| 00:00.07 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:38.37 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 01:03.28 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 01:20.47 | *** join/#asterisk klow (~textual@c-73-83-127-32.hsd1.wa.comcast.net) |
| 01:25.36 | *** join/#asterisk Arkanos-Morph (~Arkanos-M@182.64.142.227) |
| 01:29.15 | *** join/#asterisk Hello71 (~Hello71@wikipedia/Hello71) |
| 02:23.25 | *** join/#asterisk klow (~textual@c-73-83-127-32.hsd1.wa.comcast.net) |
| 02:25.59 | *** join/#asterisk mskenderian (uid158360@gateway/web/irccloud.com/x-clzxaraamvwykxxr) |
| 02:48.05 | *** join/#asterisk hfb (~hfb@cpe-108-185-247-93.socal.res.rr.com) |
| 03:41.21 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 04:06.16 | *** join/#asterisk jwpierce3 (~jwpierce3@173-16-50-64.client.mchsi.com) |
| 04:12.35 | *** join/#asterisk klow (~textual@c-73-83-127-32.hsd1.wa.comcast.net) |
| 04:51.25 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-arekqbbytqqniiqu) |
| 05:02.27 | *** join/#asterisk vader- (~Adium@pool-100-14-222-238.phlapa.fios.verizon.net) |
| 05:16.00 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 05:25.38 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-14.wall.uls.co.za) |
| 05:26.41 | *** join/#asterisk swimmercol (~lancelot@B92556C5.rev.sefiber.dk) |
| 05:35.27 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 05:41.07 | *** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66) |
| 05:42.00 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 05:50.54 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-14.wall.uls.co.za) |
| 06:03.56 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 06:08.46 | post-factum | V5_12R\X: so you say it is impossible to call from unencrypted peer to encrypted? sounds weird enough |
| 06:25.54 | *** join/#asterisk raspberrypifan (~raspberry@2604:2000:6016:be00:3c0c:ba48:3738:b6d5) |
| 06:36.45 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:38.00 | *** join/#asterisk jkroon (~jkroon@uls-154-73-35-201.wall.uls.co.za) |
| 06:43.59 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 06:44.15 | *** join/#asterisk Zogot (~Adium@185.21.52.255) |
| 06:59.46 | *** join/#asterisk AviiNL (~AviiNL@185.21.52.255) |
| 07:07.51 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:15.17 | post-factum | is there chan_sip equivalent to media_encryption_optimistic=yes in pjsip? |
| 07:22.18 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:37.31 | *** join/#asterisk fornax (~fornax@85.183.43.221) |
| 07:42.00 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 07:49.33 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:06.27 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-41-127.rn.hr.cox.net) |
| 08:17.33 | post-factum | file: it seems i've managed the issue by setting "sip:" instead of "sips:" in account id for csipsimple |
| 08:17.49 | post-factum | file: everything else is left as it was |
| 08:53.46 | *** join/#asterisk hellc2 (~hellc2@100.red-88-26-250.staticip.rima-tde.net) |
| 08:54.00 | *** join/#asterisk CeBe2 (~CeBe@a81-14-235-90.net-htp.de) |
| 09:04.35 | *** join/#asterisk klow (~textual@c-73-83-127-32.hsd1.wa.comcast.net) |
| 09:14.00 | *** join/#asterisk SunGod (~SunGod@92.55.116.125) |
| 09:30.44 | *** join/#asterisk Zogot (~Adium@185.21.52.255) |
| 10:14.29 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 10:47.08 | *** join/#asterisk c0rnoTa (~c0rnoTa@91.221.232.65) |
| 10:57.22 | *** join/#asterisk Zogot (~Adium@185.21.52.255) |
| 11:04.05 | post-factum | one more question on tls |
| 11:04.06 | post-factum | [May 10 14:03:26] ERROR[18053]: tcptls.c:609 handle_tcptls_connection: Problem setting up ssl connection: error:1407609B:SSL routines:SSL23_GET_CLIENT_HELLO:https proxy request |
