| 00:11.35 | *** join/#asterisk malachi_constant (~root@unaffiliated/malachiconstant/x-837457) |
| 00:11.40 | malachi_constant | Hey there folks. |
| 00:13.01 | malachi_constant | I'm getting 401 unauthorized in response to registration requests by my channel bank. Username and host= in my sip.conf match: http://pastebin.com/eWQtYnbn |
| 00:14.02 | malachi_constant | Any ideas? |
| 00:14.13 | [TK]D-Fender | You have TWO devices with the same IP |
| 00:14.52 | [TK]D-Fender | First bad thing |
| 00:15.06 | [TK]D-Fender | Second is a 101 grade mistake. |
| 00:15.18 | malachi_constant | Eeep. I do. |
| 00:15.21 | WIMPy | You register with the device? Isn't that the wrong way round? |
| 00:15.22 | [TK]D-Fender | Devices with a fixed host are not ALLOWED to register |
| 00:15.34 | malachi_constant | The device registers with asterisk. |
| 00:15.40 | malachi_constant | Oooh. |
| 00:15.43 | [TK]D-Fender | Which is not allowed |
| 00:15.45 | malachi_constant | So I need host=dynamic. |
| 00:16.02 | [TK]D-Fender | The point of registration is to tell the other side where to find you. If you're fixed you don't get to tell them where. |
| 00:16.09 | WIMPy | If you know the IP, there's no need to register. |
| 00:16.50 | malachi_constant | Got it. |
| 00:16.56 | malachi_constant | Thanks guys. |
| 00:29.33 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 00:46.33 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.66.133) |
| 00:54.04 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 01:00.51 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:19.42 | *** join/#asterisk [44] (~chris@unaffiliated/quintux) |
| 01:24.02 | *** part/#asterisk WangDang_ (~ger@24.140.224.212) |
| 01:37.18 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:45.40 | *** part/#asterisk rexwin (~rexwin@106.208.74.171) |
| 01:47.11 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-fgyckkcbzfyuhagq) |
| 01:51.28 | *** join/#asterisk petris (sid19918@gateway/web/irccloud.com/x-serdtdaxyzxtptra) |
| 02:11.22 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 02:39.30 | *** join/#asterisk CrashHD (~na@104.220.5.4) |
| 02:41.06 | *** join/#asterisk PaybackTony (~Adium@2601:7:700:7fa:e822:27f2:16a:e87e) |
| 02:47.00 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
| 03:14.41 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 05:20.49 | *** join/#asterisk ayrjola (~ayrjola@89.18.237.137) |
| 05:33.51 | *** join/#asterisk gryphon_ (~gryphon@82.140.120.164) |
| 06:03.40 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 06:19.34 | *** join/#asterisk mokmeister (~quassel@95.45.40.109) |
| 06:21.56 | *** join/#asterisk c0rnoTa (~c0rnoTa@kobratel.starlink.ru) |
| 06:29.01 | *** join/#asterisk D30 (~D30@58.71.19.178) |
| 06:30.01 | *** join/#asterisk zblk (~andrey@92.53.115.234) |
| 06:33.27 | *** join/#asterisk infina (~infina@unaffiliated/infina) |
| 06:35.23 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 06:47.48 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:53.17 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 06:59.51 | *** join/#asterisk ghostlines (~ghostline@i191002.upc-i.chello.nl) |
| 07:07.06 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 07:15.44 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 07:23.10 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 07:24.38 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 07:30.12 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 07:44.07 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 07:54.23 | *** join/#asterisk CeBe (~CeBe@port-92-200-130-91.dynamic.qsc.de) |
| 08:14.27 | *** join/#asterisk BarthezZ (~bart@monitoring.deheij-ict.nl) |
| 08:16.16 | *** join/#asterisk ayrjola_ (~ayrjola@89.18.237.137) |
| 08:18.51 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:19.54 | *** join/#asterisk Kilo`byte (~kilobyte@5.231.51.78) |
| 08:24.50 | *** join/#asterisk skirmisha (~skirmish@46.47.82.189) |
| 08:25.02 | skirmisha | hi guys |
| 08:25.26 | skirmisha | any workaround to make asterisk match correct peer on inbound calls? |
| 08:25.37 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 08:25.48 | skirmisha | without using registration or username |
| 08:28.01 | *** join/#asterisk D30 (~D30@58.71.19.178) |
| 08:28.04 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 08:29.05 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:30.01 | skirmisha | any ideas? |
| 08:41.53 | Gugge | skirmisha: sure, match on ip/port |
| 08:48.09 | skirmisha | that's the problem |
| 08:48.19 | skirmisha | i have multiple peers with same ip and port |
| 08:48.28 | skirmisha | and its matching first found |
| 08:49.02 | skirmisha | strange thing is that behavior is not seen in older ver of asterisk |
| 08:49.25 | skirmisha | i am currently testing on 13.1 and i am stuck with this issue and cant control incoming calls |
| 08:52.19 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:52.26 | skirmisha | is there a field that has priority and it is checked first? |
| 08:52.36 | skirmisha | then i can set under peer how to match it |
| 09:00.58 | ChannelZ | How can you have multiple peers on the same IP AND port? How would you expect the traffic to get to the right one in the first place? |
| 09:02.39 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 09:02.50 | ChannelZ | And otherwise how, if you don't want to match by IP/port, OR without registration or username, do you expect it to work? Magic? |
| 09:04.49 | skirmisha | ok lets say i have different users with different needs, but using same trunk for outgoing calls. Therefore i have separate peer for each user, but they are all with same ip. However incoming calls are matched to just wrong peer as there are many of them with same ip for outgoing calls, Thus i am not managed to control incoming calls. I have created separate peer for the incoming calls and pointing to correct context, but it is not matched as * match |
