is your file |
| 03:01.44 | Lachesis | hmm |
| 03:01.55 | Lachesis | what else could be going wrong? |
| 03:02.01 | tm1000 | *shrug* |
| 03:02.32 | Lachesis | tm1000, the softphone is known good |
| 03:02.41 | Lachesis | could it have anything to do with the my.domain.name part? |
| 03:07.45 | Lachesis | it still blows up even if i register with 501@192.168.60.2 |
| 03:09.04 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |
| 03:09.48 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.143) |
| 03:11.35 | tm1000 | Lachesis: try a different password |
| 03:11.41 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |
| 03:13.47 | TysonBrooks | What might I be doing wrong here? http://prntscr.com/15es2k only 50% of my calls are working. |
| 03:16.18 | *** join/#asterisk ChadAragorn (~ChadArago@206.251.40.221) |
| 03:16.34 | Lachesis | tm1000, looks like something's up with sip.conf and freepbx |
| 03:16.38 | Lachesis | trying to reinstall |
| 03:23.24 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 03:34.39 | *** join/#asterisk vlad_sta_ (~vlad_star@213.152.135.29) |
| 03:46.32 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 03:53.30 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 04:09.39 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.129) |
| 04:26.49 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:32.24 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 04:40.39 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.66) |
| 04:40.52 | ketas | sip has lot of terms that look so weird and confusing at first |
| 04:43.01 | *** join/#asterisk vlad_starkov (~vlad_star@213.152.135.29) |
| 04:48.22 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.130) |
| 04:58.41 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 05:22.30 | *** part/#asterisk mihamina (~mihamina@41.188.46.121) |
| 05:48.27 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
| 05:48.37 | *** join/#asterisk robert_ (~hellspawn@objectx/robert) |
| 06:09.40 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 06:50.18 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 06:54.30 | *** join/#asterisk TheCompWiz (~TheCompWi@216.7.159.250) |
| 06:54.33 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 06:54.59 | TheCompWiz | anyone know if it's possible to raise a blue-alarm on a dahdi pri span? |
| 07:00.02 | *** join/#asterisk evilman_home (kvirc@89.179.89.172) |
| 07:10.27 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.137) |
| 07:11.03 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.131) |
| 07:18.17 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 07:39.34 | *** join/#asterisk angryuser_laptop (~angryuser@ip-189.net-81-220-67.lyon.rev.numericable.fr) |
| 07:51.19 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.149) |
| 08:24.04 | *** join/#asterisk polysics (~Adium@host36-73-dynamic.16-79-r.retail.telecomitalia.it) |
| 08:24.09 | polysics | hi there |
| 08:24.16 | polysics | feeling slightly dumb here |
| 08:24.24 | polysics | ConfBridge again |
| 08:24.39 | polysics | how come the users don't hear MoH when they are alone in the conference? |
| 08:24.57 | polysics | I hear "You are currently the only person" then a join sound and audio starts |
| 08:27.10 | polysics | tbh it does look like ALL of my confbridge.conf file is ignored |
| 08:28.49 | polysics | I currently got this, yet I hear the announce_only_user_message: https://gist.github.com/polysics/e35deca0a295c9947cf2 |
| 08:39.15 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 09:08.50 | *** join/#asterisk areski (u931732@37.233.97.2) |
| 09:10.58 | polysics | WARNING[2214]: res_musiconhold.c:701 monmp3thread: poll() failed: Interrupted system call |
| 09:11.08 | polysics | any common cause for this, by chance? |
| 09:11.25 | polysics | trying to play one mp3 file using "quietmp3" ptype |
| 09:12.13 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 09:12.47 | polysics | [silent] |
| 09:12.47 | polysics | mode=quietmp3 |
| 09:12.47 | polysics | directory=/var/lib/asterisk/moh/silentmp3 |
| 09:13.34 | polysics | goal is to get silent moh to suppress ambient noise, I tried wav files but 8kHz silence is actually pretty noisy |
| 09:17.57 | TheCompWiz | what is playing the mp3? |
| 09:18.39 | TheCompWiz | streamplayer? mpg321? ??? |
| 09:19.06 | TheCompWiz | (and did you install it?) |
| 09:23.10 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 09:40.45 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 09:42.06 | *** join/#asterisk user_ (~user@82.137.193.252) |
| 09:47.11 | user_ | hi I have installed Digium card with CENTos +current asterisk + libss7 in passive TAP design I need to collect and record ss7 and wav files LEDs is Yalow I used 2 span first for RX and second span for TX and both on 1&2 wire |
| 09:47.24 | user_ | any one have idea how to configure it ? |
| 09:49.28 | user_ | I have got this MSG all time |
| 09:49.34 | user_ | [May 14 15:09:21] NOTICE[7414] chan_dahdi.c: SS7 got event: HDLC Abort(6) on span 1/0 |
| 09:49.35 | user_ | [May 14 15:09:21] NOTICE[7414] chan_dahdi.c: SS7 got event: HDLC Bad FCS(8) on span 1/0 |
| 10:12.15 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |
| 10:19.05 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 10:32.27 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |
| 10:32.37 | *** join/#asterisk mquin (~mquin@freenode/staff/mquin) |
| 10:36.56 | *** join/#asterisk mintos (~mvaliyav@114.143.43.46) |