| 11:04.08 | post-factum | [May 10 14:03:26] WARNING[18053]: tcptls.c:684 handle_tcptls_connection: FILE * open failed! |
| 11:04.16 | post-factum | https proxy request? |
| 11:04.31 | post-factum | i guess some morons scan my ip and try to use sip tls port as https proxy |
| 11:04.49 | post-factum | could i either disable those warnings or ban those morons dynamically? |
| 11:07.46 | *** join/#asterisk Sprocks (~Sprocks@BMTNON3746W-LP130-05-1176475167.dsl.bell.ca) |
| 11:12.46 | file | post-factum, no chan_sip equivalent for optimistic, oic @ csipsimple, and no on stopping the errors unless you disable errors in logger.conf and as for blocking if you get the IP you can firewall 'em... |
| 11:13.31 | post-factum | is that possible to get those ips in asterisk logs? |
| 11:13.47 | post-factum | i can see them with tcpdump only |
| 11:14.22 | file | I don't believe so, not currently, it'd be reasonable to have them there - create an issue for it |
| 11:17.43 | post-factum | in jira? |
| 11:17.50 | file | yes |
| 11:17.58 | post-factum | ok, will do now |
| 11:19.17 | *** join/#asterisk mub (~Jub@65-123-23-3.dia.static.qwest.net) |
| 11:20.05 | post-factum | file: what component should it be? core/logging? |
| 11:20.20 | file | I think there's a Core tcptls one... |
| 11:20.49 | file | otherwise just select whatever seems reasonable |
| 11:21.00 | file | if something else makes more sense it'll be changed during triage |
| 11:23.55 | post-factum | https://issues.asterisk.org/jira/browse/ASTERISK-26006 |
| 11:24.01 | post-factum | looks ok? |
| 11:24.24 | file | yes |
| 11:28.07 | *** part/#asterisk Hello71 (~Hello71@wikipedia/Hello71) |
| 11:38.18 | *** join/#asterisk acidfoo_ (~acidfoo@modemcable002.114-70-69.static.videotron.ca) |
| 11:55.13 | gregs | file: I upgraded a full patched FreePBX system from 13.8.2 to 13.9 earlier this evening and all of my PJSIP endpoints started dropping out. https://www.irccloud.com/pastebin/WzXWlEls/ |
| 11:55.18 | gregs | Any thoughts |
| 11:55.30 | file | what does the "pjsip set logger on" output show? |
| 11:55.51 | gregs | I've downgraded back to 13.8.2. Give me a minute and I'll put it back. |
| 11:56.59 | file | otherwise no idea... can't say I've seen any other people with that problem or any reports |
| 11:58.33 | file | er actually, stuff is being deleted... early... which is odd |
| 11:58.49 | gregs | Neither have I. I did a bit of poking around here and on the forums to see if anyone had come across it. All of the endpoints are on the same subnet as the FreePBX box so not sure what was causing it. |
| 11:58.57 | file | 30 seconds early roughly |
| 11:59.26 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:00.59 | gregs | file: Here's the log with pjsip set logger on: |
| 12:01.28 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 12:01.40 | gregs | http://pastebin.com/FfQfsQTM |
| 12:02.35 | file | yeah, something odd is up |
| 12:02.51 | gregs | If I go back to 13.8.2 then all is OK. |
| 12:03.07 | file | file an issue on https://issues.asterisk.org/jira with the logs (with timestamps), generated config files, against 13.9.0 and that it's a regression, and describe the environment under which it's running |
| 12:04.03 | gregs | OK. Which configs? All of them? Or just the PJSIP ones? |
| 12:04.13 | file | PJSIP ones should be fine |
| 12:04.27 | gregs | OK. Will do. Thanks. |
| 12:15.38 | *** join/#asterisk kolko (~kolko@46.48.58.17) |
| 12:17.49 | gregs | file: Quick question. Do I attach the logs after creating the ticket? I can't see anywhere on the bug form to do that? |
| 12:17.58 | file | yes |
| 12:18.01 | gregs | Thanks |
| 12:20.29 | *** part/#asterisk c0rnoTa (~c0rnoTa@91.221.232.65) |
| 12:24.11 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 12:25.59 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 12:26.11 | *** join/#asterisk pdugas (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