| 09:04.50 | skirmisha | against one of the outgoing peers/trunk set for users |
| 09:07.11 | *** join/#asterisk CustosL1men (~CustosLim@unaffiliated/cust0slim3n) |
| 09:09.03 | skirmisha | i have tested everything from type peer, user, friend, default ip. nothing is working as expected. on the outgoing trunks/peers there is only ip and type set nothing else |
| 09:09.54 | AnonGirl | skirmisha: reconsider your entire setup |
| 09:10.01 | ChannelZ | Asterisk can match by IP AND port. Many peers can have the same IP (if they are behind NAT for instance) but they will all have to have unique port numbers, otherwise it's impossible for Asterisk to send a call to the proper peer (the NAT wouldn't have any idea which device on the local side of the network to send the packets to.) |
| 09:10.55 | ChannelZ | The easiest way to do this is with registration, so the peers will get the IP and port for each peer, but if you reeeeally don't want to do that, then you've got to assign static port numbers yourself and put them in for each peer in sip.conf |
| 09:11.25 | skirmisha | there is no nat and why should i reconsider my setup. same setup with ast ver 1.4 works with no problem, now with latest 13.1 ver it does not. No patches made |
| 09:14.00 | ChannelZ | Well like I said, you either need to make them register, or hardcode the IP & port yourself for each one |
| 09:14.45 | *** join/#asterisk PaybackTony (~Adium@2601:7:700:7fa:e822:27f2:16a:e87e) |
| 09:17.48 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 09:17.55 | ChannelZ | What type were you using before? (type=xxx in sip.conf) |
| 09:22.00 | skirmisha | same on all is peer |
| 09:22.05 | skirmisha | also on ver 1.4 |
| 09:22.28 | skirmisha | there is no nat and port is same as the ip |
| 09:22.42 | ChannelZ | ok.. so then was host=dynamic or did you have them statically configured? |
| 09:23.06 | skirmisha | nope, host is ip of my peer and type is peer |
| 09:23.15 | ChannelZ | And again, it's not possible to have multiple devices on the same IP *and* port |
| 09:23.32 | skirmisha | because peer sends incoming calls from same ip it is matching one of the outgoing peers/trunks instead the one i configured |
| 09:24.11 | ChannelZ | That doesn't make sense. |
| 09:24.26 | ChannelZ | We'd really need to see some configs to figure out what the heck you've got going on |
| 09:27.37 | skirmisha | config is very simple - user a , normal user - > going to context with use peer/trunk 1, user 2 -> second context using second peer/trunk. Both trunks are identical - only host and type set. Difference is done in the context of users , So far no incoming call, that's only outgoing |
| 09:28.27 | skirmisha | third peer/trunk configured for incoming calls and forward to correct context - provider send call to us, asterisk match peer/trunk of first user |
| 09:28.54 | skirmisha | i tried to allow anonymous incoming calls but that is not matched |
| 09:29.21 | skirmisha | on ver 1.4 that do the trick if i do not specify it, anonymous is matched and call flow is managed that way |
| 09:29.34 | ChannelZ | Then you have something really screwed up for calls from your provider to be matching the peer of one of your devices. |
| 09:30.39 | ChannelZ | Allowing guest calls means any call coming from someone that can't be matched to a configured user/peer will still be accepted (into a separate context) which again suggests your config is not correct. |
| 09:31.01 | skirmisha | it is so called - asterisk multiple peers same ip |
| 09:31.34 | skirmisha | issue is that asterisk match against first peer/trunk set for user 1 outgoing calls |
| 09:31.37 | ChannelZ | So you're getting calls from your provider but have peers from your provider too? That makes no sense. |
| 09:34.03 | ChannelZ | Show us a sip debug of one of these incoming calls from your provider that is matching the wrong peer in your config. And then show us the entry in sip.conf that it's *supposed* to be matching to. |
| 09:34.23 | ChannelZ | (just delete the secret=xxx) |
| 09:34.38 | ChannelZ | (if it even has one) |
| 09:34.58 | *** join/#asterisk Darkerr (~Libor@static-84-42-235-44.net.upcbroadband.cz) |
| 09:36.30 | Darkerr | hello, is it possible to limit number of sip accounts (extensions) in asterisk? |
| 09:36.44 | skirmisha | one sec |
| 09:37.15 | ChannelZ | Darkerr, errr... sip accounts and extensions are two totally different things |
| 09:38.10 | ChannelZ | And in either case, you 'limit' them by only configuring the amount you need... ? It's kind of a backwards question, can you elaborate? |
| 09:39.48 | *** part/#asterisk c0rnoTa (~c0rnoTa@kobratel.starlink.ru) |
| 09:44.06 | Darkerr | Ok, Elastix web interface little confused me with Creating SIP account under Extensions menu, but its ok... I'd like to limit maximum Extensions that user can made in Elastix web interface... Since elastix doesnt have this feature I am looking for any possible solution |
| 09:45.19 | ChannelZ | that's an Elastix question |
| 09:46.14 | ChannelZ | There are no configuration limits within asterisk |
| 09:46.36 | skirmisha | ok here are my observations, asterisk match first peer found in config that match against IP and port |
| 09:47.01 | Darkerr | No limits in asterisk, ok that's what I needed to know. Thank you. |
| 09:50.23 | ChannelZ | skirmisha, yes. Two physically different devices cannot have the same IP AND port. |
| 09:50.29 | ChannelZ | They can have the same IP and DIFFERENT ports |
| 09:51.30 | eirirs | or different IP and same port |
| 09:51.40 | ChannelZ | goes without saying |
| 09:51.49 | eirirs | not for everyone hehe |