| 10:38.24 | WIMPy | user_: Did you remove termination? |
| 10:38.58 | *** join/#asterisk TheCompWiz (~TheCompWi@216.7.159.250) |
| 10:39.34 | user_ | do you main tapped wire ? |
| 10:39.59 | WIMPy | yes |
| 10:40.11 | user_ | now not connected |
| 10:40.21 | user_ | but I leave it as it |
| 10:40.54 | WIMPy | I would expect issues if you have three terminators on the line. |
| 10:41.06 | user_ | exactly |
| 10:42.09 | user_ | but two switch running ss7 and the third one tapped one is connected to my server |
| 10:43.10 | user_ | the ss7 still run correctly between two switches |
| 10:46.30 | user_ | I need to configure the third termination as listener only |
| 10:47.48 | WIMPy | Just to be clear: YOu did understand that I was talking about resistors? |
| 10:49.24 | user_ | I have modified resistors on the third termination before my Digium ): |
| 10:50.05 | user_ | to got Yalow LED |
| 10:51.07 | user_ | it is stander splitting for balancing E1 |
| 10:51.17 | WIMPy | Wait. 1&2? Shouldn't that be 4&5? |
| 10:52.14 | user_ | 1&2 is RX ? |
| 10:53.42 | WIMPy | There's always a chance of being wrong but my doc has ith the other way round. |
| 10:55.59 | WIMPy | Yes, I guess that was from the network side. Less sensible. |
| 10:59.36 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.66) |
| 11:01.40 | user_ | I did check 1&2 is RX on Digium |
| 11:02.10 | *** join/#asterisk TysonBrooks (~TysonBroo@c-76-23-24-235.hsd1.ut.comcast.net) |
| 11:03.21 | user_ | so 1&2 on e1 connected to 1&2 on my first span and 4&5 from e1 is connected to 1&2 to my second span |
| 11:04.01 | user_ | so I receive the TX and RX from orginal trunk |
| 11:05.04 | user_ | have you an idea how to continue ? |
| 11:08.22 | WIMPy | Have you looked at the patches that exist for tapping and any information that comes with it? |
| 11:09.32 | WIMPy | The one that allows writing pcap files that is. |
| 11:10.06 | user_ | there is two |
| 11:10.19 | user_ | http://svn.digium.com/svn/asterisk/team/moy/dahdi-tap-trunk |
| 11:10.19 | user_ | http://svn.digium.com/svn/asterisk/team/moy/dahdi-tap-1.6.2 |
| 11:10.44 | user_ | which one please ? |
| 11:11.18 | WIMPy | No idea which one to use. |
| 11:11.32 | *** join/#asterisk pbxbrian (~pbxbrian@unaffiliated/brian98) |
| 11:12.30 | user_ | is it by MOY ? that allow me to get PCAP ? |
| 11:16.14 | WIMPy | Have you tried to look on Jira? |
| 11:16.22 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 11:17.15 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 11:25.51 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 11:28.57 | *** join/#asterisk TheCompWiz (~TheCompWi@216.7.159.250) |
| 11:29.11 | TheCompWiz | anyone know how to set a blue-alarm on a PRI span? |
| 11:31.27 | WIMPy | You mean send an alarm indication? |
| 11:39.04 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 11:40.55 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 11:42.11 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 11:44.38 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 11:47.09 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 11:47.09 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 11:50.04 | *** join/#asterisk phunguy (santas@dhcp.i-p.org.uk) |
| 12:21.24 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.132) |
| 12:23.05 | *** join/#asterisk phunguy (santas@dhcp.i-p.org.uk) |
| 12:36.47 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |
| 12:38.32 | *** join/#asterisk Lachesis (~ems@unaffiliated/lachesis) |
| 12:42.07 | *** join/#asterisk phunguy (santas@dhcp.i-p.org.uk) |
| 12:49.44 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 12:50.14 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 12:50.53 | *** join/#asterisk tadspoles (~tadspoles@nc-184-3-103-49.dhcp.embarqhsd.net) |
| 12:50.54 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.125.152) |
| 12:54.29 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 13:04.19 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 13:10.24 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:12.27 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 13:14.51 | *** join/#asterisk areski (~areski@37.233.97.2) |
| 13:40.00 | *** join/#asterisk F-JJTH (~clement@car84-1-78-229-89-36.fbx.proxad.net) |
| 13:40.32 | *** join/#asterisk Tarso (~Tarso@189.61.52.46) |
| 13:40.55 | *** part/#asterisk Tarso (~Tarso@189.61.52.46) |
| 13:51.47 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 13:56.50 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 14:25.33 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.242) |
| 14:28.22 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 14:28.24 | [sr] | hi WIMPy |
| 14:28.32 | WIMPy | Hi [sr] |
| 14:30.45 | *** join/#asterisk angryuser_laptop (~angryuser@ip-189.net-81-220-67.lyon.rev.numericable.fr) |
| 14:31.07 | [sr] | whats up? |
| 14:31.09 | [sr] | cold there? |
| 14:31.32 | WIMPy | Nope, Ok. 16° |
| 14:32.03 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 14:33.12 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:33.20 | [sr] | same here, +/- |
| 14:37.50 | WIMPy | Oops. Is it just me or have 'core restart ...' and 'core stop ...' gone somewhere else? |
| 14:41.58 | igcewieling | it is just you |
| 14:42.25 | igcewieling | or you started asterisk with -c |
| 14:42.40 | WIMPy | Yes, it looks like Asterisk doesn't want to fully start again after I had to kill it a few minutes ago. |
| 14:48.54 | fling | May not I use old pci modem cards somehow? :P |