| 12:26.25 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:27.24 | gregs | file, thanks for your help. I've raised the ticket: https://issues.asterisk.org/jira/browse/ASTERISK-26007 |
| 12:27.28 | file | yup yup |
| 12:27.35 | file | gets an email on every JIRA change |
| 12:28.19 | gregs | Sorry. :-) |
| 12:48.39 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-fiuxlxwysraezxaw) |
| 12:48.39 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 12:48.58 | *** join/#asterisk Arkanos-Morph (~Arkanos-M@182.64.142.227) |
| 12:57.03 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 13:05.19 | *** join/#asterisk kkocaerkek (~kkocaerke@195.142.112.66) |
| 13:10.14 | kkocaerkek | hi all. i have a question about call transfers |
| 13:11.37 | kkocaerkek | i can read SIPTRANSFER_REFERER channel variable inside TRANSFER_CONTEXT during blind transfer |
| 13:12.15 | kkocaerkek | But i couldn't find how to read SIPTRANSFER_REPLACES variable during attended transfer |
| 13:12.41 | *** join/#asterisk seik0 (~seik0@178.68.238.61) |
| 13:12.51 | kkocaerkek | is there anybody can help me about call transfers? |
| 13:15.07 | seik0 | Hi! I have asterisk("A") -- <IAX2> -- asterisk("B") -- mobile-operator. When I do dial from cli on "B" to mobile phone - all ok. When I do dial from "A" to playback(some-file) in "B" - all ok. But when I do call from "A" to "B" and Dia(to-mobilele-phone) then I |
| 13:15.16 | seik0 | then I get hangup on "A" |
| 13:15.31 | seik0 | checked codecs - seems ok |
| 13:15.52 | seik0 | transcodings seems ok |
| 13:17.41 | seik0 | Looks like this in cli: http://pastebin.com/RPBJpf2X |
| 13:19.15 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:19.15 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:25.31 | *** join/#asterisk zamba (marius@flage.org) |
| 13:27.11 | infernix | is there a way to use 'originate' where an extension is dialled only after the origin answers? |
| 13:27.45 | infernix | e.g. originate SIP/myphone extension 103@mycompany - but only connect to extension 103 when SIP/myphone answers? |
| 13:28.50 | [TK]D-Fender | do the REVERSE |
| 13:29.17 | [TK]D-Fender | If you want B to answer then call A instead A then B... then sway the two |
| 13:29.19 | [TK]D-Fender | swap* |
| 13:29.31 | [TK]D-Fender | Put the thing to call first where it should go. |
| 13:29.35 | [TK]D-Fender | In whatever form you have to. |
| 13:30.03 | infernix | 103@mycompany always answers - it's a macro to a conference call phone |
| 13:31.04 | [TK]D-Fender | ]Go swap them |
| 13:34.41 | infernix | doesn't appear to work. for example "originate SIP/123456789@mysipprovider,90,D(wwww5w4w3w2w1w0w1w2w3w#w#) extension 1002@stations" |
| 13:34.57 | [TK]D-Fender | Of course not. |
| 13:35.12 | [TK]D-Fender | Where did the instructions tell you that you can shove all those parameters there? |
| 13:35.27 | [TK]D-Fender | You can't just shove DIAL's parameters anywhere you want |
| 13:36.08 | infernix | right but that's the issue. i can't originate 1234@mycontext extension 456@mystations |
| 13:36.20 | infernix | tech/data after originate only does SIP/foo. right? |
| 13:37.17 | seik0 | originate Local/1234@mycontext extension 456@mystations |
| 13:37.39 | infernix | ah, Local. great |
| 13:37.40 | infernix | tries |
| 13:41.30 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:46.49 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 13:47.25 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:50.25 | *** join/#asterisk rsanting (~rsanting@static-150-83-100-159.thenetworkfactory.nl) |
| 14:01.16 | *** join/#asterisk Zogot (~Adium@185.21.52.255) |
| 14:03.29 | *** join/#asterisk Hello71 (~Hello71@wikipedia/Hello71) |
| 14:03.42 | Hello71 | is asterisk 13 supposed to install a chan_pjsip.so? |