| 09:52.35 | ChannelZ | Consider the port number as a child of the IP. They go together, and the IP is matched first. |
| 09:52.57 | skirmisha | god damn |
| 09:53.07 | *** join/#asterisk ayrjola (~ayrjola@89.18.237.137) |
| 09:55.39 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-130-91.dynamic.qsc.de) |
| 09:55.49 | ChannelZ | ? |
| 09:59.03 | skirmisha | thanks guys |
| 09:59.13 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:59.20 | skirmisha | looks like i need to have this in mind when configure my setup |
| 09:59.43 | skirmisha | one question, can i match based on user agent? |
| 10:00.12 | ChannelZ | no. That would be terrible anyway |
| 10:03.41 | AnonGirl | wut |
| 10:09.14 | ChannelZ | See http://burner.com/asterisk-primer/configuring-sip/ for a summary explanation of sip peer matching (with chan_sip, doesn't apply to PJSIP) |
| 10:11.41 | ChannelZ | goes to bed |
| 10:21.38 | *** join/#asterisk pawiecki (~pawiecki@router.dir.pl) |
| 10:22.48 | pawiecki | i have a problem with pickupgroups. I have like 46 of them, and some of them are not working. What can be a problem? |
| 10:29.08 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 10:42.48 | *** join/#asterisk catphish (~catphish@unaffiliated/catphish) |
| 10:43.05 | catphish | does asterisk still need dahdi on systems where no hardware is present? |
| 10:43.36 | catphish | or has all the software timing been moved into asterisk? |
| 10:53.41 | *** part/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 10:55.00 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 10:57.04 | *** join/#asterisk lvlinux (~ruel@unaffiliated/lvlinux) |
| 11:00.22 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 11:00.33 | *** join/#asterisk wdoekes (~walter@wjd.osso.nl) |
| 11:04.04 | *** join/#asterisk jgornick (jgornick@2600:3c00::f03c:91ff:fedf:72b4) |
| 11:04.05 | *** join/#asterisk sommarnatt (~sommarnat@kungsbacka.oderland.com) |
| 11:04.22 | *** join/#asterisk dustinm (~dustinm@105.ip-167-114-152.net) |
| 11:07.02 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 11:14.20 | *** join/#asterisk mirela666 (~mirko@212.200.146.247) |
| 11:14.21 | *** part/#asterisk mirela666 (~mirko@212.200.146.247) |
| 11:16.11 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 11:27.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:40.58 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 12:01.09 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 12:21.14 | catphish | seems its still needed for meetme |
| 12:22.36 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 12:38.47 | *** join/#asterisk crised (~crised@186.67.181.203) |
| 13:00.24 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:16.18 | *** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com) |
| 13:16.30 | *** join/#asterisk [sID] (sid@ip-178-216-203-50.e24cloud.com) |
| 13:20.25 | *** join/#asterisk generalhan (~tester@about/windows/staff/generalhan) |
| 13:21.08 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:28.58 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-46-222.user.veloxzone.com.br) |
| 13:29.45 | *** join/#asterisk wonderworld (~ww@dslb-088-078-147-078.088.078.pools.vodafone-ip.de) |
| 13:30.03 | pawiecki | i've got a noob question: how can i forward port 5060 from remote host, to register softphone on my pc. I don't need voice transmission, just registering on server. |
| 13:30.35 | [TK]D-Fender | What do you mean FROM a remote host? |
| 13:31.45 | [sID] | pawiecki: iptables? |
| 13:39.00 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 13:40.23 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:50.15 | pawiecki | i have softphone on my pc, and server in remote location, i can access it via ssh. I want to register softphone to the server. Maybe i wrote it wrong. |
| 13:52.00 | [TK]D-Fender | pawiecki: You should have SIP & RTP open on the server |
| 13:52.02 | catphish | i don't think you can forward udp over ssh, you could get asterisk to listen on tcp |
| 13:52.36 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:57.01 | [TK]D-Fender | pawiecki: if the only think you have open on your server is SSH : https://www.google.ca/#q=vpn+over+ssh |
| 13:58.00 | catphish | oh yeah, ppp over ssh is a good option if your client can support that |
| 14:00.56 | pawiecki | [TK]D-Fender: catphish: thanks, checking it out. |
| 14:09.59 | *** join/#asterisk tristan-mei| (tristan@unaffiliated/tristan-mei) |
| 14:10.13 | *** join/#asterisk zyxel (~ronen@185.28.152.22) |
| 14:12.04 | *** join/#asterisk mirela666 (~mirko@212.200.146.242) |
| 14:12.12 | *** part/#asterisk mirela666 (~mirko@212.200.146.242) |
| 14:14.12 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 14:19.55 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:22.30 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:40.23 | *** join/#asterisk robmal (r@wporzo.pl) |
| 14:44.33 | *** join/#asterisk Katty (sid62315@gateway/web/irccloud.com/x-fvxyhynmwgwjyyrt) |
| 14:51.23 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 14:55.35 | *** join/#asterisk bmurt (~brendan@64-121-3-32.c3-0.upd-ubr2.trpr-upd.pa.cable.rcn.com) |
| 14:56.50 | *** join/#asterisk tristan-mei (tristan@unaffiliated/tristan-mei) |
| 15:01.46 | *** join/#asterisk bmurt (~brendan@64-121-3-32.c3-0.upd-ubr2.trpr-upd.pa.cable.rcn.com) |
| 15:06.17 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-rjqcjzuzqqrortow) |
| 15:06.30 | *** join/#asterisk qakhan (~qakhan@50-204-254-12-static.hfc.comcastbusiness.net) |
| 15:07.02 | qakhan | hi all, is there any way we can create 50 exts using a script |
| 15:07.42 | [TK]D-Fender | No, the limit is 49 |
| 15:08.16 | qakhan | [TK]D-Fender are you joking with me :) |