| 14:48.57 | fling | I have a lot of them |
| 14:49.43 | WIMPy | Yes, but not as channels for Asterisk. |
| 14:51.02 | WIMPy | (Unless yu want to use Asterisk 1.0) |
| 14:51.06 | *** join/#asterisk wonderworld (~w@ip-176-199-167-15.unitymediagroup.de) |
| 14:51.20 | *** join/#asterisk af_ (~getsmart@88-149-224-126.v4.ngi.it) |
| 14:52.29 | wonderworld | hey, is webRTC support in latest asterisk working with chrome or is still a patch or a gateway needed as mentioned here: http://code.google.com/p/sipml5/wiki/Asterisk |
| 14:53.43 | fling | WIMPy: umm :P |
| 14:54.38 | fling | WIMPy: you mean with old asterisk pci modem cards may be used as pstn ports, right? |
| 14:54.52 | WIMPy | Yes |
| 14:55.12 | WIMPy | There used to be a chan_modem. |
| 14:55.16 | fling | I don't have pstn :| |
| 14:55.45 | WIMPy | Then why do you ask about the modems? |
| 14:56.01 | fling | idk what to do with them |
| 14:56.57 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:57.37 | WIMPy | If you have enough of them, maybe you can slightly increase your servers temperature. |
| 14:58.58 | fling | Winters are cold here, ok |
| 14:59.25 | igcewieling | There is this thing called a trash bin....... |
| 15:00.22 | fling | I may playback music using old floppy drives |
| 15:00.39 | fling | But looks like old pci modem cards are useless |
| 15:01.03 | WIMPy | Unless you think there's a point in setting up a BBS today. |
| 15:01.35 | WIMPy | Probably only usefull in-house without PSTN, however. |
| 15:03.38 | igcewieling | watch this, then think about if you REALLY need all that stuff. http://www.youtube.com/watch?v=MvgN5gCuLac |
| 15:08.19 | wonderworld | hey, is webRTC support in latest asterisk working with chrome or is still a patch or a gateway needed as mentioned here: http://code.google.com/p/sipml5/wiki/Asterisk |
| 15:11.54 | *** join/#asterisk markit (~marco@88-149-177-66.v4.ngi.it) |
| 15:16.10 | *** join/#asterisk [sr] (~kvirc@smtp.decimal.pt) |
| 15:17.35 | fling | igcewieling: :D |
| 15:18.12 | igcewieling | minimalism: its the least you can do |
| 15:26.35 | igcewieling | Between Katrina and moving often I don't have a lot of stuff. |
| 15:36.03 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 15:41.16 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 15:46.00 | *** join/#asterisk areski (areski@conference/django-sprint/x-ofzlqyudqreczbvz) |
| 15:52.53 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.66) |
| 15:58.21 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:45c7:f8f8:b7d3:694a) |
| 16:01.45 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.66) |
| 16:04.41 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 16:05.48 | *** join/#asterisk tamiel (~tamiel@182.73.206.166) |
| 16:11.49 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.143) |
| 16:13.52 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.66) |
| 16:33.54 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.66) |
| 16:41.18 | *** join/#asterisk ghost75 (~trechber@188-22-108-102.adsl.highway.telekom.at) |
| 16:44.06 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.66) |
| 16:47.08 | *** join/#asterisk ghost75 (~trechber@188-22-108-102.adsl.highway.telekom.at) |
| 16:48.23 | ghost75 | what could be wrong if forwarding to mobile rings on mobile but if i pickup call, connection is dropped |
| 16:55.52 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 17:06.37 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 17:10.43 | *** join/#asterisk przerull (~philip@50.56.205.232) |
| 17:12.21 | igcewieling | "when I pick up the call the connection drops" == "you have a nat or reinvite problem" |
| 17:12.33 | przerull | hello, so i'm currently using confbridge to bridge the inbound and outbound legs of my calls, but I'm getting complaints that my channels are getting crossed. (people are able to listen into other's conversations) I never had this issue with meetme. Has anyone experienced something like this? i'm on 11.3 |
| 17:13.12 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 17:13.43 | WIMPy | przerull: Then your translation to ConfBridge is broken. |
| 17:14.09 | WIMPy | And if it's only about two parties, you should look into Bridge instead. |
| 17:14.45 | przerull | the thing is we might not always be about two parties (thinking towards the future) |
| 17:15.28 | przerull | basically my system creates a unique callid, then the inbound leg uses orignate to spool the outbound leg |
| 17:15.43 | przerull | it originates to ${callid}@outbound_leg |
| 17:15.59 | przerull | which is the extension that actually dials out to the endpoint using dial |
| 17:16.18 | przerull | the originate sends to ${callid}@bridge_outbound_leg |
| 17:16.50 | przerull | which joins it to the Confbridge conference using ConfBridge(${callid}) |
| 17:17.29 | *** join/#asterisk italorossi (~italoross@201.76.144.193.digi.com.br) |
| 17:18.29 | WIMPy | Eithr those IDs aren't unique or they are not valid conference names. |
| 17:18.57 | przerull | they are large integers |
| 17:19.40 | przerull | I highly doubt they are not unique because if they weren't I would have had the same complaints with MeetMe |
| 17:20.48 | WIMPy | If ConfBride would mix diferent conferences together I'm sure someone would have noticed before. |
| 17:21.02 | WIMPy | I'd check what a valid name is. |
| 17:21.03 | ghost75 | guess the remote party dropped the call because there was no audio |
| 17:21.09 | ghost75 | prolly nat problem |