| 14:04.04 | Chainsaw | Hello71: No, just a res_pjsip.so |
| 14:04.12 | Chainsaw | Hello71: This question sounds very familiar somehow. |
| 14:04.32 | Hello71 | can't imagine why |
| 14:04.36 | file | it's actually a few modules |
| 14:04.41 | file | chan_pjsip.so is one of them |
| 14:05.03 | Hello71 | $ echo /usr/lib/asterisk/modules/*sip* -> /usr/lib/asterisk/modules/app_adsiprog.so /usr/lib/asterisk/modules/chan_sip.so |
| 14:05.22 | file | it will only be installed if dependencies are met |
| 14:05.48 | Chainsaw | In a menu-driven system designed to catch out packagers. Guess they've succeeded. |
| 14:06.04 | Hello71 | [binary R #] net-misc/asterisk-13.8.2::gentoo USE="caps iconv samples -alsa -bluetooth -calendar -cluster -curl -dahdi -debug -doc -freetds -gtalk -http -ilbc -ldap -libedit -libressl -lua -mysql -newt -odbc -osplookup -oss -portaudio -postgres -radius (-selinux) -snmp -span -speex -srtp -static -syslog -vorbis -xmpp" VOICEMAIL_STORAGE="file -imap -odbc" 0 KiB |
| 14:07.30 | file | PJSIP requires the PJSIP library, in 13.8 and above you can either use bundling (https://www.asterisk-blog.com/2016/03/16/asterisk-13-8-0-now-easier-pjsip-install-method/) or it can be installed from the project itself or package if the distro has one |
| 14:10.34 | Chainsaw | Hello71: I don't see the bundling option working very well, so I suppose we better add back the PJSIP library package that Asterisk previously conflicted with. |
| 14:10.48 | Chainsaw | Hello71: Bug report please and I'll get it sorted. |
| 14:13.41 | Chainsaw | Hello71: Because from the description --with-pjproject-bundled is going to download in the src_compile stage. We can't have that. |
| 14:23.06 | *** join/#asterisk Echo6 (~Thunderbi@64.136.247.50) |
| 14:24.16 | *** join/#asterisk Echo6 (~Thunderbi@64.136.247.50) |
| 14:26.36 | *** join/#asterisk Echo6 (~Thunderbi@64.136.247.50) |
| 14:29.10 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:29.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:32.31 | *** join/#asterisk xdexter (~xdexter@189.7.9.21) |
| 14:32.33 | *** join/#asterisk kharwell (kharwell@nat/digium/x-wzwogvlhktrrynig) |
| 14:44.15 | *** join/#asterisk raspberrypifan (~raspberry@2604:2000:6016:be00:3c0c:ba48:3738:b6d5) |
| 15:02.35 | *** join/#asterisk KerioMorgan (~Adium@gw-us.kerio.com) |
| 15:05.23 | *** join/#asterisk KerioMorgan (~Adium@us.kerio.com) |
| 15:11.39 | *** join/#asterisk Qwell (~north@asterisk/developer/Qwell) |
| 15:11.40 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:17.47 | *** join/#asterisk elyob (~elyob@host86-143-224-192.range86-143.btcentralplus.com) |
| 15:35.06 | infernix | when calling Dial(SIP/foo,XX) the timeout specified by XX starts to count immediately after issuing the command |
| 15:36.13 | infernix | it takes a random amount of time after "SIP 355 Status: 100 Trying" is followed by "SIP 393 Status: 180 Ringing" when looking at wireshark |
| 15:36.39 | infernix | is there some way to start a timeout after the first Ringing, or even after "SIP/SDP 694 Status: 183 Session Progress" ? |
| 15:37.08 | [TK]D-Fender | no |
| 15:40.16 | infernix | meh. it takes around 5-10 seconds from "originate" to the first invite, then it takes around 5-10 seconds before the first Ringing, and the Dial timeout starts as soon as originate is getting called |
| 15:40.36 | infernix | making it hard to try to cut off the call right before a cellphone voicemail service is triggered |
| 15:41.43 | infernix | i can't think of any way to differentiate between a cellphone that gets answered or one that goes to voicemail either |
| 15:43.49 | infernix | how can I debug why it takes so long from "originate" to Invite though? sip set debug on doesn't show anything happening for that duration |