| 15:08.27 | [TK]D-Fender | You'll have to wait for Asterisk 15 since 14 is already in lock-down..... |
| 15:09.25 | *** join/#asterisk JCONeill (~scotz@host-92-13-111-132.as43234.net) |
| 15:09.33 | qakhan | ok what is that script |
| 15:09.59 | *** join/#asterisk bmurt (~brendan@64-121-3-32.c3-0.upd-ubr2.trpr-upd.pa.cable.rcn.com) |
| 15:10.22 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:10.59 | [TK]D-Fender | What script? Configuring Asterisk is YOUR job. |
| 15:11.23 | [TK]D-Fender | qakhan: How many years have you been using asterisk at this point? about 4 years based on a quick search... |
| 15:11.51 | [TK]D-Fender | qakhan: Asterisk is configured by a bunch of dumb text files. You cen generate those however you feel like and Asterisk won't care about the "how". |
| 15:12.03 | MaliutaLap | [TK]D-Fender: you know _the_ script ... it's called vim ;) |
| 15:12.33 | [TK]D-Fender | MaliutaLap: http://xkcd.com/378/ |
| 15:12.44 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 15:13.32 | MaliutaLap | qakhan: the issue with software as complex as asterisk is that there are so many different ways of doing things, and so many complex options that no script or GUI can hope to be actually useful to configure the whole thing |
| 15:14.12 | [TK]D-Fender | MaliutaLap: You mean like FreePBX? :) |
| 15:14.30 | qakhan | [TK]D-Fender so are you telling me we can have only 49 exts in an asterisk? |
| 15:14.35 | [TK]D-Fender | MaliutaLap: And yes you can script whatever you feel like and it'll be as useful as the peron writing it. |
| 15:14.41 | MaliutaLap | [TK]D-Fender: I know, I know ... but http://ars.userfriendly.org/cartoons/?id=20001111&mode=indexed |
| 15:15.03 | MaliutaLap | [TK]D-Fender: peron? or peon? ;) |
| 15:15.07 | [TK]D-Fender | [10:11][TK]D-Fenderqakhan: Asterisk is configured by a bunch of dumb text files. You can generate those however you feel like and Asterisk won't care about the "how". <------------ |
| 15:15.25 | [TK]D-Fender | qakhan: Generate config files. Reload. DONE. |
| 15:16.03 | MaliutaLap | well generate config files. reload, test, tinker, rinse, repeat ;) |
| 15:16.28 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:16.29 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:17.32 | qakhan | [TK]D-Fender and MaliutaLap i got you. but [TK]D-Fender said there is limit of 49 exts |
| 15:17.55 | [TK]D-Fender | facepalms |
| 15:17.59 | MaliutaLap | pretty sure that's not something out Fender would say |
| 15:18.14 | MaliutaLap | [TK]D-Fender: can I borrow you palm? |
| 15:18.32 | MaliutaLap | pilot if you got one ;) |
| 15:18.37 | [TK]D-Fender | MaliutaLap: I don't like where this is going... |
| 15:19.31 | MaliutaLap | [TK]D-Fender: it's just for my face - it's that dire that my palm->face alone isn't enough |
| 15:20.12 | MaliutaLap | [TK]D-Fender: if I could borrow the whole hand then a third hand typing might actually help |
| 15:21.16 | MaliutaLap | or I could use play it in poker - My hand is [TK]D-Fender's, the highest hand possible -> I win ... QED ;) |
| 15:21.42 | [TK]D-Fender | MaliutaLap: Face-high specifically... |
| 15:22.00 | MaliutaLap | true, true |
| 15:22.17 | MaliutaLap | [TK]D-Fender: why? what was your grotty mind thinking? ;P |
| 15:23.02 | MaliutaLap | considers the Zaphod Beeblebrox third arm. It might help me play guitar and keyboard betterer |
| 15:24.12 | MaliutaLap | tries for bed and sleep again - 40 hours is enough and the sillies are kicking in hard |
| 15:24.38 | *** join/#asterisk areski (~areski@229.112.135.37.dynamic.jazztel.es) |
| 15:24.48 | [TK]D-Fender | MaliutaLap: https://www.youtube.com/watch?v=iQ8ml7eENuI |
| 15:28.17 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 15:33.33 | *** join/#asterisk kharwell (kharwell@nat/digium/x-ftbzaccyecoxqjbr) |
| 15:38.30 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-znxonrpezneypozl) |
| 15:50.41 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 15:52.33 | *** part/#asterisk kharwell (kharwell@nat/digium/x-ftbzaccyecoxqjbr) |
| 15:58.49 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-yssfsotybbmdnkkt) |
| 16:01.24 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 16:02.12 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:03.21 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 16:03.44 | *** join/#asterisk FreezingAlt (~FreezingC@135.0.41.14) |
| 16:09.08 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:82fa:5bff:fe0a:dfef) |
| 16:15.40 | *** join/#asterisk mjordan (mjordan@nat/digium/x-ayarliksmxptfvrt) |
| 16:15.40 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:19.00 | *** join/#asterisk ipalmer (~IceChat77@host81-133-140-79.in-addr.btopenworld.com) |
| 16:19.59 | ipalmer | hello all, I'm trying to merge 2 conferences together, the conference names are dynamic so can be different every time, I know I need to use th originate application but am struggling to understand how to do it, any ideas anyone |
| 16:21.07 | [TK]D-Fender | ipalmer: Originate a Local channel that connects to the first side .... and on answer direct it to dialplan to connect to the other side |
| 16:21.07 | WIMPy | Maybe you should make the task a little harder and move the participants of the one conference in to the other. |
| 16:22.07 | ipalmer | D-Fender: thanks, that would allow all users in conference one to hear all users in conference 2? |
| 16:22.35 | [TK]D-Fender | ipalmer: It allows whatever you allow that channel to hear/say |
| 16:22.50 | WIMPy | I doubt it would be a good user experience to connect two conferences together. Unless one of them was muted. |
| 16:24.01 | ipalmer | D-Fender: ok, is this possible using AMI? |
| 16:24.26 | ipalmer | WIMPy: possibly not but it's what has been requested of me from the powers that be |
| 16:25.19 | WIMPy | Off course you can originate via AMI. |
