| 00:00.23 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:10.12 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 01:17.24 | *** join/#asterisk fullstop (~fullstop@107.170.53.155) |
| 01:52.07 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:02.55 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 02:17.17 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 03:32.20 | *** join/#asterisk [Outcast] (~outcast@gw-outbound.broadinstitute.org) |
| 03:56.50 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 04:02.33 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 04:10.57 | *** join/#asterisk hos7ein (~chatzilla@94.101.189.1) |
| 04:20.38 | *** join/#asterisk [Outcast] (~outcast@gw-outbound.broadinstitute.org) |
| 04:39.22 | *** join/#asterisk [Outcast] (~outcast@gw-outbound.broadinstitute.org) |
| 05:02.51 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:1dd0:221:6aff:feb8:e0b2) |
| 05:17.35 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 05:56.59 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 06:11.06 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 06:16.02 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 06:16.24 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 06:25.57 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 06:28.50 | *** join/#asterisk [Outcast] (~outcast@64.206.121.41) |
| 06:43.42 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 06:47.10 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 06:47.54 | *** join/#asterisk CeBe (~CeBe@brsg-d9baf213.pool.mediaways.net) |
| 06:56.55 | *** join/#asterisk ruben23 (~OpenDIAL@112.198.77.114) |
| 06:57.03 | ruben23 | hi there guys |
| 07:03.37 | *** join/#asterisk jhlavacek (~jirka@ata35-1-78-208-220-3.fbx.proxad.net) |
| 07:06.42 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:4cb4:fa0b:968d:361d) |
| 07:07.18 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 07:08.36 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 07:09.08 | Zogot | morning |
| 07:12.34 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:12.46 | MaliutaLap | wrong TZ - it's 17:13 :P |
| 07:13.26 | Zogot | http://www.total-knowledge.com/~ilya/mips/ugt.html :p |
| 07:14.23 | MaliutaLap | pfft |
| 07:14.34 | MaliutaLap | join the one true TZ - GMT+10 :P |
| 07:15.37 | ChannelZ | The rotation of the earth sort of doesn't work that way. |
| 07:17.19 | MaliutaLap | It does down here in .au ;) |
| 07:23.52 | ruben23 | <PROTECTED> |
| 07:24.00 | ruben23 | im gusing dahdi channels FXS |
| 07:25.25 | *** join/#asterisk cervajs2 (~cervenka@178.148.broadband4.iol.cz) |
| 07:27.30 | MaliutaLap | are the trunks registering? what does the Dial() command look like |
| 07:45.37 | *** join/#asterisk [Outcast] (~outcast@64.206.121.41) |
| 07:56.51 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:00.59 | *** join/#asterisk wanna (~wanna@194.183.244.5) |
| 08:02.50 | *** join/#asterisk aaabbb (~kvirc@yauza2-gw.primelink.ru) |
| 08:02.54 | aaabbb | hi there |
| 08:05.57 | aaabbb | I use directmedia for almost all peers, but I want that some peers don't use it, even if they have directmedia=yes |
| 08:06.02 | aaabbb | how to do this? |
| 08:15.46 | sekil | directmedia=no? |
| 08:15.50 | sekil | per peer |
| 08:18.39 | aaabbb | Peer A and peer B use direct RTP |
| 08:18.39 | aaabbb | Peer A and peer C use direct RTP |
| 08:18.39 | aaabbb | Peer B and peer C don't use direct RTP and must proxy all RTP traffic via Asterisk |
| 08:18.50 | aaabbb | how to do this? |
| 08:19.40 | sekil | probably in the dialplan |
| 08:20.52 | aaabbb | and how about directmediadeny/directmediapermit? just can't figure out to what peer apply this |
| 08:21.16 | aaabbb | to caller or to called |
| 08:31.58 | *** join/#asterisk areski (~areski@80.174.128.93.dyn.user.ono.com) |
| 08:39.37 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 08:50.18 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 08:51.11 | EmleyMoor | I'm trying to understand how Asterisk can handle the # key in the dialplan - is it necessary to use a pattern if there's a # anywhere in the extension? |
| 08:58.49 | *** join/#asterisk NoobSaibot (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
| 09:00.53 | *** join/#asterisk aness (~aness@cpe-74-68-148-41.nyc.res.rr.com) |
| 09:09.55 | *** part/#asterisk ruben23 (~OpenDIAL@112.198.77.114) |
| 09:33.05 | EmleyMoor | Hmmm... now anything I dial with * at the beginning never seems to make it to my Asterisk setup... |
| 09:34.21 | EmleyMoor | resets his phone |
| 09:35.26 | *** join/#asterisk [Outcast] (~outcast@64.206.121.41) |
| 09:35.51 | EmleyMoor | Address Incomplete shows on my phone... |
| 09:36.28 | EmleyMoor | May have found it |
| 09:36.38 | EmleyMoor | Yes! |
| 09:48.29 | *** join/#asterisk mcrownover_ (~mcrownove@remote.gawest.com) |
| 10:04.34 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 10:15.43 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 10:18.00 | *** join/#asterisk stevenm (~stevenm@212.57.232.170) |
| 10:29.45 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 10:31.18 | *** join/#asterisk cervajs2 (~cervenka@178.148.broadband4.iol.cz) |
| 10:33.38 | *** join/#asterisk NoobSaibot_ (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
| 11:03.53 | *** join/#asterisk areski (~areski@130.98.135.37.dynamic.jazztel.es) |