| 15:44.25 | [TK]D-Fender | First suspect : DNS lookups |
| 15:44.33 | [TK]D-Fender | including DNS-SRV |
| 15:45.13 | infernix | even consecutively within a minute? |
| 15:46.11 | infernix | would dnsmgr help? |
| 15:46.12 | [TK]D-Fender | That I can't answer... |
| 15:46.53 | *** join/#asterisk alex999 (~alex999@host172-102-static.15-188-b.business.telecomitalia.it) |
| 15:48.09 | file | can also be a problem if your local hostname can't be resolved |
| 15:49.16 | infernix | they should be, but not sure how to verify. no level of debugging that i turn on is showing anything like that, so let me strace it |
| 15:50.03 | infernix | ok, bad plan, way too much data |
| 15:51.52 | infernix | doh. a wait somewhere. user error :> |
| 15:54.55 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 15:55.45 | file | just Wait(), never leave me alone with a Hangup() |
| 15:59.16 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 16:08.15 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 16:09.37 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 16:14.42 | *** join/#asterisk asdfergtehgwer (6da503a7@gateway/web/freenode/ip.109.165.3.167) |
| 16:15.53 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 16:18.33 | asdfergtehgwer | Hi. Does any body know why asterisk 11.21.2 recreates long inbound transfered calls? Like this: after an incoming call is answered, it is put into queue then is connected to a static agent. Agent transfers the call to another agent, and this call after some time (~11m) is suddenly stopped and, judging by logs, is created again. So the caller isn't hangupped, he just hears the queue again then agent, etc. |
| 16:20.46 | asdfergtehgwer | FreePBX 12.0.76.2, AsteriskNOW. |
| 16:21.53 | asdfergtehgwer | What can I do to debug a call that gets recreated? A physical call that is split into two by asterisk/ |
| 16:23.00 | asdfergtehgwer | These calls have different identifiers, in logs they are two different calls, yet it's one person calling, no transfers or anything during recreation. |
| 16:27.40 | *** join/#asterisk miralin (~Thunderbi@194.8.128.120) |
| 16:28.40 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-hsbmfvvesbxhpsbx) |
| 16:32.28 | *** join/#asterisk lnxslck (~lnxslck@av3-84-90-162-98.netvisao.pt) |
| 16:39.53 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 16:47.11 | *** join/#asterisk CeBe (~CeBe@a81-14-235-90.net-htp.de) |
| 17:06.10 | *** join/#asterisk jiremek (4e7e41e5@gateway/web/freenode/session) |
| 17:06.13 | jiremek | Hello. |
| 17:06.25 | jiremek | need help about realtime config for pjsip. |
| 17:06.30 | jiremek | Someone could help me ? |
| 17:07.53 | *** join/#asterisk amonk (~amonk@unaffiliated/amonk) |
| 17:08.12 | file | what's your question? |
| 17:10.48 | jiremek | The realtime configuration doesn't work for identify by username |
| 17:11.01 | *** join/#asterisk Arkanos-Morph (~Arkanos-M@61.12.72.138) |
| 17:11.03 | jiremek | but I think I've done the right config ! |
| 17:11.22 | jiremek | then the result is a log error like: Unable to find an endpoint to qualify contact |
| 17:11.44 | jiremek | and pjsip show identifies say "No objects found." |
| 17:13.44 | newtonr | jiremek, pastebin a log showing the errors and the SIP trace |
| 17:14.56 | jiremek | First: the config: http://pastebin.com/raw/VauFH3Ak |
| 17:15.51 | *** join/#asterisk raspberrypifan (~raspberry@2604:2000:6016:be00:3c0c:ba48:3738:b6d5) |
| 17:16.06 | jiremek | 2nd, the log: res_pjsip/pjsip_options.c:343 qualify_contact: Unable to find an endpoint to qualify contact sip:jiremek2@ippi.fr:5060 |
| 17:25.40 | jiremek | no idea ? |
| 17:33.59 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 17:39.36 | *** join/#asterisk gusto (~gusto@77.47.94.250.dynamic.cablesurf.de) |
| 18:07.43 | *** join/#asterisk CeBe1 (~CeBe@a81-14-235-90.net-htp.de) |