| 17:22.37 | przerull | thanks |
| 17:22.55 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.141) |
| 17:23.20 | przerull | well on one of my production machines, core show channels shows exactly two copies of confbridge(${callid}) one for each channel |
| 17:24.07 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 17:24.18 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 17:27.02 | przerull | i'll cheis it possible that my conferences could be crossing with conferences on other physical machines? |
| 17:27.21 | przerull | i've got 14 production boxes that each carry about 100 simultanious channels |
| 17:27.41 | WIMPy | "machines"? |
| 17:28.41 | przerull | say one my my production computers is crossing with the conversations on a different computer in my network |
| 17:29.14 | WIMPy | Is the any NAT going on? |
| 17:29.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 17:29.48 | WIMPy | I have seen "crosstalk" at network level. |
| 17:30.23 | przerull | all of my boxes are on the same lan. I only have one peer, my carrier |
| 17:30.57 | WIMPy | But they all have public IPs? |
| 17:31.05 | przerull | yes |
| 17:31.13 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 17:32.07 | WIMPy | That doesn't make issues there likely. |
| 17:37.27 | *** join/#asterisk areski (~areski@217-67-199-154.itsa.net.pl) |
| 17:37.40 | przerull | could it be related ty eh bridge_profile and user_profiles? |
| 17:38.27 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 17:38.49 | WIMPy | Doesn't make sense to me. |
| 17:39.32 | WIMPy | Off course it's always possible that you found a bug, it just seems unlikely for that kind of bug to have gone undiscovered so far. |
| 17:47.49 | *** join/#asterisk n3hxs (~ed@pool-108-16-94-10.phlapa.fios.verizon.net) |
| 17:48.03 | ghost75 | asterisk-de so empty |
| 17:51.23 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 17:55.46 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 17:56.04 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 17:56.17 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:56.30 | saint_ | hi all - can someone tell me what is the best way to upgrade an asterisk version, and specially how to keep the selected choices in make menuselect ? I don t remember what I turned on / off in my previous version. is there a way to carry that on ? |
| 17:59.22 | WIMPy | Copy menuselect.makeopts over. But there might be new options, off course. |
| 18:00.14 | saint_ | WIMPy: if there are new options, would digium team note them in UPGRADE.TXT ? |
| 18:00.43 | WIMPy | I don't think make options are noted there. |
| 18:00.48 | WIMPy | (or anywhere) |
| 18:01.09 | [TK]D-Fender | possibly the raw changelog. |
| 18:02.45 | saint_ | mmhh.. so to be safe, i should run the one in the old version, run the one in the new version , and compare line by line .. |
| 18:03.26 | *** join/#asterisk Free99 (~Free99@cpe-66-108-105-10.nyc.res.rr.com) |
| 18:06.57 | [TK]D-Fender | saint_: you should also be clear as to what version you care coming FROM, and which one you are going TO. |
| 18:07.27 | saint_ | [TK]D-Fender: just 1.8.5 cert 1 to cert 2 |
| 18:11.46 | [TK]D-Fender | saitthat implies minor bug and security fixes. nothing really changes there |
| 18:12.01 | [TK]D-Fender | YIt generally doesn't change within the entire branch |
| 18:18.43 | *** join/#asterisk italorossi (~italoross@201.76.144.193.digi.com.br) |
| 18:20.15 | *** join/#asterisk v4x (~v4x@unaffiliated/v4x) |
| 18:27.43 | *** join/#asterisk angryuser_laptop (~angryuser@ip-189.net-81-220-67.lyon.rev.numericable.fr) |
| 18:31.42 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 18:35.43 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 19:00.23 | *** join/#asterisk tadspoles (~tadspoles@nc-184-3-103-49.dhcp.embarqhsd.net) |
| 19:03.57 | *** join/#asterisk italorossi (~italoross@201.76.144.193.digi.com.br) |
| 19:05.54 | *** join/#asterisk italoros_ (~italoross@187.61.168.117) |
| 19:22.29 | apb1963 | I'm trying to update a 1.4 extensions.conf file to 11.2.1 It defines a variable at the top of the file: var = value It would appear from the docs (and the error in the logfile) that this can no longer be done that way. |
| 19:23.25 | apb1963 | As I understand it, I now have to use SET, and it has to be part of an exten => ... such as exten => set(var=value) is that correct? |
| 19:23.30 | igcewieling | apb1963: incorrect |
| 19:24.20 | igcewieling | maybe you are missing your [globals] section? |
| 19:24.42 | igcewieling | apb1963: major changes are documented in the UPGRADE-*.txt files. |
| 19:25.55 | apb1963 | Which part is incorrect? The 1.4 original I'm trying to upgrade, or the way I've interpreted the docs? |
| 19:26.06 | apb1963 | for 11. |
| 19:26.52 | apb1963 | Yo're saying it still can be done the way it was done in 1.4 with a [globals] section? |
| 19:27.36 | apb1963 | Or that it should have been done that way in the first place? |
| 19:27.40 | apb1963 | Or something else entirely? |
| 19:28.00 | *** join/#asterisk areski (~areski@217-67-199-154.itsa.net.pl) |
| 19:29.35 | *** join/#asterisk tadspoles (~tadspoles@nc-184-3-103-49.dhcp.embarqhsd.net) |
| 19:33.25 | apb1963 | jiggles his network cable |
| 19:35.32 | [TK]D-Fender | [15:22]apb1963I'm trying to update a 1.4 extensions.conf file to 11.2.1 It defines a variable at the top of the file: var = value It would appear from the docs (and the error in the logfile) that this can no longer be done that way. <- it's not "at the top" .. it's IN the [globals] section as igcewieling told you |