| 11:10.25 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 11:10.41 | *** join/#asterisk adsc (1f180acd@gateway/web/freenode/ip.31.24.10.205) |
| 11:19.54 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 11:21.40 | adsc | i have a lot of registration spam on my asterisk, how do I stop that? |
| 11:22.34 | *** join/#asterisk tzafrir_ (~tzafrir@local.xorcom.com) |
| 11:49.59 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 11:52.04 | *** join/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
| 11:52.25 | Tuju | where i can find all 1.8 sip.conf config syntaxes? |
| 11:53.30 | Tuju | it seems that lot of discussion is about old syntax and it's real difficult to adapt those into my version. |
| 12:00.24 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:09.31 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:11.55 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 12:13.11 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:14.21 | EmleyMoor | Anyone here any good with t38modem? I'm getting media type not found errors... |
| 12:17.34 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
| 12:17.37 | WIMPy | adsc: decrease the log level or disconnect from the internet. |
| 12:17.50 | WIMPy | Tuju: In the sample config. |
| 12:19.07 | *** join/#asterisk LiuYan (~root@unaffiliated/liuyan) |
| 12:23.59 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 12:26.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:26.21 | adsc | i tried core set verbose 0 and core set debug 0 but the spam is still in the console |
| 12:26.26 | Tuju | WIMPy: ack. |
| 12:26.36 | Tuju | is there a way to debug on single peer's packets? |
| 12:26.56 | Tuju | sip set debug peer x, sip set debug on dumps way too much stuff. |
| 12:26.58 | WIMPy | adsc: logger.conf |
| 12:27.44 | WIMPy | Tuju: You can only restrict it to a single IP. So if you have multiple peers comming from the sam IP, no. |
| 12:27.54 | adsc | i'm also in the process of setting up the firewall so that it only accepts sip from known addresses, this should help, right? |
| 12:28.25 | WIMPy | adsc: Yes |
| 12:30.34 | Tuju | WIMPy: how do i restrict it? |
| 12:32.01 | [TK]D-Fender | Tuju: "sip set debug ip [ip]" |
| 12:32.10 | WIMPy | The way you just wrote yourself. |
| 12:32.17 | Tuju | i did that but i still get tons of others. |
| 12:32.19 | *** join/#asterisk Champi (Champi@damn.e-leet.be) |
| 12:32.47 | Tuju | is there a way to list active debugged peers list? |
| 12:32.53 | *** join/#asterisk joesmoe (~joesmoe@2601:1:b200:b9:211:32ff:fe18:ef25) |
| 12:33.28 | WIMPy | There can only be one or all. |
| 12:33.46 | Tuju | then it's not filtering 'em. |
| 12:34.00 | WIMPy | Each sp set debug .. replaces whatever you had before. |
| 12:34.15 | Tuju | i'm trying to debug misbehaving trunk and i get desktop phone packets on screen. |
| 12:34.32 | Tuju | and that desktop phone is not taking part of call, any way. |
| 12:35.03 | WIMPy | Sounds like you just used "on" instead of "ip <ip>" or "peer <peer>". |
| 12:35.54 | Tuju | nope, i did not. :) |
| 12:36.07 | Tuju | said: sip debug peer elion |
| 12:36.46 | WIMPy | And that peer exists and has a valid ip associated with it? |
| 12:37.37 | Tuju | yep. |
| 12:37.52 | Tuju | it appears to be Ericsson MTAS or something. |
| 12:37.56 | *** join/#asterisk FreezingCold (~FreezingC@CPE602ad06bea2a-CM602ad06bea27.cpe.net.cable.rogers.com) |
| 12:38.09 | Tuju | http://www.ericsson.com/ourportfolio/products/multimedia-telephony-application-server-mtas |
| 12:40.30 | [TK]D-Fender | [08:32]Tujui did that but i still get tons of others. <- show us |
| 12:41.06 | [TK]D-Fender | Tuju: And you can't specifically debug a peer that hasn't registered |
| 12:41.26 | Tuju | yes, i know that. |
| 12:42.55 | [TK]D-Fender | Tuju: Show us your attempt to enable sip debug on a specific and getting other stuff after |
| 12:44.13 | *** join/#asterisk riess82 (~riessma@80-121-3-100.adsl.highway.telekom.at) |
| 12:48.17 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 13:00.59 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-34-104.user.veloxzone.com.br) |
| 13:09.45 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 13:11.42 | *** join/#asterisk claudiu (c3d8da0a@gateway/web/freenode/ip.195.216.218.10) |
| 13:11.52 | Tuju | any idea why some devices fail to register because registration is attempted with ip-address, instead of hostname? |
| 13:12.01 | Tuju | device config has hostname. |
| 13:17.32 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:20.06 | *** part/#asterisk LiuYan (~root@unaffiliated/liuyan) |
| 13:21.16 | claudiu | Hi there! I am not sure if this is on purpose but when I try to mute a channel from a Stasis Application and the channel was added to a bridge, then the mute request will destroy the channel. |
| 13:22.53 | claudiu | by the way, the Asterisk version is 12.5.0 |
| 13:26.27 | *** join/#asterisk adsc (1f180acd@gateway/web/freenode/ip.31.24.10.205) |
| 13:27.12 | adsc | i have the following error message: chan_sip.c:14435 check_auth: username mismatch, have <phone1>, digest has <phone2> |
| 13:27.35 | adsc | but I have a peer defined in sip.conf with defaultuser=phone2 |
| 13:28.47 | adsc | when I do sip show peers, i see them (status unmonitored) |
| 13:29.16 | adsc | this started when I changed the type from friend to peer |
| 13:30.10 | Tuju | adsc: you can put 'qualify=yes' for peer config to see is that registered. |