| 16:25.38 | WIMPy | But If you're already on AMI, you could also transfer the users. |
| 16:26.15 | ipalmer | WIMPy: I was toying with doing that but they want to be able to put them back into their original conferences afterwards |
| 16:34.41 | DivideBy0 | in the ARI, I can start snooping, but to stop snooping, I should destroy the snooping channel, right? |
| 16:34.55 | file | yes |
| 16:35.05 | DivideBy0 | thanks |
| 16:35.58 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 16:49.27 | pabelanger | snoop on to them, as they snoop on to us |
| 16:52.59 | file | DivideBy0, how are you finding the mechanism for snooping? |
| 16:57.45 | *** join/#asterisk areski (~areski@81.Red-83-53-30.dynamicIP.rima-tde.net) |
| 16:59.55 | *** join/#asterisk stkoch (58d12049@gateway/web/freenode/ip.88.209.32.73) |
| 16:59.57 | *** join/#asterisk wanda__ (~xivo@modemcable094.94-70-69.static.videotron.ca) |
| 17:00.13 | wanda__ | hi' IM looking for asterisk-fr? |
| 17:00.33 | [TK]D-Fender | Are you? |
| 17:00.36 | WIMPy | Start it. |
| 17:00.45 | [TK]D-Fender | You seem to be about 3 charaters short... |
| 17:01.09 | wanda__ | my profile http://valerie.dagrain.numerimoire.net/?page_id=25 |
| 17:02.10 | wanda__ | and currently here http://www.asterisk-france.org/threads/3430-Blog-XiVO-des-nouvelles-sur-des-tutos-et-contributeurs |
| 17:02.25 | wanda__ | I need some advices about a presentation I prepare about XiVO |
| 17:02.43 | wanda__ | I need advices from students in telecommunication :D |
| 17:03.10 | wanda__ | and I would like to contact someone from Asterisk to thanks the community |
| 17:03.13 | wanda__ | on the slide :D |
| 17:03.15 | *** part/#asterisk catphish (~catphish@unaffiliated/catphish) |
| 17:03.32 | wanda__ | that's it *____* |
| 17:03.58 | DivideBy0 | file: still learning. I'm trying to snoop on a channel I just created before it enters my stasis app and haven't succeeded yet. I will report back more once I use it more |
| 17:04.22 | wanda__ | So I listen your advices :) |
| 17:05.28 | *** part/#asterisk ipalmer (~IceChat77@host81-133-140-79.in-addr.btopenworld.com) |
| 17:10.21 | stkoch | hi, I have a problem with asterisk 1.8.30.0 and voip calls with high packet loss. Sometimes a call works for 30 seconds, somtimes 120 seconds. After that the speech has dropouts and the call is aborted within 10 seconds. If there are dropouts an WARNING message comes with it: WARNING[2259]: channel.c:1513 __ast_queue_frame: Exceptionally long voice queue length queuing to TAPI/1 |
| 17:10.25 | stkoch | How to avoid call cancelation and let the call open to wait for lesser packet loss again? |
| 17:10.57 | stkoch | I have tried a bit with jitter settings, it doesn't help. |
| 17:11.31 | [TK]D-Fender | stkoch: use a codec that survives PL better like G.729 or iLBC, and adjust your rtp timeout. |
| 17:11.34 | [TK]D-Fender | rtp.conf <- |
| 17:11.38 | WIMPy | That means that TAPI/1 doesn't process the voice frames. Whatever kind of channel that is. |
| 17:12.16 | stkoch | WIMPy: it's an lantiq soc in a router with openwrt |
| 17:12.40 | stkoch | WIMPy: fxs port |
| 17:14.09 | WIMPy | You probably need to tune your jitterbuffer. |
| 17:18.04 | stkoch | [TK]D-Fender: I have set rtptimeout=120; but the call terminates after 10 seconds |
| 17:19.20 | stkoch | WIMPy: jbenable=yes; jbforce=yes; jbimpl=fixed; |
| 17:21.47 | [TK]D-Fender | I'm not even sure what TAPI technically uses for voice... |
| 17:22.50 | stkoch | [TK]D-Fender: https://dev.openwrt.org/browser/packages/net/asterisk-1.8.x/src/configs/lantiq.conf.sample?rev=32047 |
| 17:24.19 | [TK]D-Fender | What hardware? |
| 17:25.16 | *** join/#asterisk deranged (Bit@lolnerd.net) |
| 17:26.01 | stkoch | [TK]D-Fender: http://wiki.openwrt.org/toh/astoria/arv752dpw22 |
| 17:27.28 | stkoch | [TK]D-Fender: lantiq danube |
| 17:28.02 | [TK]D-Fender | stkoch: That's running * on OpenWRT itself right? |
| 17:28.34 | [TK]D-Fender | stkoch: So the timeout is between the TAPI driver layer and * on the same device? |
| 17:30.48 | stkoch | [TK]D-Fender: on this router openwrt 14.09 is running. also on android softphone (connected to sipgate) there are dropouts after the same time but the call doesn't abort |
| 17:31.12 | stkoch | [TK]D-Fender: so could asterisk give the TAPI device dummy data as long the packet dropouts there? |
| 17:31.31 | WIMPy | It's that TAPI device that can't handle jitter. |
| 17:32.05 | WIMPy | But it's not the missing parts but the backlog thereafter that causes the problem. Well, or maybe not, but it's what causes that message. |
| 17:32.30 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 17:39.50 | [TK]D-Fender | stkoch: It shouldn't jitter being all internal.. That'd be a pretty serious CPU issue |
| 17:41.01 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 17:51.04 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:55.43 | *** part/#asterisk CeBe1 (~CeBe@port-92-200-130-91.dynamic.qsc.de) |
| 17:56.37 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 17:58.46 | *** join/#asterisk rexwin (~rexwin@106.208.7.38) |
| 18:04.59 | rexwin | when I run commands in cli it doesnot log in messages log, what needed to be done for logging output of asterisk cli output? |
| 18:05.20 | *** join/#asterisk cyford (~allen@76.122.73.37) |
| 18:06.33 | stkoch | WIMPy, [TK]D-Fender: it is possible to drop the backlog that causes the problem by asterisk? Have you seen the jitter settings in the lantiq.conf.sample link? Is it possible that asterisk does protect the lantiq fxs from jitter with silence sounds? |
| 18:07.57 | [TK]D-Fender | * doesn't do CNG |
| 18:08.10 | [TK]D-Fender | It will buffer things a bit.. but no real concealment, iIRC |
| 18:08.22 | [TK]D-Fender | rexwin: logger.conf <- |