| 19:35.40 | [TK]D-Fender | apb1963: that has not changed. |
| 19:36.00 | [TK]D-Fender | apb1963: Set() is for RUN-TIME setting of variables and has been the same since 1.2 |
| 19:36.21 | igcewieling | [globals] has been available since at least version 0.65 |
| 19:44.32 | apb1963 | No... it's at the top. I'm not saying it's right, I'm saying that's the current situation as it was given to me. No, it doesn't work - I made the assumption it did in 1.4, perhaps it never worked.... i really don't know. I'm just trying to do it properly. |
| 19:45.15 | [TK]D-Fender | Stop saying "top" |
| 19:45.19 | [TK]D-Fender | it's not "top'. |
| 19:45.28 | apb1963 | top of the file |
| 19:45.33 | [TK]D-Fender | It's in the [globals] section. |
| 19:45.47 | [TK]D-Fender | LOCATION OF THAT SECTION IS NOT IMPORTANT |
| 19:45.52 | [TK]D-Fender | Stop saying "top" |
| 19:46.03 | apb1963 | It's uppermost in the file |
| 19:46.06 | przerull | igcewieling: thanks for all the help these past few weeks. I really appreciate it. So what are your views on running asterisk at realtime priority (I think it's -p) |
| 19:46.10 | [TK]D-Fender | I can put [globals] at the bottom of 10000 likns of OTHER dialplan. |
| 19:46.24 | apb1963 | likns? |
| 19:46.31 | [TK]D-Fender | apb1963: ASECTION NAME NOT LOCATION |
| 19:46.48 | apb1963 | it's not in a section |
| 19:46.54 | igcewieling | przerull: When I did that bad things happened but it was likely because I was also running a database on the same box |
| 19:46.55 | [TK]D-Fender | [globals] <- has to be THERE. Doesn't matter if that is at the TOP or at the BOTTM. |
| 19:47.01 | [TK]D-Fender | it has to be IN THAT SECTION |
| 19:47.03 | przerull | ah |
| 19:47.06 | [TK]D-Fender | IT HAS TO BE |
| 19:47.21 | [TK]D-Fender | Hopefully this is clear now.... |
| 19:47.31 | apb1963 | ok so if I have var=value followed by [globals] then var is still in the globals section? |
| 19:47.34 | przerull | thanks I had a feeling that might be the case |
| 19:47.58 | igcewieling | apb1963: works the same as any other context |
| 19:48.12 | igcewieling | [TK]D-Fender: apparently it was not clear enough. 8-| |
| 19:48.20 | [TK]D-Fender | [15:47]apb1963ok so if I have var=value followed by [globals] then var is still in the globals section? <- NO |
| 19:48.22 | apb1963 | thats what I thought... so then this particular var is not in any section |
| 19:48.35 | igcewieling | apb1963: and that is why it is failing |
| 19:48.36 | [TK]D-Fender | apb1963: there is no other place that is even VALID |
| 19:49.09 | apb1963 | igcewieling: I get that. That's why I'm asking questions. |
| 19:49.10 | [TK]D-Fender | apb1963: You can't put "var=value" just anywhere. The ONLY place that is valid is under [globals] |
| 19:49.17 | igcewieling | You'd think he'd check the extensions.conf.sample which came with his asterisk for an example. |
| 19:49.18 | [TK]D-Fender | ^^^ |
| 19:49.32 | [TK]D-Fender | apb1963: Put it anywhere else and it is IGNORED |
| 19:49.56 | apb1963 | I was GIVEN this file. This is how it IS. It's a file from 1.4. |
| 19:50.09 | apb1963 | I get that it doesn't work in 11. |
| 19:50.24 | apb1963 | I do not know if it worked in 1.4 |
| 19:50.51 | apb1963 | I understand it is IGNORED |
| 19:51.05 | apb1963 | I understand I can fix it by putting it in [globals] |
| 19:51.40 | apb1963 | What I'm ASKING, is if there is any other way to set it... like by using the SET function. |
| 19:52.09 | [TK]D-Fender | apb1963: Set is called when actually EXECUTED in the dialplan. |
| 19:52.20 | apb1963 | Fine. So that will work then. |
| 19:52.20 | [TK]D-Fender | apb1963: one is gloabl which means it's ALWAYS there. |
| 19:52.33 | apb1963 | Yes, I get that. |
| 19:52.46 | apb1963 | This particular variable is used in exactly one place... |
| 19:52.48 | [TK]D-Fender | apb1963: So do you understand variables at all? Then you should know what Set() does. |
| 19:52.58 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 19:52.59 | apb1963 | So there's no need for it to be a global. |
| 19:53.19 | [TK]D-Fender | apb1963: We don't knwow hat you're using it for. |
| 19:53.24 | [TK]D-Fender | apb1963: We can't answer that for you./ |
| 19:53.37 | apb1963 | I know. It's used as part of an extension. There, and only there. |
| 19:53.52 | apb1963 | I simply wanted to know if there was any other way to do it. |
| 19:53.56 | apb1963 | Now I know there's not. |
| 19:53.58 | apb1963 | Thank you |
| 19:54.45 | apb1963 | goes back to self-flagellation |
| 20:12.38 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 20:14.33 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 20:15.04 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 20:23.23 | *** join/#asterisk italorossi (~italoross@ec2-54-232-199-58.sa-east-1.compute.amazonaws.com) |
| 20:26.46 | *** join/#asterisk suporte85 (~guardadig@201-43-142-229.dsl.telesp.net.br) |
| 20:37.23 | *** join/#asterisk italoros_ (~italoross@187.61.168.117) |
| 20:40.41 | *** join/#asterisk Mahmoud (~Mahmoud@unaffiliated/mahmoud) |
| 20:41.26 | Mahmoud | hello, what are my options for integrating * with VoIP providers, like GTalk, Skype...etc? (no SIP stuff) |
| 20:46.58 | [TK]D-Fender | Skype requires some 3rd party conversion. Gtalk is supported with chan_motif |
| 20:48.26 | Mahmoud | are there any problems with GTalk's integration? |