| 13:30.31 | adsc | thanks, i'll try that |
| 13:31.01 | Tuju | imo type=peer is history, nowdays you can use type=friend trunks too. |
| 13:31.36 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 13:31.42 | adsc | it's not a trunk, it's a phone device, and i switched to peer because of a website about asterisk security |
| 13:31.58 | [TK]D-Fender | adsc: You need them to be "friend" and not "peer" if they're going to be at the same IP |
| 13:32.42 | adsc | ah, okay |
| 13:32.54 | [TK]D-Fender | peer= auth by IP |
| 13:32.59 | adsc | yeah, they are both registered from the same Siemens Gigaset box |
| 13:33.06 | [TK]D-Fender | peerand that hits the FIRST one that matches.... |
| 13:34.16 | adsc | the article talked about how friend is bad because it can be used to circumvent user/pw check |
| 13:34.42 | adsc | but I added deny=0.0.0.0/0.0.0.0 and permit=192.168.1.16/24 |
| 13:34.48 | adsc | so I guess I should be fine |
| 13:35.19 | *** join/#asterisk [Outcast] (~outcast@64.206.121.41) |
| 13:36.55 | [TK]D-Fender | [09:34]adscthe article talked about how friend is bad because it can be used to circumvent user/pw check <- the only thing that stops a PW check is if you tell it not to. |
| 13:37.16 | adsc | okay, thanks |
| 13:37.25 | adsc | it works now with type=friend |
| 13:37.37 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 13:37.43 | adsc | but I guess I can't stop the registration brute force spam |
| 13:37.55 | adsc | because the router doesn't have a full nat, only port forwarding |
| 13:38.10 | adsc | so I can't restrict the SIP forwarding based on source IP unfortunately |
| 13:38.48 | adsc | is it possible to configure logging to not show failed registrations from unknown ips? |
| 13:38.49 | *** join/#asterisk aross42 (~aross@h46-190.reznet.ucalgary.ca) |
| 13:38.53 | [TK]D-Fender | [09:37]adscbut I guess I can't stop the registration brute force spam <- sure you can... that's what FIREWALLS are for |
| 13:39.07 | [TK]D-Fender | adsc: fail2ban <--------- |
| 13:39.22 | adsc | yeah, it does have a firewall, but the port forwarding of the router circumvents it |
| 13:39.54 | [TK]D-Fender | adsc: fail2ban on your server will block off the spam after a few tries |
| 13:40.46 | adsc | hmmm, thanks, but I can't install things on this server |
| 13:41.46 | [TK]D-Fender | adsc: well isn't that productive.... |
| 13:55.49 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-blfyyoblsxgijvas) |
| 13:55.49 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:57.01 | *** join/#asterisk ggayan (~ggayan@190.215.47.74) |
| 13:57.46 | cervajs2 | any experience with asterisk11+vp8_patch+webrtc+video(vp8) ? |
| 14:01.43 | adsc | if you do core set verbose X in console, it only changes it for the console, right? |
| 14:03.38 | Zogot | is there somewhere to report bugs in asterisk wiki? |
| 14:04.09 | Zogot | really minor one, https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime the Endpoint Population, the 6th mysql statement has ` instead of ' |
| 14:04.23 | Zogot | '102` |
| 14:05.17 | adsc | isn't the point of a wiki that users can edit it? |
| 14:05.52 | Zogot | didnt see a means to register |
| 14:05.59 | newtonr | Zogot, typically just add a comment on the page |
| 14:06.41 | Zogot | newtonr: hehe, i dont see that either :p |
| 14:06.46 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 14:08.33 | newtonr | the wiki edit access is restricted to very active community members to maintain a certain quality level. Though if anyone wants access to contribute regularly, feel to free to ask and if you can show some decent content then you'll likely be given edit access. |
| 14:08.44 | newtonr | Zogot, https://signup.asterisk.org |
| 14:09.15 | newtonr | Zogot, Signing up there gives you an account used across wiki.asterisk.org, issues.asterisk.org, reviewboard.asterisk.org , etc |
| 14:09.34 | newtonr | then you can login to the wiki and add comments |
| 14:10.05 | Zogot | newtonr: ah thanks a bunch. |
| 14:10.53 | newtonr | Zogot, or if you find major issues with the wiki you can file them as a bug on issues.asterisk.org |
| 14:11.07 | Zogot | newtonr: a lot of it was generated though correct? |
| 14:11.17 | Zogot | or perhaps i was mistaken |
| 14:12.23 | file | some is generated, some is not |
| 14:12.36 | newtonr | The command reference sections under "Asterisk X Documentation" sections are generated from source documentation |
| 14:12.48 | newtonr | most everything else is hand-crafted |
| 14:13.32 | file | out of wood |
| 14:13.38 | newtonr | For example everything under here is auto-generated: https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Command+Reference |
| 14:13.43 | *** join/#asterisk danjenkins_ (~dan@nat/digium/x-thbqcgehqxtkkycg) |
| 14:15.08 | newtonr | Zogot, also, I'll just fix the typo you found instead of you having to go leave a comment. |
| 14:15.17 | *** part/#asterisk ggayan (~ggayan@190.215.47.74) |
| 14:15.26 | Zogot | newtonr: ah ok, il delete my comment then :p |
| 14:15.38 | newtonr | :D |
| 14:15.52 | newtonr | At least you now have an account for the future! |
| 14:16.41 | Zogot | newtonr: yeh thats great. do want to contribute where i can, there was another little bug i found with the alembic scripts a few days ago |
| 14:17.11 | Zogot | does everything go through the issue tracker or is there some other means to follow commits to the project? |