| 18:14.23 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 18:21.55 | *** join/#asterisk kharwell (kharwell@nat/digium/session) |
| 18:21.56 | jiremek | Hello, Anyone could help me about pjsip identify and realtime ? |
| 18:24.14 | *** part/#asterisk post-factum (~post-fact@vulcan.natalenko.name) |
| 18:32.49 | *** join/#asterisk klow (~textual@c-73-83-127-32.hsd1.wa.comcast.net) |
| 18:36.22 | newtonr | jiremek, you pasted the config, but you didn't paste a log file with SIP trace |
| 18:36.51 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 18:42.07 | *** join/#asterisk kharwell (kharwell@nat/digium/x-xqhymfbjxizotqvp) |
| 18:49.11 | *** join/#asterisk KerioMorgan (~Adium@gw-us.kerio.com) |
| 18:54.36 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:11.49 | *** join/#asterisk mskenderian (uid158360@gateway/web/irccloud.com/x-djjjacvixkcxirwa) |
| 19:14.21 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 19:17.10 | *** join/#asterisk RadicalDev (~jfindley@104-181-196-33.lightspeed.hstntx.sbcglobal.net) |
| 19:17.59 | RadicalDev | Anyone know how to get a switchvox to use the connection information in the SDP? |
| 19:25.09 | *** join/#asterisk swimmercol (~lancelot@B9255616.rev.sefiber.dk) |
| 19:29.12 | infernix | can I check for subscriber presence before initiating a Dial(SIP/foo)? e.g. prevent messages like [May 10 21:28:05] WARNING[22208][C-00000014] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)? |
| 19:29.50 | infernix | it's a locally registered sip client for what it's worth |
| 19:32.52 | newtonr | RadicalDev, most people here are working with Asterisk directly or low-level Asterisk issues. Here is the Switchvox Q&A site: http://support.digium.com/Answers#!/feedtype=RECENT_REPLY&dc=Switchvox&criteria=ALLQUESTIONS |
| 19:34.03 | [TK]D-Fender | infernix, "core show function DEVICE_STATE" |
| 19:37.09 | RadicalDev | newtonr: Yeah =/ I normally do, too. Figured I'd ask, though. |
| 20:01.15 | *** join/#asterisk karelk (~karel@31.10.159.144) |
| 20:14.03 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 20:14.23 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 20:14.41 | *** join/#asterisk mxmasster (~mxmasster@207.171.12.148) |
| 20:26.40 | *** part/#asterisk mxmasster (~mxmasster@207.171.12.148) |
| 20:37.34 | infernix | so i have a bit of a headscratcher here |
| 20:38.19 | infernix | if I set up a Monitor in an extension within the context of an incoming call and then follow it with a Dial(Local/MyDeskPhone@stations), it works fine |
| 20:39.06 | infernix | but if I remove that Monitor from that incoming call context, and move it into the MyDeskPhone extension in stations context - it will record a 0 second long wav file. it basically seems to stop recording immediately |
| 20:45.10 | *** join/#asterisk adeel (~adeel@2602:ffc1:1:face:194b:9864:fbcd:3f5a) |
| 20:52.48 | infernix | i'm essentially trying to record calls only when they are picked up on a local sip desk phone and not when carrier cell phone is picking up. so an incoming call gets a Dial(Local/MyDeskPhone&Local/MyCellPhone) and i only want to activate Monitor if deskphone picks up |
| 20:53.49 | infernix | can't seem to get it done with Monitor inside the MyDeskPhone extension, only in the incoming call extension itself *before* the Dial(Local/MyDeskPhone&Local/MyCellPhone) |
| 20:55.51 | infernix | and I know it's not being bridged directly, wireshark shows packets from deskphone to asterisk server only - so it's not that |
| 20:59.59 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:00.20 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 21:13.31 | *** join/#asterisk swimmercol (~lancelot@B9255616.rev.sefiber.dk) |
| 21:18.01 | infernix | huh.that's weird. MixMonitor does record it |