| 20:48.55 | Mahmoud | I'm doing this for a business, it will be critical to work nicely |
| 20:49.09 | pabelanger | Mahmoud, I wouldn't run a business off it |
| 20:49.20 | Mahmoud | why? |
| 20:50.00 | Mahmoud | how does it compare to FS's Dingaling module? |
| 20:50.16 | pabelanger | Because it is a free service, you get what you pay for |
| 20:50.20 | [TK]D-Fender | Google Talk is the problem |
| 20:50.23 | [TK]D-Fender | It is a FREE service |
| 20:50.34 | pabelanger | You'd run a business off a free service? |
| 20:50.49 | [TK]D-Fender | If they own that number then when they decide to stop offering it for free, or just cancel your account for your usage of it then your business is in big trouble |
| 20:50.58 | [TK]D-Fender | This is not a service you should base a business on |
| 20:51.00 | Mahmoud | but would chan_motif cause any unreliability issue that a Google GTalk client wouldn't? |
| 20:51.10 | [TK]D-Fender | And chan_motif DOES work |
| 20:52.14 | pabelanger | You also don't get quality of service using google talk |
| 20:52.17 | pabelanger | vs an ITSP |
| 20:52.29 | Mahmoud | yeah |
| 20:52.43 | Mahmoud | What i care is getting same software reliability as in using a Google Gtalk client |
| 20:52.53 | [TK]D-Fender | It works. |
| 20:53.09 | pabelanger | until google changes the protocol |
| 20:53.14 | pabelanger | which has happened in the past |
| 20:53.17 | Mahmoud | does Google often do that? |
| 20:53.21 | Mahmoud | oh |
| 20:53.23 | [TK]D-Fender | Which is something that has happened before and could happen again |
| 20:53.31 | pabelanger | I have not used chan_mofit, but seems to be the best right now |
| 20:53.47 | pabelanger | Google does not announce when they change it, so service just stops working |
| 20:53.54 | pabelanger | Again, not the best setup for a business |
| 20:54.10 | pabelanger | for testing and development, sure. But not production |
| 20:54.38 | Mahmoud | the thing is, there are many clients with those apps: gtalk, skype, WeChat...etc |
| 20:54.42 | [TK]D-Fender | Quicksand makes for a poor foundation. |
| 20:54.57 | [TK]D-Fender | Mahmoud: How many clients there are does not matter. |
| 20:55.06 | [TK]D-Fender | Mahmoud: When the protocol changes they all stop working |
| 20:55.17 | pabelanger | Mahmoud, Why do you want to use google talk? |
| 20:55.20 | Mahmoud | i get this. but is there any other better alternative of such cool client? |
| 20:55.39 | Mahmoud | people use it, and it's a relatively old one that was integrated so I assume it's most reliable cool interface |
| 20:55.51 | [TK]D-Fender | a REAL provider with a protocol that is STANDARDIZED |
| 20:55.58 | [TK]D-Fender | People are idiots |
| 20:56.05 | Mahmoud | i do have that |
| 20:56.16 | Mahmoud | i want to adapt my idiotic users |
| 20:56.34 | [TK]D-Fender | So go do it |
| 20:56.38 | Mahmoud | money comes from them. they seem to not be complete idiots (which is how they collected some money) |
| 20:57.12 | Mahmoud | but is there any better alternative to GTalk (SIP /IAX excluded) |
| 20:57.18 | [TK]D-Fender | Mahmoud: Just because you're good at X does not mean you are not a complete idiot when it comes t Y |
| 20:57.28 | Mahmoud | agree |
| 20:57.49 | pabelanger | Mahmoud, you still have not stated what you want to do. So how can we recommend a protocol |
| 20:57.50 | [TK]D-Fender | [16:57]Mahmoudbut is there any better alternative to GTalk (SIP /IAX excluded) <- YOU seem to be specifically avoiding STANDARDS. YOU are putting your platform at risk |
| 20:58.47 | Mahmoud | I do use standards, but want to expand it to non-standards too |
| 20:58.52 | [TK]D-Fender | Mahmoud: You want "cool" and "stable", and you are the one avoiding "standards". You have a serious philosphical issue. What you are asking for does not exist. |
| 20:59.11 | Mahmoud | I want the msot stable cool |
| 20:59.21 | Mahmoud | s/msot/most/ |
| 20:59.23 | [TK]D-Fender | Mahmoud: No such thing |
| 20:59.49 | Mahmoud | i.e. there are plenty of cool non-standard voip stuff, they are crappy. but among them there is one that is least crappy, I want it |
| 20:59.59 | [TK]D-Fender | Mahmoud: ANY of them can be ripped out from under you in multiple ways. |
| 21:00.11 | [TK]D-Fender | Mahmoud: Gtalk is good NOW. |
| 21:00.16 | [TK]D-Fender | Mahmoud: That could change TOMORROW |
| 21:00.32 | pabelanger | Mahmoud, how much are you willing to pay for it? |
| 21:00.44 | Mahmoud | zero USD |
| 21:00.47 | [TK]D-Fender | Mahmoud: There is no point to farther elaboration on this. |
| 21:01.03 | [TK]D-Fender | Mahmoud: Expect every solution you look at to fail at some point then |
| 21:01.05 | pabelanger | http://en.wikipedia.org/wiki/Project_management_triangle |
| 21:01.11 | [TK]D-Fender | Mahmoud: And you are basing a business on this |
| 21:01.15 | pabelanger | Mahmoud, you can only pick two ^ |
| 21:01.18 | [TK]D-Fender | Mahmoud: Good luck with this suicide plan |
| 21:01.33 | Mahmoud | heh |
| 21:01.37 | Mahmoud | it is risky... |
| 21:01.58 | Mahmoud | last but not least: any thoughts on how mod_motif compares against FS's DIngaling? |
| 21:02.03 | [TK]D-Fender | IT WORKS |
| 21:02.05 | [TK]D-Fender | what is unclear |
| 21:02.07 | [TK]D-Fender | It works |
| 21:02.13 | [TK]D-Fender | it does exactly what it's supposed to |
| 21:02.30 | [TK]D-Fender | And when Gtalk changes their protocol then BOTH die. |
| 21:02.54 | Mahmoud | i get that |
| 21:07.13 | file | Google is unifying their messaging and using something besides XMPP, so how long Google Voice will be around is... probably not that long |