| 18:08.49 | [TK]D-Fender | rexwin: that will log OUTPUT. Not sure you can log CLI COMMANDS you actually issue |
| 18:10.12 | *** join/#asterisk ZerOlegend (~ZerOlegen@98-125-41-229.dyn.centurytel.net) |
| 18:16.58 | crised | Has anyone used skype from a SIP phone_ |
| 18:17.00 | crised | ? |
| 18:17.35 | stkoch | [TK]D-Fender: but the beep beep beep sound comes right after call termination... |
| 18:19.20 | [TK]D-Fender | crised: Probably not from a phone.... |
| 18:19.30 | [TK]D-Fender | crised: Better odds from a PBX |
| 18:19.50 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 18:21.36 | cyford | in the dialplan how do i get the name of owner of the extenstion |
| 18:22.38 | mjordan | that concept doesn't exist. |
| 18:23.10 | mjordan | when you say "owner", what do you mean? |
| 18:23.23 | [TK]D-Fender | Extension = dialplan. Nobody owns it. |
| 18:23.57 | [TK]D-Fender | maybe "ls -la /etc/asterisk/extensions.conf" :) |
| 18:24.05 | [TK]D-Fender | There's an owner :) |
| 18:24.26 | mjordan | heh |
| 18:25.04 | [TK]D-Fender | considers looking up the regex to pull that back with a SHELL() |
| 18:25.05 | cyford | allen = extenstion 200 |
| 18:25.25 | [TK]D-Fender | You're using those dangeroulsy vague terms again... |
| 18:25.41 | [TK]D-Fender | dangerously* |
| 18:25.59 | eirirs | vague talk is just a trick to conceal the incompentence. :) |
| 18:26.16 | cyford | i can run a system command to get it from sip_additional.conf |
| 18:27.41 | [TK]D-Fender | eirirs: It highlights it for me. Showing no clue doesn't inspire to expect any more to come :) |
| 18:27.56 | [TK]D-Fender | cyford: You're in the WRONG CHANNEL for this...... |
| 18:28.16 | crised | [TK]D-Fender: ok, Skype supports more phone numbers than voip.ms in my country, like local toll free numbers, |
| 18:28.18 | cyford | no i was askin if there was a way in asterisk, |
| 18:28.30 | [TK]D-Fender | CyThose concepts are FREEPBX STRUCTURES |
| 18:28.41 | cyford | but after hearing there isnt i decided todo it in cli |
| 18:28.46 | [TK]D-Fender | This is no anything inherently "Asterisk" |
| 18:29.14 | [TK]D-Fender | cyford: "database show" <----- it's in there |
| 18:29.50 | [TK]D-Fender | crised: So go for it |
| 18:30.28 | crised | [TK]D-Fender: setup asterix... it takes work |
| 18:30.33 | cyford | i thought it may be as easy as ${CALLERID(name)} i just didnt know the variable,... and when i dont know and cant find info , i ask the pro's |
| 18:30.38 | crised | I have a freebsd box where I could do it... |
| 18:31.50 | [TK]D-Fender | crised: I didn't say you needed Asterisk for this |
| 18:31.51 | WIMPy | cyford: We can't help you without knowing what you're actually looking for. But as [TK]D-Fender already said, it might not be an Asterisk question at all. |
| 18:32.16 | [TK]D-Fender | [13:29][TK]D-Fendercyford: "database show" <----- it's in there |
| 18:32.19 | crised | [TK]D-Fender: "not from a phone" .. |
| 18:32.41 | [TK]D-Fender | crised: WE don't. Doesn't mean you can't. |
| 18:32.47 | crised | [TK]D-Fender: oh you mean, just use skype... yeha it's an option, more uncomfortable |
| 18:32.53 | [TK]D-Fender | crised: You're in #asterisk. That's what we use. |
| 18:32.55 | cyford | it was already answered, i was looking for the name that is tied to an extenstion |
| 18:32.59 | crised | :) |
| 18:33.16 | [TK]D-Fender | crised: Witha Skype business acount you can access them via SIP. |
| 18:33.26 | [TK]D-Fender | cyford: And it's in there |
| 18:33.30 | crised | [TK]D-Fender: are they $$? |
| 18:33.31 | WIMPy | cyford: Such a thing doesn't exist. |
| 18:33.40 | cyford | in a dialplan instead of showing ${exten} i want to show the name of it... |
| 18:34.03 | cyford | yes i heard that, then i said i will do it through cli |
| 18:34.12 | cyford | thanks |
| 18:34.15 | [TK]D-Fender | cyfordin a dialplan instead of showing ${exten} i want to show the name of it... <- This is a very poor way to try to express what you're looking for. |
| 18:34.25 | [TK]D-Fender | You can do it DIRECT in the dialplan... |
| 18:34.41 | WIMPy | Extensions don't have names. The Extension is the "name". |
| 18:34.41 | cyford | how? |
| 18:34.51 | [TK]D-Fender | [13:32][TK]D-Fender[13:29][TK]D-Fendercyford: "database show" <----- it's in there |
| 18:35.32 | cyford | ok |
| 18:35.35 | cyford | thanks |
| 18:36.07 | [TK]D-Fender | prepares to re-paste it again for quadruple-nesting.... |
| 18:36.42 | cyford | i meant callerid of the exten WIMPy |
| 18:36.52 | cyford | callerid=Terri Jones <2020> |
| 18:37.02 | WIMPy | Extensions don't have caller IDs, either. |
| 18:37.21 | WIMPy | Extensions are destinations. Nothing else. |
| 18:37.32 | [TK]D-Fender | cyford: I hope you're not wasting time continuing to look at raw device config files.... |
| 18:37.53 | WIMPy | Maybe they are something else in FreePBX, but that's not what it's about here. |
| 18:38.25 | cyford | so u dont program a phone with an extenstion |
| 18:38.38 | WIMPy | What? |
| 18:38.50 | cyford | so u dont program a phone with an extenstion |
| 18:38.55 | stkoch | [TK]D-Fender: it is possible it this error happens that asterisk restarts the channel? |
| 18:39.12 | [TK]D-Fender | xtI really have no experience with TAPI. |
| 18:39.18 | [TK]D-Fender | stkoch: I really have no experience with TAPI. |
| 18:39.26 | WIMPy | Phones have users, not extensions. |
| 18:39.33 | WIMPy | Extensions can call phones. |
| 18:39.47 | [TK]D-Fender | More accurately, phones have device configs. |
| 18:40.11 | mjordan | stkoch: Not trying to be discouraging, but you're using a third party channel driver whose site doesn't appear to have had an update pushed to it in the better part of three years. Your mileage is going to vary a lot. |
| 18:40.20 | cyford | in the device config is a list of exenstions right lol? |