| 14:17.24 | file | everything goes through the issue tracker |
| 14:17.40 | file | there's also the commits mailing list to actually see the commits |
| 14:18.00 | Zogot | ah ok, im on the ?normal? mailing list |
| 14:18.31 | Zogot | great found it, thanks file |
| 14:18.44 | newtonr | http://lists.digium.com/mailman/listinfo/asterisk-commits |
| 14:18.49 | newtonr | too late |
| 14:19.50 | newtonr | Zogot, you can also look at reviewboard.asterisk.org to see reviews happening on code before it gets committed. |
| 14:20.51 | newtonr | Helping to review code if you are capable is always appreciated. Otherwise you can always help test patches or new features. |
| 14:21.14 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-szacmcpwtruifatm) |
| 14:21.18 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Code+Review |
| 14:21.56 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-toseqyvxzaskwlel) |
| 14:22.00 | Zogot | ah great links, thanks newtonr |
| 14:24.38 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 14:25.27 | *** join/#asterisk crised (~crised@186.67.181.203) |
| 14:27.47 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 14:27.55 | *** part/#asterisk fullstop (~fullstop@107.170.53.155) |
| 14:28.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 14:29.04 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-flsucaylaxwnpvqd) |
| 14:29.05 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:29.44 | newtonr | Zogot, no problem. Feel free to message me in here or privately if you need help contributing in any way. |
| 14:29.57 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:30.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:31.34 | *** join/#asterisk jepperl (3e2c8712@gateway/web/cgi-irc/kiwiirc.com/ip.62.44.135.18) |
| 14:34.05 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 14:35.16 | jepperl | Hello guys. I am having a problem where if i make a call from outside into an extension, everything works perfect, but when i try to call my cellphone from the very same extension, i do not get any sound on the extension. I get the sound FROM the extension and from the perspective of my cellphone, everything works as intended. But no audio comes ou |
| 14:35.16 | jepperl | t to the extension. I have pcap'ed the RTP packets and i am able to decode them and hear what should have been transmitted to the extension, but on the extensions pcap trace no signs of the packets exists. Is this a common issue anyone might have a quick answer for? :) |
| 14:36.08 | adsc | if you see spawn extension blabla exited non-zero in console, is this a fault in the dial plan or normal? |
| 14:37.04 | Tuju | i've to admit that sip calls are the most diffcult thing i've debugged on unix. |
| 14:37.14 | Katty | hello my asterisk does not work at all how to to fix pls??? is urgent thx. |
| 14:37.16 | [TK]D-Fender | adsc: Normal |
| 14:37.21 | Tuju | and i've been around with these systems since 1993. |
| 14:37.26 | [TK]D-Fender | adsc: Not a faullllllt.. the call is done |
| 14:37.50 | adsc | I ask because I have a problem that every second incoming call or so is terminated immediately |
| 14:38.20 | [TK]D-Fender | adsc: Go look at the calls in more detail then |
| 14:39.44 | adsc | the only thing that happens in console is this (verbosity=10, debug=3) : == Using SIP RTP CoS mark 5 |
| 14:39.51 | adsc | nothing else |
| 14:41.06 | newtonr | jepperl, sounds like a typical one way audio issue. Could be firewall, SIP ALG, mis-configuration, a variety of things. No real quick answer. |
| 14:42.03 | Katty | [TK]D-Fender: how to fix my asterisk??? |
| 14:42.22 | [TK]D-Fender | adsc: Well you aren't looking at SIP DEBUG... so there's a gap to fill |
| 14:46.21 | adsc | i turned sip debug on, let's see |
| 14:46.22 | *** join/#asterisk jiku (~jiku@182.71.136.242) |
| 14:46.25 | jepperl | newtonr, i guess you're right. Its hard to describe, its probably nat :) i will go a bit more in depth with that |
| 14:46.42 | *** join/#asterisk aness (~aness@cpe-74-68-148-41.nyc.res.rr.com) |
| 14:47.01 | Tuju | it would help-a-lot to have a call-debug-flow-parser that would higlight and colorize meaningful information from different packets. |
| 14:47.38 | WIMPy | Tuju: Well, wireshark is a little nicer. |
| 14:47.39 | Tuju | wireshark probably has that but it's useless when those packets appear in remote host. |
| 14:48.11 | Tuju | wireshark also has some remote probe, but i have only found it available on windows. |
| 14:48.27 | Tuju | why the life has to be so unfair? I'm just asking. |
| 14:48.33 | Qwell | create the pcap with tcpdump |
| 14:48.38 | WIMPy | There are many tools to capture pcap files for later analysis. |
| 14:49.18 | Tuju | yep, i've done that past, it's just tedious as well driving your pupils screwed with those debug listings. |
| 14:49.57 | WIMPy | Well, if you're not in to S&M, don't use SIP. |
| 14:50.14 | Tuju | ideally I would give remote host name to wireshark and it would make ssh tunnel to that remote host and do its thing there and show it on my local screen. |
| 14:50.30 | Tuju | WIMPy: well, i am. |
| 14:50.37 | Qwell | Well, you can't, so use tcpdump. |
| 14:50.56 | Tuju | tcppumpupthejam |
| 14:51.38 | WIMPy | You could try ssh tcpdump|wireshark. |
| 14:52.24 | Tuju | i've tunneled browser via ssh and it wasn't that hard, howcome nobody hasn't come up with such remote probe solution? |
| 14:53.24 | WIMPy | You can do the same with wireshark, but that's not an efficient way to do it. |
| 14:53.46 | mjordan | probably because ssh -X works pretty well |
| 14:55.44 | adsc | i have the sip debug output of an immediately terminated call here, but I have no idea what to look for: http://pastie.org/9559071#1-3,5,9-10,54,57,59-60 |
| 14:56.32 | adsc | mind that this only happens every other call |
| 14:56.36 | [TK]D-Fender | adsc: Looking for 41315280456 in default (domain 192.168.1.225) SIP/2.0 404 Not Found |
| 14:56.36 | adsc | or so |
| 14:56.45 | [TK]D-Fender | adsc: Looks pretty clear to me... no match in your dialplan |
| 14:56.59 | adsc | but why does it work if I immediately call after that? |
| 14:57.02 | adsc | the same number? |
| 14:57.20 | adsc | wouldn't it never work if the dialplan was the problem? |
| 14:57.28 | [TK]D-Fender | adsc: I don't see a good one here |
| 14:57.39 | adsc | that's only the failed one |
| 14:57.43 | [TK]D-Fender | adsc: you didn't give us anything to compare |
| 14:57.47 | [TK]D-Fender | adsc: Go fix that3 |
| 14:57.51 | adsc | alright, let me try to get a good one |
| 14:58.17 | cervajs2 | file: can you please check https://github.com/onsip/SIP.js/issues/97#issuecomment-55751442 and comment if egreenmachine has right about asterisk SIPoWS implementation? |
| 14:58.59 | Tuju | remote debugging with asterisk is especially tricky when there is a lot of terminals, hence lot of traffic and problematic calls or attempts are short, hard to pinpoint and hence the debugging control/triggering should be remotely available in the same gui. |
| 14:59.34 | Tuju | sea of trouble with one drop of information you need to find and figure out. :-/ |
| 15:00.39 | Tuju | rpcapd is the remote probe for WS |
| 15:02.03 | Tuju | wireshark -i <(ssh root@firewall tcpdump -s 0 -U -n -w - -i eth0 not port 22) could that work? |
| 15:02.55 | WIMPy | "udp port 5060" should be enough. |
| 15:03.11 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:03.34 | Tuju | http://www.pawelko.net/linux/17-Rpcapd-For-Linux-Remote-Sniffing-With-Etherealwireshark claims to have 'compiled for linux' - where's the source? |
| 15:04.40 | Tuju | http://www.pawelko.net/linux/38-Here Compiling rpcapd for linux |
| 15:06.22 | adsc | this is the successful sip debut output: http://pastie.org/9559100 |
| 15:06.45 | [TK]D-Fender | adsc: Looking for 41315280456 in fromtwilio (domain 192.168.1.225) |
| 15:06.53 | [TK]D-Fender | adsc: different CONTEXT |
| 15:07.02 | [TK]D-Fender | adsc: You're not paying attention to where you are sending your calls |
| 15:07.37 | [TK]D-Fender | Good peer = Found peer 'twilioip-3' for '+41799445543' from 107.21.231.147:5060 |
| 15:07.41 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 15:07.42 | [TK]D-Fender | Bad peer = Found peer 'twilio' for '+41799445543' from 107.21.211.20:5060 |
| 15:07.51 | adsc | hmmm |
| 15:07.55 | adsc | i see |
| 15:08.30 | adsc | thanks, I'll try to fix this |
| 15:08.32 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:08.35 | protocoldoug | Anyone have a solution for how to access the Asterisk CLI while running Asterisk in a Docker container? |
| 15:08.54 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 15:10.41 | Tuju | <PROTECTED> |
| 15:15.33 | anonymouz666 | anyone in here already make a bridge "half" SRTP? client1 <-> SRTP <-> GW01 <-> SRTP <-> Asterisk <-> RTP <-> client2 ? |
| 15:16.46 | [TK]D-Fender | anonymouz666: I don't see why this wouldn't work wince * is a B2BUA |
| 15:17.01 | [TK]D-Fender | client1 and client2 don't ahve to be anything alike |
| 15:18.11 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-yrwtfhiwdzxuszny) |
| 15:18.37 | adsc | it seems I was able to fix it, but please help me understand... |
| 15:18.53 | adsc | I have two contexts in sip.conf for incoming and outgoing calls |
| 15:19.13 | adsc | and I have two contexts in extensions.conf, also for incoming and outgoing |
| 15:19.40 | adsc | i fixed it now by adding the correct extensions context in sip.conf for each sip context |
| 15:20.46 | adsc | i thought it worked like that: on incoming call, asterisks looks in extensions for a match and then tries to find the listed endpoint in sip.conf |
| 15:21.13 | [TK]D-Fender | adsc: No, it idetifies the SOURCE to determine WHAT dialplan applies to it |
| 15:21.35 | adsc | hmmm |
| 15:21.40 | [TK]D-Fender | adsc: Match peer -> send to peer's context in dialplan. |
| 15:21.48 | [TK]D-Fender | adsc: You've missed the basics |
| 15:21.59 | adsc | yeah, it's really ad hoc |
| 15:22.08 | [TK]D-Fender | adsc: It doesn't pick a context to dump them in out of thin air. |
| 15:22.16 | adsc | i'm actually software developer and have nothing to do normally with telephony |
| 15:22.18 | [TK]D-Fender | Not ad-hoc. the method is very explicit |
| 15:22.39 | adsc | no i mean my "experience" with asterisk |
| 15:22.45 | [TK]D-Fender | #1 : Who are you? #2 : What do you want? |
| 15:22.58 | adsc | I want to go home and drink a beer |
| 15:23.26 | adsc | anyway, thanks for all the help and helping me understand, you have been very patient and kind |
| 15:23.26 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-rqngtidmlfdjjyun) |
| 15:24.17 | [TK]D-Fender | adsc: Glad it's now clear for you. |
| 15:24.17 | pabelanger | okay |
| 15:24.22 | pabelanger | n00b question |
| 15:25.23 | pabelanger | how do I control the context for blindxfer in features.conf for app_queue? |