| 21:08.08 | [TK]D-Fender | TelecomRoulette (tm) |
| 21:08.15 | pabelanger | GGG file writes google channel driver for asterisk; Google changes protocols |
| 21:09.09 | file | what the new protocol is like also remains unknown |
| 21:10.25 | pabelanger | Did they announce it at google IO, or was it just a blog posting? |
| 21:12.16 | file | I didn't watch Google I/O, but I read it on the netz |
| 21:25.15 | Mahmoud | file: why would they unify all their protocols? |
| 21:26.04 | file | because using a single one means maximum compatibility, less stuff to manage, and ultimately more features to more stuff? |
| 21:26.41 | file | and using their own means they can rapidly do stuff, as is the Google way |
| 21:34.45 | Mahmoud | would it be XML based or text based? |
| 21:35.02 | Mahmoud | i think XML is a bad idea, honestly. for anything. |
| 21:46.29 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:47.22 | *** join/#asterisk debuggerboy (~debuggerb@117.196.174.93) |
| 21:48.03 | *** part/#asterisk debuggerboy (~debuggerb@117.196.174.93) |
| 21:53.50 | *** join/#asterisk suporte85 (~guardadig@177.138.244.187) |
| 22:00.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:05.39 | *** join/#asterisk Rahail (~Rahail@67.214.121.163) |
| 22:05.42 | Rahail | HI every one |
| 22:06.28 | Rahail | I am using iax2 to register point a to point b for anyresaon if soemthign goes wrong to point a after this get fix or if internet drop when internet come back point a will not register to point b every time i have to restart both any idea |
| 22:06.32 | Rahail | how can i fix this |
| 22:12.35 | Rahail | hello |
| 22:16.59 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:40.55 | [TK]D-Fender | Rahail: http://svnview.digium.com/svn/asterisk/trunk/configs/iax.conf.sample?view=markup |
| 22:42.48 | Rahail | [TK]D-Fender sorry what should i look for on that page |
| 22:42.59 | Rahail | iax.conf |
| 22:43.04 | Rahail | should i repalce with mine with this |
| 22:43.15 | [TK]D-Fender | Rahail: Somthing that obviously sets rules for how often it should re-register. |
| 22:43.39 | [TK]D-Fender | Rahail: Read the parameters and set them right for YOU |
| 22:43.50 | Rahail | ~pastebin |
| 22:43.50 | infobot | A "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://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 22:44.16 | Rahail | http://pastebin.com/HVc0e8QA |
| 22:48.40 | *** join/#asterisk areski (~areski@217-67-199-154.itsa.net.pl) |
| 22:53.02 | Rahail | should i add this 2 ; minregexpire = 60 |
| 22:53.02 | Rahail | ; maxregexpire = 60 |
| 22:55.28 | [TK]D-Fender | Rahail: What do you think? |
| 22:55.53 | Rahail | i think this exprice regis on 60s |
| 22:55.57 | Rahail | ? |
| 22:56.45 | Rahail | or is it for if registraion exprie it need to send reques until x s then stop |
| 22:56.52 | [TK]D-Fender | Rahail: And is that good enough for you? |
| 22:57.23 | Rahail | I am not sure [TK]D-Fender this why i am here so people who is more exprince then me |
| 22:57.28 | Rahail | can give me advice |
| 23:00.03 | [TK]D-Fender | Rahail: I advise you tell think if 60 SECONDS is enough for you to wait for it to register again |
| 23:00.37 | Rahail | no |
| 23:00.42 | Rahail | i need this to register right way |
| 23:00.46 | Rahail | if register exprice |
| 23:00.52 | Rahail | or if register drop for any reason |
| 23:01.02 | [TK]D-Fender | Rahail: You don't understand. * don't magically KNOW when your internet is "back" |
| 23:01.24 | [TK]D-Fender | Rahail: So you have to tell it HOW OFTEN to try |
| 23:01.24 | Rahail | can i just put continue trying |
| 23:01.24 | Rahail | then |
| 23:01.34 | [TK]D-Fender | it does... its a question of HOW OFTEN |
| 23:01.44 | Rahail | letsay every 5s |
| 23:01.57 | [TK]D-Fender | Rahail: that is not a RETRY timeout. |
| 23:02.03 | [TK]D-Fender | Rahail: You aren't thinking very hard on this. |
| 23:02.25 | Rahail | :( |
| 23:02.52 | [TK]D-Fender | Rahail: Do you want it wasting time every FIVE SECONDS registering even when it's fine? |
| 23:03.13 | Rahail | no i do not want that way if is regsiter |
| 23:03.18 | Rahail | when its not regsiter then I want it |
| 23:03.31 | [TK]D-Fender | Rahail: Does not work that way. |
| 23:03.53 | Rahail | OK then which way would you recomend |
| 23:04.23 | Mahmoud | any better FOSS SIP PBX than *? |
| 23:04.28 | [TK]D-Fender | Rahail: The way the parameters tell you it can. |
| 23:04.35 | Mahmoud | suppose that i need sip end-to-end |
| 23:04.50 | Rahail | [TK]D-Fender can you tell me the parmeter |
| 23:04.51 | Rahail | please |
| 23:05.00 | [TK]D-Fender | Mahmoud: Do you go McDonalds and ask them if Burger Kink is better? What kind of answer do you think you'll get? |
| 23:05.22 | Mahmoud | [TK]D-Fender: i wouldn't ask a salesman. but I would certainly not mind asking an Engineer :) |
| 23:05.44 | [TK]D-Fender | Mahmoud: the answer is NO ONE. |
| 23:06.04 | [TK]D-Fender | Mahmoud: So the short answer is :Asterisk |
| 23:06.22 | Mahmoud | does * still use mod_sip as implemented years ago? how about Sofia? |
| 23:06.43 | [TK]D-Fender | Mahmoud: Where do you EVER see * using mod_sip? |
| 23:07.06 | Mahmoud | hmm i somehow recall it doing so back in around 2007 |
| 23:07.19 | Rahail | [TK]D-Fender which parmeter i should use on iax conf |
| 23:07.27 | [TK]D-Fender | Rahail: THOSE |
| 23:07.37 | Rahail | 60 |
| 23:07.51 | [TK]D-Fender | "60" isn't a parameter |
| 23:08.04 | Rahail | so this in 60s it will exprice and it will trying another 60s to register |
| 23:08.12 | Rahail | ; minregexpire = 60 |
| 23:08.13 | Rahail | ; maxregexpire = 60 |
| 23:08.15 | Rahail | this right |
| 23:08.23 | [TK]D-Fender | Rahail: What does it SAY? |
| 23:08.50 | Rahail | top of it |
| 23:08.54 | [TK]D-Fender | [19:07]Mahmoudhmm i somehow recall it doing so back in around 2007 <- your seemingly old memories of Asterisk are incorrect. |
| 23:09.04 | Rahail | minu and maxi amount of the time iax2 peer can request as reg |
| 23:12.13 | Rahail | [TK]D-Fender so how many 60s should i put on those parment |
| 23:12.18 | Rahail | parameter to be safe |
| 23:12.55 | [TK]D-Fender | Rahail: What is "safe"? You need to have a basic understanding of time. |
| 23:13.01 | file | Mahmoud, I have no idea - I don't work for Google |
| 23:13.15 | [TK]D-Fender | file: You get PM'd for that? |
| 23:13.25 | file | I don't think so, I just noticed it |
| 23:14.51 | Rahail | [TK]D-Fender but time you would put |
| 23:15.37 | [TK]D-Fender | Rahail: Depends how much I'd want to batter my server with the hosts that would be hitting it. |
| 23:15.48 | *** join/#asterisk corretico (~luis@186.96.91.239) |
| 23:16.33 | Rahail | SO if i put 360 so registration will expire in 360s right |
| 23:16.52 | Rahail | and after that it will send the req again and again untill it regsiter ? |
| 23:17.01 | Rahail | or after trying 360s it will stop trying |
| 23:17.06 | [TK]D-Fender | NEITHER |
| 23:17.11 | [TK]D-Fender | you seem to have a reading problem |
| 23:17.23 | Rahail | i think so |
| 23:17.26 | Rahail | not getting in my head |
| 23:17.35 | [TK]D-Fender | It will force the peer to register EVERY 360 seconds |
| 23:17.42 | [TK]D-Fender | this has NOTHING to do with "failure" |
| 23:17.47 | [TK]D-Fender | EVERY 360 seconds |
| 23:18.33 | [TK]D-Fender | So when you suggested "5 seconds" imagine all of your friends knocking on your door every 5 seconds to say "are you alive?" |
| 23:18.47 | [TK]D-Fender | And if you can't answer fast enough ... they think you're dead |
| 23:19.01 | Rahail | for how long they will try |
| 23:19.14 | [TK]D-Fender | UNTIL IT SUCCEEDS |
| 23:19.17 | Rahail | lately i have to reboot both end to get regsiter.. |
| 23:19.19 | [TK]D-Fender | 'and then MORE |
| 23:19.36 | [TK]D-Fender | If the interval is long then maybe it's just WAITING a long time |
| 23:19.50 | [TK]D-Fender | And you aren't setting it right now/ |
| 23:20.34 | Rahail | i do not get this part hten i do not have this parameter.. however it regstier once in while i lost regsitraion for failure of net or antying.. that time it dont register |
| 23:20.43 | Rahail | so i had to restart the asterisk |
| 23:20.57 | Rahail | in order to get the registraion work.. right after i restart boom it register |
| 23:22.06 | [TK]D-Fender | because restarting resets the clock |
| 23:22.14 | [TK]D-Fender | Doesn't mean it isn't still running. |
| 23:22.49 | *** join/#asterisk mivok (~mark@c-68-48-166-138.hsd1.md.comcast.net) |
| 23:22.55 | Rahail | what can i do that will do it auto lot of time i am not in computer |
| 23:23.09 | Rahail | and call dont pass cuz of the failure |
| 23:23.24 | Rahail | or should i put on cronn to restart asterisk every 3 hr |
| 23:26.01 | Rahail | ? |
| 23:29.25 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 23:47.17 | *** join/#asterisk tadspoles (~tadspoles@nc-184-3-103-49.dhcp.embarqhsd.net) |
| 23:49.58 | *** join/#asterisk Free99 (~Free99@cpe-66-108-105-10.nyc.res.rr.com) |
| 23:50.15 | Free99 | hey everyone. how do find more info on E911 requirements? |
| 23:50.25 | Rahail | [TK]D-Fender i think i should have ask if any one can make me sh script |
| 23:50.42 | Rahail | and i run that on cron to check registraion on every x minute if no registraion then restart the asterisk |
| 23:50.58 | WIMPy | Free99: The website of the authority responsible? |
| 23:52.00 | Free99 | WIMPy, issue is that my system is going to be used internationally.. I know I'm not obligated to provide the service to anyone outside the NANPA areas, but I'd kind of like to, so long as it isn't ridiculous |
| 23:52.28 | *** join/#asterisk suporte85 (~guardadig@177.138.244.187) |
| 23:52.53 | WIMPy | Free99: By now the same is true for the whole EU as well, as far as I know. |
| 23:53.41 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 23:53.41 | Free99 | if you're doing VoIP on a ship, then... how the heck do I link location with register? lol. Doesn't seem easily possible at all |
| 23:53.41 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:53.56 | WIMPy | True story. |
| 23:54.35 | Free99 | ah well. Guess I'll just record a message stating that emergency services are not available if they dial 911 or the other usual emergency numbers |
| 23:54.46 | WIMPy | Probably the reason while mobile phones need GPS. Although I don't think ist's used that way anywhere. |
| 23:56.36 | Free99 | WIMPy, just out of curiosity, when one registers, is your user/pass being sent cleartext? |
| 23:56.42 | Free99 | I haven't setup TLS yet |
| 23:57.09 | WIMPy | I think it was Thuraya who use GPS to find out which country you're in. |
| 23:57.20 | WIMPy | no |
| 23:57.59 | WIMPy | It's not strong encryption, but I guess you could call it reasonable mangled. |
| 23:58.11 | Free99 | you mean like md5 or something? |
| 23:58.24 | WIMPy | yes |
| 23:59.48 | Free99 | that's... not so great tbh |