| 18:40.24 | WIMPy | was talking about phones, not Asterisk. But then we don;t really know what it's about. |
| 18:40.43 | cyford | atleast all the phones i programed say it |
| 18:40.55 | *** join/#asterisk bkruse (~Adium@64.89.97.113) |
| 18:41.12 | WIMPy | No. Extensions are in extensions.conf and NOT in device settings. |
| 18:42.07 | cyford | ok, i always calleded them dialplan exten. vs user extens.. but ill change it sorry |
| 18:42.27 | cyford | just confusing going across platforms lol |
| 18:43.21 | stkoch | so there is no solution that asterisk already provide? or the internet connection must good enough to have only little jitters... |
| 18:44.17 | WIMPy | stkoch: It's really hard to say waht's going on there. I'd try to tune the JB settings. Especially a max size. |
| 18:45.18 | cyford | [TK]D-Fender i can use the db lookup methode for sip and iax and go the cli route for pjsips |
| 18:45.38 | [TK]D-Fender | cyford: Should work for ALL |
| 18:45.48 | cyford | ok thanks |
| 18:46.15 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 18:50.27 | drmessano | "Asterisk - A big ball of ringy-dingy telly-welly..... stuff" |
| 18:50.51 | drmessano | If Digium marketing uses that, I want a some free something |
| 18:53.40 | drmessano | "BYOLOL - What do you mean you can't set up my e-mail on my Nokia 3360?" |
| 18:56.24 | stkoch | WIMPy, [TK]D-Fender: csipsimple on android shows packet loss of 53.4 % ! |
| 18:56.55 | WIMPy | Well, at that values, I wouldn;t waste any time on trying to make it work. |
| 18:59.29 | file | raises eyebrow |
| 18:59.46 | [TK]D-Fender | Get a real system to run * on and a real FXO interface to match |
| 18:59.55 | [TK]D-Fender | Sounds like this thing is jsut loaded to bits... |
| 19:00.12 | WIMPy | Or a working internet connection? |
| 19:00.20 | drmessano | How is my 12 yr old Linksys running OpenWRT not a real system? |
| 19:00.25 | drmessano | Screw you guys, im going home |
| 19:02.56 | drmessano | Someone should make an adapter with a male<-->female 5.5mm/2.1mm coax plug that T's off to a pigtail with a regular in line and a mini USB on it for an RPi.. Then You could sell a router "upgrade" kit |
| 19:03.08 | drmessano | regulator* |
| 19:03.31 | WIMPy | What? |
| 19:03.31 | drmessano | "Raspberry Pi Co-Processor" |
| 19:04.19 | WIMPy | Like the first ARMs were used as coprocessors to 6502s? |
| 19:04.22 | drmessano | WIMPy, a Splitter that splits the DC going into a 12v router with a 5.5/2.1 plug, + regulator and USB to power an RPI |
| 19:04.43 | WIMPy | Ah |
| 19:04.54 | stkoch | WIMPy: I'm behind a proxy server with max. of 100 Mbit/s |
| 19:05.57 | WIMPy | 100MBit should be good for 1000 calls. But you don't use a proxy for UDP, do you? |
| 19:08.07 | *** join/#asterisk Alex_Bkash (7c06ecf5@gateway/web/freenode/ip.124.6.236.245) |
| 19:09.28 | [TK]D-Fender | ~savemoney |
| 19:09.28 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 19:09.43 | cyford | over 2000 calls for 100mbs full duplex |
| 19:09.45 | *** join/#asterisk Navion (~Navion@rivendell/users/navion) |
| 19:10.36 | WIMPy | You can get even more. Or maybe a little less. It all depends on your configuration. |
| 19:12.14 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:15.16 | *** join/#asterisk blinky42 (~sb@c-76-124-208-67.hsd1.nj.comcast.net) |
| 19:22.05 | stkoch | WIMPy: this connection is shared with hundred others with only one public IP for all |
| 19:25.14 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 19:27.09 | *** join/#asterisk FreezingAlt (~FreezingC@135.0.41.14) |
| 19:28.47 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 19:33.09 | *** join/#asterisk PaybackTony (~Adium@2601:7:700:7fa:e822:27f2:16a:e87e) |
| 19:42.26 | stkoch | Thanks WIMPy [TK]D-Fender mjordan ... |
| 19:52.22 | *** join/#asterisk bipolar (~bipolar@offsite.guru) |
| 19:56.40 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 20:03.51 | *** part/#asterisk stkoch (58d12049@gateway/web/freenode/ip.88.209.32.73) |
| 20:03.51 | *** join/#asterisk FreezingDroid (~FreezingC@135.0.41.14) |
| 20:04.37 | *** join/#asterisk Sprocks (~Sprocks@BMTNON3746W-LP140-04-845509158.dsl.bell.ca) |
| 20:04.39 | *** join/#asterisk areski (~areski@80.174.128.98.dyn.user.ono.com) |
| 20:11.19 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 20:25.19 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 20:30.03 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:33.10 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 20:41.45 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 20:45.21 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:52.48 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 20:53.10 | DivideBy0 | I had an ARI originate hit my timeout and then hungup with a reason 16/normal, shouldn't it hangup with a reason of 18/expired or 102/timeout? |
| 20:53.59 | WIMPy | Where/what timed out? |
| 20:54.28 | DivideBy0 | the originated channel. you can give it a timeout when you originate it in the ARI |
| 20:55.16 | WIMPy | As in it was ringing but not answered in the given time? |
| 20:55.25 | DivideBy0 | yup |
| 20:56.06 | WIMPy | That would be more like 19. |
| 20:56.34 | DivideBy0 | agreed, 19 would be even better |
| 20:56.45 | *** join/#asterisk overyander (~Jeff@12.49.160.131) |
| 20:57.40 | DivideBy0 | I spent more time than I'd like to admit tracking it down. A better Hangup Cause would have made me look in the right place sooner |
| 20:58.47 | WIMPy | You get used to that. |
| 20:59.05 | overyander | Can you use t.38 udptl over a tcp session or do i have to use rtp for t.38 if i'm using tcp instead of udp? |
| 21:02.04 | DivideBy0 | WIMPy: thanks. that's very reassuring |
| 21:02.16 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:09.18 | *** join/#asterisk rexwin (~rexwin@106.208.7.38) |
| 21:09.27 | mjordan | DivideBy0: interestingly enough, the core dial API looks like it attempts to set the hangup cause on the channel to 19. I'm not sure why that wouldn't have propagated up, as nothing else appears like it tries to set a different hangup cause on the channel on that path (although I did just do a cursory look). |
| 21:09.37 | mjordan | DivideBy0: what was the cause returned on the ARI event? |
| 21:09.40 | rexwin | I am trying to play an Auto-Attendant. but nothing happens |
| 21:09.43 | rexwin | http://pastebin.com/J9UqRafu |
| 21:10.01 | mjordan | "[Feb  9 07:06:12] WARNING[2373][C-00000000]: file.c:701 ast_openstream_full: File press-1 does not exist in any format" |
| 21:10.06 | mjordan | What do you think that means? :-) |
| 21:10.44 | rexwin | some wav file is missing, i suppose |
| 21:10.45 | WIMPy | I'd prefer to play with a rea attendant rather than an automated one. |
| 21:11.28 | rexwin | WIMPy;-) |
| 21:11.32 | mjordan | rexwin: and where are the sound files located? |
| 21:12.36 | file | overyander, UDPTL is UDP and is a separate protocol from RTP. |
| 21:13.27 | DivideBy0 | mjordan: Received event: ChannelDestroyed cause_txt: Normal |
| 21:13.48 | DivideBy0 | mjordan: I can create a simple reproducable test later tonight |
| 21:13.58 | rexwin | google says /var/lib/asterisk/sounds/e |
| 21:15.19 | mjordan | DivideBy0: that'd be handy. If you do, please attach the test case/logs to an issue at issues.asterisk.org, as that sounds like a bug to me |
| 21:15.21 | DivideBy0 | mjordan: I looked in at my debug log, and there's no reason set. I only saw it on the ARI event |
| 21:17.26 | DivideBy0 | ok. I'll post everything tonight |
| 21:20.38 | Alex_Bkash | Not a room question. but any one can tell me how can i add javascript+ ajax effects on a existing SVG file?? |
| 21:21.45 | robmal | Like svgjs.com ? |
| 21:23.28 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 21:24.39 | Alex_Bkash | ok let me check thanks |
| 21:25.05 | overyander | file, i'm having some issues with reliability. i was thinking that using TCP between my server and sip provider instead of UDP would increase stability, but I didn't know if I should use UDPTL or RTP for the t.38. UDPTL having 'UDP' in the name kinda implies that it's UDP and not TCP, but I wanted to confirm. |
| 21:25.07 | DivideBy0 | mjordan: I tried to create a jira account, but nothing came through, and I cant reset my password. Do I need to be approved? |
| 21:31.50 | rexwin | thanks mjordan. it worked after I put a vm file |
| 21:31.53 | wdoekes | overyander: all media is sent over udp; so yes, both rtp and udptl use udp. |
| 21:31.55 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 21:33.10 | overyander | so enabling tcp only applies to the sip packets and not the media packets? |
| 21:33.15 | wdoekes | correct |
| 21:33.27 | overyander | oh, didn't know that. thanks |
| 21:33.28 | wdoekes | if you want media over tcp, you'll need a tunnel (e.g. openvpn) |
| 21:33.36 | wdoekes | in rare cases, that can help |
| 21:40.08 | overyander | the t38pt_usertpsource option, does that set t.38 rtp to true? i just heard back from my provider, they only support rtp and not udptl. looking in my sippeers table (realtime sip) the only option related to t38 and rtp is t38pt_usertpsource. |
| 21:40.29 | file | T.38 using RTP is not supported |
| 21:41.30 | overyander | even with the FFA module? |
| 21:41.45 | file | correct |
| 21:42.05 | overyander | what is the t38pt_usertpsource option for? |
| 21:42.40 | file | Use the source IP address of RTP as the destination IP address for UDPTL packets |
| 21:42.41 | file | <PROTECTED> |
| 21:42.41 | file | <PROTECTED> |
| 21:42.41 | file | <PROTECTED> |
| 21:43.19 | file | shamelessly copied from sip.conf.sample |
| 21:46.03 | overyander | thanks file |
| 21:47.13 | overyander | so, if my provider only supports RTP and not UDPTL and Asterisk only support UDPTL then why is it that some of the faxes are coming through using t.38? about 1 out of 20 are successful, the others time out. |
| 21:48.14 | file | no idea - they may be incorrect |
| 21:48.41 | file | or it's using standard RTP and not T.38 and by chance it's working |
| 21:48.48 | file | dunno! |
| 21:53.25 | wdoekes | overyander: 'fax set debug on' and check the fax log configured in logger.conf |
| 21:53.52 | file | a SIP trace would also show what it is using |
| 21:58.10 | wdoekes | or that. but I bet the faxes come through over rtp, and the logs may help to figure out the reliability problems |
| 22:00.16 | overyander | the packet captures of the failed faxes and the one successful fax all show that it's UDPTL packets. I'm not sure what my provider is thinking. |
| 22:02.06 | *** join/#asterisk marquist (~mcooper@browndeer.switchvox.com) |
| 22:55.36 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-incmwdjxbkthbifx) |
| 22:56.31 | *** join/#asterisk moy (sid47040@gateway/web/irccloud.com/x-hhhrzxkzyiucuqwk) |
| 23:00.32 | *** join/#asterisk eschmidbauer (~chatzilla@unaffiliated/eschmidbauer) |
| 23:01.08 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 23:02.00 | *** join/#asterisk moy (sid47040@gateway/web/irccloud.com/x-zhgbdomrqjnmllar) |
| 23:05.36 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 23:17.56 | *** part/#asterisk mjordan (mjordan@nat/digium/x-ayarliksmxptfvrt) |
| 23:32.54 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 23:33.52 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:41.10 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-vkgclhcpxifcnoxz) |
| 23:44.05 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 23:49.19 | *** join/#asterisk Sprocks (~Sprocks@BMTNON3746W-LP130-03-1242451596.dsl.bell.ca) |