| 15:25.36 | pabelanger | WARNING[2750]: features.c:2136 builtin_blindtransfer: Extension '1' does not exist in context 'app_queue_gosub_virtual_context' |
| 15:25.52 | pabelanger | I want to change the virtual context to something else |
| 15:28.16 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 15:29.09 | Tuju | wireshark doesn't have that remote capture feature anymore. |
| 15:29.46 | Tuju | https://www.winpcap.org/archive/ probe src is there. |
| 15:31.50 | *** join/#asterisk josemslopes (~josemslop@adonis.ipbrick.com) |
| 15:37.21 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:1dd0:221:6aff:feb8:e0b2) |
| 15:39.59 | mjordan | pabelanger: try the TRANSFER_CONTEXT channel variable on the inbound channel. |
| 15:40.13 | mjordan | ideally, make it inheritable, just in case you are using Local channel members |
| 15:41.40 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 15:41.44 | pabelanger | mjordan, ya, that's what I am trying now. Thanks. I was looking for a config file setting at first |
| 15:47.21 | *** join/#asterisk roentgen (~none@openvpn/community/support/roentgen) |
| 15:47.50 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:47.53 | spicyramen_ | newbie question, how to force * to play file in ulaw |
| 15:47.54 | spicyramen_ | exten => 1000,1,Playback(demo-echotest) |
| 15:48.00 | spicyramen_ | now it invokes .gsm file |
| 15:48.04 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 15:49.08 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 15:50.10 | marceloamorim | guys, I`m moving ael to normal way, I`m trying to put gotoif in my context, the syntax at the documentation is GotoIf(condition?[labeliftrue:[labeliffalse]]). I`m trying to use this way, same => n,GotoIf(${CHANISAVAIL} != ""?[dial:[got]]) |
| 15:51.06 | [TK]D-Fender | marceloamorim: that is not a valid expression |
| 15:51.15 | Qwell | marceloamorim: Conditions must be wrapped in $[] |
| 15:51.16 | [TK]D-Fender | $[] <--------- expression |
| 15:51.28 | rmudgett | spicyramen_: By default the gsm files are installed. You need to install the ulaw versions. Use "make menuselect" in the Core Sound Packages and select the ulaw versions. |
| 15:51.39 | *** join/#asterisk e4voip (uid13742@gateway/web/irccloud.com/x-cjjuifhbyxipzplw) |
| 15:51.52 | spicyramen_ | thanks srmudgett |
| 15:51.57 | [TK]D-Fender | spicyramen_: It will pick the most efficient format you have based on the call you have |
| 15:51.58 | marceloamorim | uhmm, thx |
| 15:52.47 | spicyramen_ | I have a webrtc call and want to play prompts for callers via * my browser is sending ulaw,alaw, & opus |
| 15:55.14 | marceloamorim | ast_yyerror(), syntax error, unexpected '!=', expecting $end; Input: |
| 15:56.08 | [TK]D-Fender | marceloamorim: You also have quotes on ONE side of your previously malformed expression |
| 15:56.22 | [TK]D-Fender | marceloamorim: and I'm betting that that var is BLANK so you have NOTHING on the left side... |
| 15:57.20 | marceloamorim | I wish to compare if chanisavail is null |
| 15:58.43 | Tuju | wireshark -i <(ssh root@firewall tcpdump -s 0 -U -n -w - -i eth0 not port 22) did anyone try that? |
| 15:58.53 | Tuju | mine starts but doesn't display anything. |
| 15:59.15 | [TK]D-Fender | marceloamorim: Yes well you can't have something that is BLANK on one side |
| 16:00.20 | [TK]D-Fender | marceloamorim: http://www.voip-info.org/wiki/view/Asterisk+Expressions |
| 16:01.21 | marceloamorim | mmm, let me check |
| 16:01.26 | Tuju | http://wiki.wireshark.org/CaptureSetup/Pipes |
| 16:02.12 | *** join/#asterisk [Outcast] (~outcast@64.206.121.41) |
| 16:07.39 | Tuju | it was ssh-keys that failed. now it ACTUALLY WORKS. whee. |
| 16:14.16 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 16:17.29 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:25.26 | *** join/#asterisk riess82 (~riessma@188-22-108-59.adsl.highway.telekom.at) |
| 16:26.02 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 16:27.40 | *** join/#asterisk zamba (marius@flage.org) |
| 16:32.05 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 16:35.42 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 16:35.50 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:39.34 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-ndygcxcmloimetsq) |
| 16:43.46 | *** join/#asterisk danjenkins__ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 16:44.32 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 16:46.47 | *** join/#asterisk NoobSaibot (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
| 17:01.07 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 17:02.10 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 17:08.35 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:09.24 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:14.04 | cervajs2 | can someone pls check https://github.com/onsip/SIP.js/issues/97#issuecomment-55751442 and comment if egreenmachine has right about asterisk SIPoWS implementation? |
| 17:16.40 | *** join/#asterisk areski (~areski@80.174.128.93.dyn.user.ono.com) |
| 17:20.50 | *** join/#asterisk [Outcast] (~outcast@64.206.121.41) |
| 17:20.51 | *** join/#asterisk NoobSaibot_ (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
| 17:21.29 | Tuju | http://sipp.sourceforge.net/ |
| 17:22.55 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:23.21 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:23.32 | *** join/#asterisk NoobSaibot_ (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
| 17:24.08 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-tnynznzhslfhecit) |
| 17:24.34 | mjordan | cervajs2: that RFC didn't exist when we wrote the SIP over WS code. If someone has an interoperability bug they'd like to file, they can do so on issues.asterisk.org. |
| 17:26.03 | cervajs2 | mjordan: ok i will try fill it. thanks |
| 17:30.23 | *** join/#asterisk FreezingCold (~FreezingC@CPE602ad06bea2a-CM602ad06bea27.cpe.net.cable.rogers.com) |
| 17:32.18 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 17:33.06 | *** join/#asterisk X-Rob (sid14615@gateway/web/irccloud.com/x-hfwhnxuexhukpsmy) |
| 17:37.12 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 17:37.27 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:37.41 | Zogot | ahoyhoy |
| 17:37.50 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:38.40 | Tuju | is 407 something that always leads into call termination? |
| 17:39.00 | Tuju | it's "Authentication required" |
| 17:41.11 | file | all calls inevitably terminate, but if you mean if that leads to call failure in all cases... no |
| 17:41.43 | Tuju | in my case next one is BYE. |
| 17:43.20 | Tuju | http://tuju.fi/tmp/elion.407-20140916.png |
| 17:44.59 | file | if .38 is Asterisk then something challenged it for authentication, it had no credentials or they failed, and then the calling leg was hung up because that side failed |
| 17:45.01 | file | just a guess. |
| 17:45.34 | *** join/#asterisk oesteve (~oscar@145.Red-88-15-97.dynamicIP.rima-tde.net) |
| 17:46.28 | Tuju | .38 is asterisk, yes. |
| 17:47.16 | Tuju | it's registered to that .4 trunk and everything works fine if I pick up the call with phone that is directly connected to that .38. |
| 17:47.45 | file | without a full sip debug there can be only guesses |
| 17:47.54 | Tuju | picture's case is when i pick call behind another asterisk box. It connects but call breaks instantly. |
| 17:49.18 | Tuju | http://fpaste.org/133997/89708141/ |
| 17:49.58 | file | 217.159.187.4 wants to authenticate and Asterisk has no credentials |
| 17:50.02 | Tuju | i'm not sure are all packets there. |
| 17:50.39 | Tuju | file: it has, sip show registry shows that it's registed, i can call outbound calls, i cal receive fully inbound with phone that is connected to .38 |
| 17:51.12 | Tuju | .38 has insecure=port,invite but it doesn't really change anything if i change that setting. |
| 17:51.26 | file | insecure only controls how calls from something are matched |
| 17:51.31 | file | it does not control outbound authentication |
| 17:51.37 | Tuju | ack |
| 17:51.57 | file | whatever 217.159.187.4 is is requesting authentication on that INVITE, so if you have a working one for an INVITE to that... then compare |
| 17:52.06 | Tuju | there is another trunk that is between these two asterisk boxes. |
| 17:52.31 | Tuju | yep, i thought to try that. |
| 17:52.41 | Tuju | just today noticed that another phone works fine. |
| 17:53.13 | Tuju | it's just real pita, this broken path is my work phone. |
| 18:07.17 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 18:08.54 | *** join/#asterisk cunningpike (~cunningpi@2602:304:6eaf:6280:1583:19a5:e8b2:1387) |
| 18:16.02 | *** join/#asterisk oesteve (~oscar@145.Red-88-15-97.dynamicIP.rima-tde.net) |
| 18:20.14 | *** join/#asterisk cunningpike (~cunningpi@2602:304:6eaf:6280:1583:19a5:e8b2:1387) |
| 18:24.49 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-lelwjnfbmkwimuok) |
| 18:29.38 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:47.25 | *** join/#asterisk LiohAu (~LiohAu@bac69-8-88-165-4-152.fbx.proxad.net) |
| 18:52.34 | *** join/#asterisk w3rt (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 18:52.47 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:56.26 | Tuju | file: i did debugging |
| 18:57.25 | Tuju | now it's clear that working call just does initial INVITE, rings the phones and one says 200 OK, then it sends ACK's back and call starts. |
| 18:57.59 | Tuju | but the problematic two-asterisk setup does second INVITE - and that causes 407. |
| 18:59.37 | Tuju | btw, wireshark doesn't really support sip call debugging, there is no call-seq filtering etc. |
| 19:01.55 | Tuju | that second invite is 'in-dialog' |
| 19:04.34 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:14.36 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:23.46 | Tuju | i think i figured why it fails. |
| 19:24.56 | Tuju | my second trunk doesn't have that many codecs and once 200 OK comes back from there, middle asterisk has to 'reclaim its promises about codecs' and restart SDP negotiations and do in-dialog invite. |
| 19:36.08 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:45.25 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:47.52 | *** join/#asterisk oesteve (~oscar@145.Red-88-15-97.dynamicIP.rima-tde.net) |
| 19:48.51 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 19:50.22 | *** join/#asterisk Iamnacho (~Iamnacho@ip72-213-56-241.om.om.cox.net) |
| 19:52.52 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 19:54.12 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-34-104.user.veloxzone.com.br) |
| 19:55.31 | Tuju | well, it wasn't that. it was: middle asterisk tried to put proxies to talk each other and itself hop away from the middle, that caused in-dialog re-invite and that lead to 407 auth required that didn't work with ericssson. |
| 19:55.45 | Tuju | that was a HAIRY ONE. |
| 19:55.46 | *** join/#asterisk mirela666 (~mirko.bra@95.180.126.160) |
| 19:57.06 | *** join/#asterisk [Outcast] (~outcast@64.206.121.41) |
| 20:01.16 | EmleyMoor | Is there anywhere I can get some help on t38modem? It seems not to be able to find support for any of the media types when my asterisk puts a call through to it... |
| 20:06.04 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-34-104.user.veloxzone.com.br) |
| 20:07.54 | *** join/#asterisk areski (~areski@80.174.128.93.dyn.user.ono.com) |
| 20:08.22 | Tuju | EmleyMoor: what is that? |
| 20:09.38 | Tuju | i'm so happy that i got my hack working. |
| 20:10.17 | Tuju | i think i've violated all possible labour rights working hour laws, so it's guten nacht for this channel... |
| 20:10.23 | *** part/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
| 20:10.54 | *** join/#asterisk yoavz (yoavz@yoavz.net) |
| 20:12.54 | EmleyMoor | Tuju: It's a modem emulation for handling faxes. |
| 20:13.21 | WIMPy | too late |
| 20:13.52 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:16.10 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:26.16 | *** join/#asterisk 44UAAAAAX (~snadge@2404:9400:1:0:216:3eff:fef0:6d8) |
| 20:27.23 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:36.17 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 20:38.59 | *** join/#asterisk smellis_werk (~smellis@2001:470:c452:2::9) |
| 20:40.49 | dan_j | Is it possible to specify global variables within specific contexts? Or can globals only be specified in one place? |
| 20:41.51 | EmleyMoor | dan_j: You can certainly access, and change the value of, a global variable within a context - if that's what you mean... |
| 20:46.59 | dan_j | Can I just do variable=value, like you do in the [globals] context? Or do I have to use Set()? |
| 20:48.02 | EmleyMoor | dan_j: You do it with Set() |
| 20:53.16 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:59.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:06.46 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 21:28.50 | *** join/#asterisk mokmeister (~quassel@95.45.41.60) |
| 21:34.19 | *** join/#asterisk aross42 (~aross@h46-190.reznet.ucalgary.ca) |
| 21:36.53 | *** join/#asterisk mcrownover (~mcrownove@remote.gawest.com) |
| 21:37.10 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:56.44 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 22:02.54 | *** part/#asterisk cervajs2 (~cervenka@178.148.broadband4.iol.cz) |
| 22:05.57 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 22:07.54 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 22:16.12 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 22:16.45 | *** join/#asterisk CunningPike (~CunningPi@d154-5-54-220.bchsia.telus.net) |
| 22:27.59 | *** part/#asterisk MarkS- (~mark@unaffiliated/mark21) |
| 22:34.58 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 22:36.56 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 22:37.21 | alexw | What happens when g729 licences are exceed? |
| 22:38.27 | lvlinux | if i understand it correctly, the g729 license is on the honor system---it will still work but you are supposed to pay for the channels that you will use. |
| 22:38.28 | [TK]D-Fender | it will negotiate G.729 anyway and the moment * has to transcode anything the call will drop like a rock |
| 22:38.41 | lvlinux | oh? well TK knows better than me... |
| 22:39.22 | alexw | Why can't it switch to alaw? |
| 22:39.29 | alexw | ulaw etc |
| 22:40.17 | [TK]D-Fender | Because it doesn't |
| 22:40.30 | [TK]D-Fender | * doesn't know it's out until the very moment it has to transcode |
| 22:40.31 | alexw | because it was designed that way to make more money I guess :) |
| 22:40.37 | [TK]D-Fender | Other calls are possibly in process |
| 22:40.45 | [TK]D-Fender | ANY one of those might drop off and free up licences |
| 22:40.53 | [TK]D-Fender | basically "too bad". Cover what you need |
| 22:41.20 | alexw | so maybe I should stick to ulaw/alaw to my VSP |
| 22:41.38 | alexw | and then g729 to the deskphones |
| 22:41.51 | alexw | as the PBX is hosted on another site |
| 22:42.18 | alexw | I don't want to be bombarded with calls from clients and it goes to silence - but we only have 2-3 deskphones |
| 22:42.32 | [TK]D-Fender | It won't go silent. |
| 22:42.42 | [TK]D-Fender | It'll go "click" as the calls die instantly |
| 22:42.42 | *** join/#asterisk CunningPike (~CunningPi@d154-5-54-220.bchsia.telus.net) |
| 22:42.48 | lvlinux | do your phones support ilbc? you could use that if so. |
| 22:44.01 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 22:47.36 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 23:10.02 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 23:11.30 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 23:13.12 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 23:13.32 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 23:13.33 | *** mode/#asterisk [+o file] by ChanServ |
| 23:16.42 | *** join/#asterisk diametric (~diametric@2604:3400:dc1:43:216:3eff:fe27:bf9d) |
| 23:16.50 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 23:22.37 | *** join/#asterisk AlHafoudh (~AlHafoudh@echo.freevision.sk) |
| 23:42.56 | *** join/#asterisk Arsenick (~arsenick@fedora/Arsenick) |