| 21:18.06 | infernix | ok then, time to rewrite |
| 21:34.43 | dan_j | What character terminates the end of each line that the AMI sends? |
| 21:34.49 | dan_j | It doesnt seem to be \n |
| 21:54.37 | *** join/#asterisk Arkanos-Morph (~Arkanos-M@182.73.228.227) |
| 21:54.43 | *** join/#asterisk simplydrew_ (~simplydre@unaffiliated/simplydrew) |
| 22:16.06 | *** join/#asterisk saxa (~sasa@2-234-206-126.ip224.fastwebnet.it) |
| 22:17.16 | *** part/#asterisk saxa (~sasa@2-234-206-126.ip224.fastwebnet.it) |
| 22:19.55 | *** join/#asterisk klow (~textual@50.141.109.29) |
| 23:06.40 | *** join/#asterisk V5_12R\X (~V5_12R\X@c-24-9-79-75.hsd1.co.comcast.net) |
| 23:07.06 | V5_12R\X | Is # a special extension? I'm getting an "already in use" warning |
| 23:08.10 | *** part/#asterisk kharwell (kharwell@nat/digium/x-xqhymfbjxizotqvp) |
| 23:08.38 | WIMPy | no |
| 23:10.29 | V5_12R\X | So I should be able to use it as an extension, correct? |
| 23:10.39 | WIMPy | yes |
| 23:10.57 | V5_12R\X | Interesting. Well that doesnt appear to work. |
| 23:11.01 | WIMPy | If your phone plays along. |
| 23:11.22 | V5_12R\X | yeah see * is refusing to register that extension, saying it's in use. |
| 23:12.51 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 23:15.41 | [TK]D-Fender | tip: Stop calling devices "extensions". |
| 23:15.51 | [TK]D-Fender | extension is a number you dial from a phone |
| 23:15.59 | [TK]D-Fender | it is not a username or account or section name. |
| 23:17.58 | V5_12R\X | [TK]D-Fender Tip: I was referring to a dialplan extension, and the error message was "Unable to register extension", so the terminology is correct. |
| 23:18.38 | WIMPy | Look at your dialplan where it's defined. |
| 23:19.10 | V5_12R\X | WIMPy nothing else in the context that could be causing an issue that I can see |
| 23:19.31 | WIMPy | What about includeS? |
| 23:22.04 | V5_12R\X | http://pastebin.com/SdrT5MQd |
| 23:22.11 | V5_12R\X | its about as simple as it gets |
| 23:23.22 | V5_12R\X | WIMPy but yeah, you answered my question. I'll put it up on the bench here in a little while and test it out, see if I'm missing something or its a bug, file a report, etc. |
| 23:23.58 | V5_12R\X | I just couldnt find anything on the ol Google there and wanted to make sure I wasn't nuts |
| 23:24.02 | [TK]D-Fender | Show us the full load |
| 23:26.08 | V5_12R\X | And there it is... ok nevermind they're doing some other stupid stuff. |
| 23:27.12 | WIMPy | There is what? And who is they? |
| 23:29.43 | V5_12R\X | It was an include, and they is the customer |
| 23:32.03 | V5_12R\X | but like I said, just wanted to make sure that I wasnt nuts |
| 23:33.24 | WIMPy | can't help with that one. |
| 23:33.43 | V5_12R\X | lol |
| 23:36.39 | [TK]D-Fender | Just because you found it ... doesn't mean you're not still nuts. |
| 23:36.49 | [TK]D-Fender | Further testing seems to be called for... |
| 23:38.47 | V5_12R\X | While I am most certainly nuts, I was unaware my occlusion extended to the obvious. |
| 23:39.52 | [TK]D-Fender | Correlation at best |
| 23:40.02 | *** join/#asterisk Arkanos-Morph (~Arkanos-M@182.73.228.227) |
| 23:46.37 | *** join/#asterisk karelk (~karel@31.10.159.144) |
| 23:50.48 | *** join/#asterisk charolastra (~quassel@91-115-90-140.adsl.highway.telekom.at) |
| 23:51.39 | *** join/#asterisk CunningPike (~CunningPi@70-234-244-146.lightspeed.gdrpmi.sbcglobal.net) |
| 23:52.26 | charolastra | hi, excuse my ignorance but how do i dial out? is dialing +436801234567 enough or is a leading 0 or * required? |
| 23:52.58 | [TK]D-Fender | Depends what you are dialing out to. |
| 23:53.08 | [TK]D-Fender | that other side has expectations on the format you are to dial |
| 23:53.14 | [TK]D-Fender | We can't answer what that is for you |
| 23:59.29 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |