| 00:00.16 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:06.26 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-bctzureegkxkwesb) |
| 00:09.55 | *** join/#asterisk Kerberos76 (~Kerberos7@107.182.184.234.16clouds.com) |
| 00:14.22 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-sbmgjuvcbgkbpyxm) |
| 00:57.00 | sweets_ | is it possible to do 2 actions in 1 ExecIf statement? e.g. ExecIf( $['${FOO}' = 'bar']?Playback(success)&SayAlpha(${beta}):Playback(fubar)&Set(yousuck=1)) ? |
| 00:58.15 | WIMPy | no |
| 00:58.33 | sweets_ | ok, that's what i thought |
| 01:12.08 | *** join/#asterisk LemensTS (~Matt@70.238.131.133) |
| 01:13.41 | LemensTS | If I have a Polycom 550 4 line phone, does that mean a sip call can come in and I hit hold, and it go to line 1, then another one comes in I hit hold and it goes to line 2, etc etc until line 4 is on hold too? |
| 01:16.23 | *** join/#asterisk gushi (~danm@prime.gushi.org) |
| 01:17.12 | gushi | Hey all -- I'm trying to redirect a caller from a menu using dial(), where dial() is calling the sip ID of a peer, but it's claiming it's not found, and treats it as a host to look up. |
| 01:17.19 | gushi | What am I doing wrong? |
| 01:18.06 | gushi | Dial("SIP/DanMahoney_950-00001c57", "SIP/MichaelMcnally_home,14,t") in new stack <-- this tries to look up a hostname called MichealMcnally_home. |
| 01:18.16 | WIMPy | Have you checked with 'sip show peers' or 'sip show peer <name>' that it's correct? |
| 01:19.02 | gushi | MichaelMcNally_home/Micha is what it shows in sip.conf |
| 01:19.38 | gushi | but that's just a column cutoff. It does show that it's a registered peer. |
| 01:19.44 | gushi | and "OK" |
| 01:20.33 | WIMPy | Hmm. I've never thought about peer names possibly being case sensitive, but it's "n" vs "N". |
| 01:22.37 | gushi | ...curse the irish (noting my last name is Mahoney) :) |
| 01:31.28 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:31.31 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 01:33.53 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 01:39.29 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 01:46.02 | Xadof | sweets_: just in case you need help structuring the dialplan to exec multiple commands on that condition, it would be something like: same => n,GotoIf( $[ "${FOO}" = "bar" ]?foo_is_bar:foo_not_bar) \ same => n(foo_is_bar),Playback(success) \ same => n,SayAlpha(${beta}) \ same => n,Goto(foo_bar_check_done) \ same => n(foo_not_bar),Set(yousuck=1) \ same => n(foo_bar_check_done),NoOp(or |
| 01:46.02 | Xadof | put the next shared cmd here) |
| 01:49.08 | *** join/#asterisk saint_ (~saint_@gateway/vpn/privateinternetaccess/saint/x-59675695) |
| 01:51.53 | *** part/#asterisk LemensTS (~Matt@70.238.131.133) |
| 01:53.39 | *** join/#asterisk hyperized (~Hyper@unaffiliated/hyperized) |
| 01:54.50 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 01:58.34 | Xadof | gushi: depending on how the phone is configured (or possibly on the phone's capabilities) it could do either of those. Usually it seems that SIP phones unnecessarily use one line per registration context, which would mean that you would have to have one sip account per extension. However, the phone should be able to multiplex one registration context without a problem, if the device |
| 01:58.34 | Xadof | manufacturers were intelligent enough to do so. I've little faith in device manufacturers so I imagine that it will not do that. |
| 01:58.56 | Xadof | oops, that was directed at LemensTS |
| 02:07.39 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 02:19.27 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 02:19.36 | *** join/#asterisk D30 (~D30@58.71.19.178) |
| 02:23.09 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-58-46-12.lsanca.fios.verizon.net) |
| 02:26.21 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 02:33.55 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:38.48 | Xadof | From what I gather of Ice, if one negotiating party is undisputably the controlling party, then as soon as that party confirms a binding on all streams that are in attempt to be bound (rfc 5245 "until at least one valid candidate pair for each media stream is found"), the SDP proceeds (in the case of webrtc, to dtls handshake). If this is correct, I imagine I could break protocol and force |
| 02:38.48 | Xadof | asterisk to start_ice as controlling + force the tie-breaker to (2**64-1) so that it would (effectively) always win any dispute. The goal here, is to avoid remote webrtc user agents (chrome) that are behind a NAT from waiting for binding requests back from interfaces that will not respond (ie a blackberry network interface I troubleshot earlier today). Does anybody know if this functionality |
| 02:38.48 | Xadof | "should" work, in theory, and/or what side effects I might experience in terms of delays when both agents want to fight over the controlling state, assuming both agents follow protocol at that stage aside from these modifications? |
| 02:39.43 | Xadof | None of our asterisk interfaces are NATd, and we are never doing direct media on webrtc agents |
| 03:39.05 | *** join/#asterisk aruntomar (~aruntomar@123.201.76.229) |
| 03:48.19 | *** join/#asterisk aruntomar (~aruntomar@123.201.76.229) |
| 04:00.56 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 04:19.01 | *** join/#asterisk creativx (~ntttt@226.62-97-205.bkkb.no) |
| 04:30.39 | *** join/#asterisk ganbold_ (~Ganbold@173.244.215.173) |
| 04:48.05 | *** join/#asterisk hfp (~hfp@MTRLPQ0736W-LP140-03-1168094354.dsl.bell.ca) |
| 05:06.44 | *** join/#asterisk aruntomar (~aruntomar@123.201.76.229) |
| 05:31.20 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 05:42.40 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 05:48.51 | *** part/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 05:52.03 | *** join/#asterisk tuxd00d (~tuxd00d@ip70-162-151-19.ph.ph.cox.net) |
| 05:56.28 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:00.21 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-dgklfcpcgnfyaroe) |
| 06:15.17 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 06:31.39 | *** join/#asterisk bulkorok (~Benjamin_@85.183.61.47) |
| 06:31.54 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 06:32.46 | *** join/#asterisk demarcoz (~demarcoz@92-108-158-179.dynamic.upc.nl) |
| 06:37.58 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 06:39.37 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 06:57.15 | *** join/#asterisk CeBe (~CeBe@port-92-200-62-22.dynamic.qsc.de) |
| 06:58.52 | *** join/#asterisk Zogot (~Adium@D4B2620A.static.ziggozakelijk.nl) |
| 06:59.38 | *** join/#asterisk Iamnacho (~Iamnacho@ip72-213-61-117.om.om.cox.net) |
| 07:02.54 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 07:09.58 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 07:10.55 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:11.07 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.125.9) |
| 07:25.31 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-56-139.rn.hr.cox.net) |
| 07:26.57 | *** join/#asterisk OurRoyalGabe (~quassel@cpe-104-162-60-254.nyc.res.rr.com) |
| 07:35.06 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 07:41.48 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 07:50.10 | *** join/#asterisk CeBe (~CeBe@wlan-141-23-108-205.tubit.tu-berlin.de) |
| 07:51.58 | *** join/#asterisk CeBe1 (~CeBe@wlan-141-23-108-205.tubit.tu-berlin.de) |
| 08:22.05 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.127.7) |
| 08:26.10 | *** join/#asterisk Cust0sL1men (~CustosLim@unaffiliated/cust0slim3n) |
| 08:30.37 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 08:37.24 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 08:39.02 | *** part/#asterisk c0rnoTa (~c0rnoTa@109.188.127.7) |
| 08:47.42 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:48.18 | *** join/#asterisk ketas (~ketas@97-39-190-90.dyn.estpak.ee) |
| 08:49.57 | *** join/#asterisk sommarnatt (~sommarnat@kungsbacka.oderland.com) |
| 08:55.06 | *** join/#asterisk ketas (~ketas@97-39-190-90.dyn.estpak.ee) |
| 08:57.14 | *** join/#asterisk karamell (~maxi@2001:470:1f1b:1b7::ffff) |
| 08:58.42 | *** join/#asterisk karamell (~karamell@2001:470:1f1b:1b7::ffff) |
| 09:14.37 | *** join/#asterisk petris (sid19918@gateway/web/irccloud.com/x-fuxkizmgmphajhkf) |
| 09:15.39 | *** join/#asterisk sekil (~sekil@109-92-28-226.static.isp.telekom.rs) |
| 09:21.27 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:42.05 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 10:04.44 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:c4d7:3d8c:da0e:9074) |
| 10:21.00 | *** join/#asterisk fergus (~sergei@mail.55h.by) |
| 10:27.19 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 10:29.19 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 10:44.27 | *** join/#asterisk CeBe (~CeBe@port-92-200-62-22.dynamic.qsc.de) |
| 10:44.52 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 10:53.56 | *** join/#asterisk LooserOuting (~LooserOut@xd9bdba0b.dyn.telefonica.de) |
| 11:03.03 | LooserOuting | Hello. I need some advice. I want to read the P-Preferred-Identity Header in an incoming invite with the function SIP_HEADER() and save it in a variable using SET() |
| 11:03.41 | LooserOuting | The probles is that the PPI contains a plus |
| 11:04.05 | LooserOuting | and i get a sytax error because of this |
| 11:04.31 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 11:05.30 | LooserOuting | how can I store the PPI in a variable ? |
| 11:06.46 | *** join/#asterisk CeBe (~CeBe@port-92-200-62-22.dynamic.qsc.de) |
| 11:10.12 | *** join/#asterisk Draecos (~Draecos@106-68-156-144.dyn.iinet.net.au) |
| 11:11.41 | LooserOuting | And does asterisk have functions to extract displayname user domain params from a value ? |
| 11:21.21 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:25.36 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 11:45.53 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 11:46.14 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 11:46.20 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-62-22.dynamic.qsc.de) |
| 11:47.51 | *** join/#asterisk eschmidbauer (~eschmidba@unaffiliated/eschmidbauer) |
| 11:49.40 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:53.17 | *** join/#asterisk bulkorok (~Benjamin_@85.183.61.47) |
| 11:59.35 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:00.35 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 12:00.44 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 12:00.44 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:04.16 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 12:29.02 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:35.10 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 12:39.15 | *** join/#asterisk jo_sh_ua (~joshua@unaffiliated/jo-sh-ua/x-0543518) |
| 12:46.44 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-gmgckbnirpdncqkk) |
| 12:48.43 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 12:54.31 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 12:57.06 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:02.05 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:02.49 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-eniazvosfxuhrrfk) |
| 13:08.58 | *** join/#asterisk ghostlines (~ghostline@50-205-197-158-static.hfc.comcastbusiness.net) |
| 13:13.28 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:14.59 | *** join/#asterisk davlefouAMD (~david@197.15.213.2) |
| 13:17.33 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:22.15 | *** join/#asterisk lorsungcu_ (uid65806@gateway/web/irccloud.com/x-ayholppsvbfdvkix) |
| 13:22.29 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 13:22.46 | *** join/#asterisk rexwin_ (~rexwin@106.208.134.183) |
| 13:23.02 | rexwin_ | j #windows-server |
| 13:25.36 | eschmidbauer | you're still in #asterisk |
| 13:26.25 | [TK]D-Fender | hasn't seen a good "Windows Shaming" in a while... |
| 13:26.37 | [TK]D-Fender | wonders if there is still a collar for that.... |
| 13:28.39 | eschmidbauer | nows your chance |
| 13:28.55 | eschmidbauer | Does asterisk even run on windows? |
| 13:29.42 | [TK]D-Fender | "sorta". But asking any further will lead to more than just "shaming". |
| 13:32.58 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:35.20 | *** part/#asterisk italorossi (~Adium@187.60.66.11) |
| 13:36.40 | *** join/#asterisk Leseratte (~florian@2001:470:70bc::1) |
| 13:38.12 | *** join/#asterisk ghostlin_ (~ghostline@50-205-197-158-static.hfc.comcastbusiness.net) |
| 13:45.29 | *** join/#asterisk aross42 (~aross@dsl-eclipsefnd.storm.ca) |
| 13:45.44 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:46.50 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:51.33 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 13:51.45 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:51.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:51.53 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-awgwumjzedulaagb) |
| 13:51.53 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:54.18 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:58.45 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 14:08.40 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:10.23 | *** join/#asterisk linocisco (~linocisco@111.84.193.66) |
| 14:18.42 | linocisco | has anybody used Grandstream GXP1405 with asterisk? |
| 14:18.49 | [TK]D-Fender | ~polls |
| 14:18.52 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 14:19.24 | eschmidbauer | n |
| 14:20.51 | linocisco | any granstream phone with asterisk? because I am setting up NTP server on asterisk local server and all phone should get time from local server |
| 14:21.04 | *** join/#asterisk kharwell (kharwell@nat/digium/x-lclkrhorwketuced) |
| 14:22.15 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:23.18 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-bnwypyjkzekmauvv) |
| 14:24.12 | [TK]D-Fender | linocisco: Asterisk has nothing to do with this |
| 14:24.27 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 14:24.37 | [TK]D-Fender | linocisco: And if you want help with what you're doing with it... you should actually show what you've done. |
| 14:26.56 | linocisco | [TK]D-Fender, ok. I have Grandstream.py which is used for provisioning. but I dont why I can't get correct time on my phone |
| 14:27.16 | [TK]D-Fender | Nobody here know what the script is doing. |
| 14:27.27 | [TK]D-Fender | So far that tells us nothing |
| 14:27.33 | linocisco | [TK]D-Fender, let me get the file |
| 14:29.57 | trurl | linocisco: is your dhcpd announcing the ntp server? |
| 14:30.27 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-tjkeejrpttkxpufb) |
| 14:30.38 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-mxewcyudzukpaoyt) |
| 14:30.39 | linocisco | trurl, thanks for your question. http://pastebin.centos.org/24181/ |
| 14:30.57 | linocisco | trurl, it is my ntpd server |
| 14:31.24 | [TK]D-Fender | inappropriate address 192.168.1.8 for the fudge ... <--- that doesn't look good |
| 14:31.32 | [TK]D-Fender | Next show us the phone is configured to even look at it |
| 14:32.54 | *** join/#asterisk anonymouz666 (~anonymouz@201-29-71-247.user.veloxzone.com.br) |
| 14:35.04 | trurl | linocisco: a) i'd switch to chrony b) are you using dhcpv4 and or v6 and are those sending the ntp server address in field 42 (option ntp) |
| 14:35.42 | linocisco | trurl, how can I check? |
| 14:36.38 | trurl | have you setup an dhcpd? |
| 14:36.41 | [TK]D-Fender | a) prove the phone was ever told to use your NTP server. b) that you aren't blocking it with your firewall. c) that it's responding with the right time from another client. |
| 14:36.50 | [TK]D-Fender | Stop looking for DHCP |
| 14:36.56 | [TK]D-Fender | Prove the phone was TOLD to use it |
| 14:37.07 | [TK]D-Fender | Or where it is expecting to find out about it from |
| 14:37.26 | [TK]D-Fender | Just randomly looking elsewhere is not the way to start this process |
| 14:38.18 | [TK]D-Fender | PROVE the phone was either explicitly told where to go for this, or that it defaults to accepting it from somewhere else; and then verify that that source is properly advertising it. |
| 14:38.26 | [TK]D-Fender | Then prove if the request even makes it to your server |
| 14:39.09 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:39.41 | linocisco | [TK]D-Fender, I am uploading to imagebin |
| 14:40.06 | linocisco | [TK]D-Fender, http://ibin.co/1yQQyRqGxqwW |
| 14:40.28 | [TK]D-Fender | [10:31][TK]D-Fenderinappropriate address 192.168.1.8 for the fudge ... <--- that doesn't look good |
| 14:40.39 | *** join/#asterisk YasT_ (~YasT_@247.Red-213-98-81.staticIP.rima-tde.net) |
| 14:40.59 | linocisco | [TK]D-Fender, http://ibin.co/1yQRd2JsFk8r |
| 14:41.36 | linocisco | [TK]D-Fender, the thing is after phone is powered offf and |
| 14:41.55 | anonymouz666 | sip set debug peer [TK]D-Fender |
| 14:42.06 | *** join/#asterisk russellb (~russellb@redhat/russellb) |
| 14:42.06 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:42.33 | linocisco | [TK]D-Fender, powered on again. date is correct and but time is not correct and whenever I check time on phone via webpage, time zone is changed to automatically and only after I changed it back to my time zone, it is correct |
| 14:45.07 | *** join/#asterisk mjordan (mjordan@nat/digium/x-lstpscrsgcjajuur) |
| 14:45.07 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:45.48 | [TK]D-Fender | Then that's a phone issue |
| 14:46.03 | [TK]D-Fender | Maybe it is doing something like taking a TZ from DHCP. |
| 14:46.09 | [TK]D-Fender | Go read their admin guide |
| 14:48.21 | linocisco | [TK]D-Fender, http://pastebin.centos.org/24301/ |
| 14:49.16 | linocisco | [TK]D-Fender, it is grandstream.py used for provisioning. but whenever I changed on the line "self._timeZone = 'auto'" from auto to something else. it is not effected |
| 14:49.21 | [TK]D-Fender | <PROTECTED> |
| 14:50.56 | linocisco | [TK]D-Fender, what does that mean? |
| 14:51.09 | [TK]D-Fender | it means your provisioning is setting a TZ |
| 14:51.22 | [TK]D-Fender | and anything you set manually I expect will get overridden |
| 14:51.40 | [TK]D-Fender | (any time it re-reads those settings) |
| 14:52.51 | linocisco | [TK]D-Fender, so what should I do? could you please correct me? my time zone is GMT+6:30 |
| 14:53.13 | [TK]D-Fender | Look what it is actually pulling there.... |
| 14:54.11 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 14:55.32 | trurl | myanmar or cocos islands? (just curious :D) |
| 14:57.07 | linocisco | trurl, myanmar |
| 14:59.09 | *** part/#asterisk rexwin_ (~rexwin@106.208.134.183) |
| 15:12.35 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:17.56 | *** join/#asterisk ghostlines (~ghostline@50-205-197-158-static.hfc.comcastbusiness.net) |
| 15:19.13 | *** join/#asterisk ghostlines (~ghostline@50-205-197-158-static.hfc.comcastbusiness.net) |
| 15:27.44 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-rbjmdgmftlogsnsu) |
| 15:28.15 | anonymouz666 | Nokia just bought Alcatel-Lucent for $16.6 billion |
| 15:30.03 | malcolmd | cha-ching |
| 15:33.50 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 15:34.38 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:36.51 | *** join/#asterisk russellb (~russellb@redhat/russellb) |
| 15:36.51 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:38.57 | *** join/#asterisk saint_ (~saint_@c-71-59-68-185.hsd1.nj.comcast.net) |
| 15:42.55 | *** join/#asterisk saint_ (~saint_@gateway/vpn/privateinternetaccess/saint/x-59675695) |
| 15:48.06 | PHunter | Has anyone successfully moved the default spool directory? I have mine changed at it appears to use default /var/spool/asterisk anyways.. even after hard restart |
| 15:54.24 | [TK]D-Fender | What do you palify as having "moved" it? |
| 15:54.29 | [TK]D-Fender | qualify* |
| 15:59.54 | PHunter | asterisk.conf, spooldir changed |
| 16:00.07 | PHunter | storing voicemails and all spool info to another location |
| 16:00.21 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 16:00.24 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 16:06.01 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 16:06.03 | *** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com) |
| 16:08.20 | *** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com) |
| 16:13.40 | *** join/#asterisk [Outcast] (outcast@nat/cisco/x-bqfqxbhgeqsrgryz) |
| 16:14.47 | *** join/#asterisk [Outcast] (outcast@nat/cisco/x-zbgqkznzsanlbnzl) |
| 16:16.06 | *** join/#asterisk [Outcast] (outcast@nat/cisco/x-slytkaifashvsawv) |
| 16:19.56 | *** join/#asterisk linocisco (~linocisco@111.84.193.117) |
| 16:21.59 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 16:28.55 | [TK]D-Fender | PHunterasterisk.conf, spooldir changed <- show us the full config |
| 16:48.47 | voip | morning |
| 16:55.50 | *** join/#asterisk bmg505 (~leon@196-215-75-98.dynamic.isadsl.co.za) |
| 17:14.41 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 17:17.05 | *** join/#asterisk c|oneman (cloneman@1337.montrealdark.com) |
| 17:18.09 | c|oneman | hey, this has nothing to do with asterisk, but are you guys aware of limitaitons when forwarding to 1-800 Number for certain countries? I had a problem forwarding to 1-800 for calls from Australian mobile, they would just drop |
| 17:18.51 | c|oneman | e.g. they would get to our ivr fine on a local aussie number, then one of our queues was forwarding to a 1-800, and they would get kicked |
| 17:22.49 | drmessano | 1-800 numbers are typically regional in nature |
| 17:23.41 | drmessano | So they are subject to not working or being routed in an unexpected way from a specific carrier |
| 17:25.02 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:26.00 | PHunter | [TK]D-Fender: I read somewhere else that it wont grab it and was easier to link the directories. So we did that and it appears to be working fine. |
| 17:26.20 | [TK]D-Fender | Show the config anyway.... |
| 17:26.58 | [TK]D-Fender | (the entire file) |
| 17:31.49 | PHunter | http://pastebin.com/UtZ03dqy |
| 17:32.22 | file | remove (!) from [directories] |
| 17:32.37 | file | that would make it work |
| 17:33.47 | [TK]D-Fender | (!) = ignore any actual changes... or contents below |
| 17:34.31 | PHunter | Im sorry, but that sounds like a useless feature.. thanks for the info. |
| 17:35.22 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:35.38 | [TK]D-Fender | IMO it is .... made a good bit worse by not actually being documented as such in the sample config. |
| 17:36.07 | file | it's not a useless feature because that's not exactly what it does |
| 17:36.20 | file | it's used for templates |
| 17:36.20 | PHunter | Well in this case it did. |
| 17:36.32 | PHunter | Well, it locks a section of configs.. |
| 17:37.06 | PHunter | I went to Asterisk Advanced in Vegas and learned the ups and downs and have NEVER seen _any_ reference to (!) |
| 17:37.33 | PHunter | Maybe thats in the super-ultra-advance class post DCAP |
| 17:39.38 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 17:39.56 | file | the reason why it's done that way is because the values in there may or may not be the default, depending on how Asterisk has been built - and noone has, as of yet, made the asterisk.conf file created at build time |
| 17:42.15 | file | (referring to directories in asterisk.conf) |
| 17:55.06 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 17:56.50 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 18:03.46 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 18:10.04 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.127.7) |
| 18:10.40 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:12.20 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 18:12.51 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 18:23.21 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 18:30.14 | *** join/#asterisk KungFuJesus (~adam@72.49.188.141) |
| 18:30.33 | KungFuJesus | Hello, I was in here a few days ago debugging possible IRQ misses causing issues |
| 18:30.47 | KungFuJesus | only time will tell if what I did fixed the problem but that seemingly is in order, at least for now |
| 18:30.51 | *** join/#asterisk CustosLimen (~CustosLim@unaffiliated/cust0slim3n) |
| 18:31.09 | KungFuJesus | I have another concern that is very strange, my TE134 call gives inconsistently ringback for inbound calls |
| 18:31.20 | KungFuJesus | e.g. sometimes it will work, sometimes it won't |
| 18:31.55 | KungFuJesus | is there anyway to diagnose this, or is this some sort of known issue with a workaround? I've just about exhausted my google-fu |
| 18:34.13 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:42.23 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-rmueucypfjrnefjw) |
| 18:47.11 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-jjejpapfplwqcexi) |
| 18:48.14 | *** join/#asterisk sofltech (~sofltech@c-174-61-92-126.hsd1.fl.comcast.net) |
| 18:48.49 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-62-22.dynamic.qsc.de) |
| 18:51.46 | sofltech | Hi all, I'm getting a message on a Dial command "exited non-zero" (from an Originate request). The problem is that the next line in the dialplan is not executed. At first, I thought it was an error, but its a very similar issue to what is described here: http://forums.digium.com/viewtopic.php?t=72845 |
| 18:55.27 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-gcurhlnzveaunjpo) |
| 18:59.38 | *** part/#asterisk italorossi (~Adium@187.60.66.11) |
| 19:07.34 | *** join/#asterisk DeeJayTwo (~DeeJayTwo@unaffiliated/deejaytwo) |
| 19:07.35 | DeeJayTwo | Hi |
| 19:07.52 | DeeJayTwo | I just hacked a bit into app_voicemail.c to fix a socket stuck in CLOSE_WAIT issue... |
| 19:08.00 | DeeJayTwo | I compile it and load it as module |
| 19:08.10 | DeeJayTwo | Then...noerror and just see asterisk stopping... |
| 19:08.45 | DeeJayTwo | I tried to add ast_debug lines in load_module(void) but never see them... |
| 19:08.49 | DeeJayTwo | any idea? |
| 19:12.47 | *** join/#asterisk aross42 (~aross@dsl-eclipsefnd.storm.ca) |
| 19:20.11 | *** join/#asterisk klausskelvin (~klausskel@179.154.130.103) |
| 19:20.32 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 19:31.25 | sofltech | I'm getting an "exited non-zero" message for a dial call and it seems like the line following the dial command is not executing once that happens. |
| 19:36.25 | *** join/#asterisk ipengineer_ (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:38.28 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 19:39.34 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:54.17 | Xadof | DeeJayTwo: it sounds like you must have a typo or that your fix wasn't quite a fix. Sharing your diff on pastebin may help us to provide you feedback. If there's an open issue on jira for what you are trying to patch, then please share the issue# so that we can get better context on your patch. If there isn't an open issue on jira, consider opening one if you think there is an issue, and |
| 19:54.17 | Xadof | you can share your patch there as well. I'll look back later today in case I can help but I'm going to be away for a few hours. |
| 20:01.35 | PHunter | Scenario: I made Asterisk 11 from source, installed, started it, etc. If I want to add an addon, should I stop asterisk, and make install again? Or can I do it without stopping? |
| 20:03.07 | PHunter | in theory, if im not mistaken, i can make install over the top, and just load the module.. |
| 20:03.34 | [TK]D-Fender | Generally, yes |
| 20:03.57 | PHunter | But im not sure how well it will work if the service is running when doing that.. |
| 20:07.12 | [TK]D-Fender | You're adding. |
| 20:07.42 | PHunter | true but those files that are currently there, wont they be attempting to 'overwrite' them? |
| 20:07.43 | [TK]D-Fender | Depends how what you're adding can actually come in CONFLICT with something that is already running. |
| 20:08.09 | PHunter | yeah im trying to decide how much trouble I want to get in. |
| 20:08.19 | [TK]D-Fender | Imagine that it doesn't unload and realod those so that you're still running to olde code... because it is LOADED. |
| 20:08.26 | [TK]D-Fender | But on on a reload will the new cade take |
| 20:08.37 | [TK]D-Fender | code* |
| 20:08.48 | PHunter | yeah its just a cdr_mysql module |
| 20:08.55 | PHunter | only difference, the config is already there |
| 20:09.15 | PHunter | its literally copy the module, load it and im home free. |
| 20:09.25 | PHunter | the rest was already in place. |
| 20:10.41 | PHunter | inb4 "cdr_adaptive_odbc.." |
| 20:15.49 | *** join/#asterisk Taeylan (258469ce@gateway/web/freenode/ip.37.132.105.206) |
| 20:15.56 | Taeylan | Hi |
| 20:17.02 | Taeylan | Been trying to configure Asterisk 13 (PJSIP) on a OpenWrt device for a couple of days and I have hit a roadblock, can please someone help me? |
| 20:18.05 | Taeylan | Apparently everything is well configured and I can receive and emit phone calls but they cut off as soon as someone picks up with the message: bridge.c:776 bridge_base_init: Bridge bc626ace-2153-4b0b-9545-7d7c0fb0281a: Could not create class basic. No technology to support it. |
| 20:18.42 | Taeylan | I have been trying to find if there are any bridge technologies packages to install but I cannot find anything |
| 20:19.22 | Taeylan | Perhaps a configuration entry to enable the bridging technologies |
| 20:19.27 | Taeylan | Any idea? |
| 20:21.09 | *** join/#asterisk steelbrain (~steelbrai@39.35.28.33) |
| 20:21.17 | steelbrain | Good Evening People |
| 20:21.47 | steelbrain | Any idea how to pass variables in the AMI Redirect command? |
| 20:23.01 | [TK]D-Fender | You don't. |
| 20:23.20 | [TK]D-Fender | The reason it isn't in the documentation for it is because it doesn't do that. |
| 20:23.45 | steelbrain | Is it like on the roadmap? |
| 20:23.49 | [TK]D-Fender | 'No |
| 20:23.52 | steelbrain | or is there a workaround or something? |
| 20:24.01 | [TK]D-Fender | There is already a command to set variables on a channel |
| 20:24.22 | [TK]D-Fender | You should read the list |
| 20:24.32 | steelbrain | But wouldn't that mean that I will have to do a wait(5) while my background daemon sets that variable |
| 20:24.47 | steelbrain | and will it automatically make the variable set via AMI available in scope, like instantly? |
| 20:25.21 | [TK]D-Fender | what scope? |
| 20:25.28 | [TK]D-Fender | It's a channel variable.... |
| 20:25.46 | steelbrain | I meant the DialPlan scope |
| 20:25.52 | [TK]D-Fender | before you redirect... go set it |
| 20:25.54 | steelbrain | How do I pass something there |
| 20:26.01 | [TK]D-Fender | There is no "scope" in the dialplan |
| 20:26.23 | steelbrain | ${asdasd} <-- asdasd is looked up in a scope or something and set if it exists or set to null if it doesn't |
| 20:26.46 | steelbrain | Perhaps I am not using the right word for it :P |
| 20:27.11 | [TK]D-Fender | There is no such thing as a scope. Channel variables exist in a channel. That is the only "scope" |
| 20:27.44 | steelbrain | and how does someone access a channel variable :-) I appreciate all your help |
| 20:28.01 | [TK]D-Fender | Who is "someone"? Acces from where? How? |
| 20:28.32 | [TK]D-Fender | You are being too vague and it is leading to questions with no substance to respond to. |
| 20:28.34 | steelbrain | Access from dialplan like [from-internal]\n exten => DIAL(SIP/${SOMECHANVAR}/${EXTEN}) |
| 20:28.54 | [TK]D-Fender | that is a var in the chanel. |
| 20:29.00 | steelbrain | Yayyyyyyyy |
| 20:29.05 | [TK]D-Fender | So go set it via AMI ... then redirect via AMI\ |
| 20:29.15 | steelbrain | One last question, |
| 20:29.33 | [TK]D-Fender | There are 2 separate commands for this. There is precisely zero need to themn to be integrated. |
| 20:29.49 | steelbrain | Oh Which ones? <-- extra question |
| 20:30.20 | [TK]D-Fender | [16:23][TK]D-FenderThere is already a command to set variables on a channel |
| 20:30.22 | [TK]D-Fender | [16:24][TK]D-FenderYou should read the list |
| 20:30.31 | steelbrain | and the last question is, How can I set variable in a channel that is going to be created in a redirect call, I mean a new channel is created in a redirect, right? |
| 20:30.33 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+AMI+Actions |
| 20:30.45 | steelbrain | I already have that opened, but thanks again :) |
| 20:30.51 | [TK]D-Fender | No channel is created in a redirect |
| 20:30.56 | [TK]D-Fender | that *IS* a channel |
| 20:31.06 | steelbrain | Yayyyy! |
| 20:31.10 | [TK]D-Fender | You are just having it stop what it's doing and go somewhere else in the dialplan. |
| 20:31.12 | steelbrain | You rock buddy! |
| 20:31.24 | steelbrain | <3 <3 <3 <3 <3 |
| 20:31.26 | steelbrain | Have a good day |
| 20:31.28 | PHunter | Channel Variables have differences too |
| 20:31.37 | [TK]D-Fender | Unless they're the same! |
| 20:31.39 | PHunter | if you set _VARIABLE |
| 20:31.46 | PHunter | it will be used once in another context |
| 20:31.56 | PHunter | if you move it again I believe its not available |
| 20:31.56 | [TK]D-Fender | NO. |
| 20:32.02 | PHunter | Oh wait |
| 20:32.06 | PHunter | ignore me |
| 20:32.12 | [TK]D-Fender | !context |
| 20:32.17 | PHunter | <-- noob |
| 20:32.23 | PHunter | im thinking something else |
| 20:32.41 | PHunter | context vars or something. |
| 20:32.52 | [TK]D-Fender | Nope, but keep at it... |
| 20:32.56 | PHunter | been a long day. |
| 20:32.58 | PHunter | i give up |
| 20:33.01 | [TK]D-Fender | packs up to head home... |
| 20:33.05 | PHunter | o/ |
| 20:33.08 | [TK]D-Fender | Try again tomorrow then... |
| 20:33.18 | PHunter | Maybe.. I can only do small doses |
| 20:33.33 | PHunter | this phone system implosion is a headache |
| 20:33.38 | Taeylan | Any idea why I do not have any bridge technology showing up in a PJSIP Asterisk 13 installation? |
| 20:35.13 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 20:35.44 | *** join/#asterisk ctpdump (~tcpdump@2602:ffda:da:2:216:3eff:feae:d69c) |
| 20:39.11 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 20:53.18 | klausskelvin | Sorry, first post on IRC, asterisk newbie. What should I change in the configuration for getting the sip debug to be written to disk ( verbose file ) ? -- asterisk version Asterisk 1.8.13.1 |
| 20:59.02 | mjordan | klausskelvin: either on the CLI, enable 'sip set debug on' or set 'sipdebug=yes' in sip.conf |
| 21:06.25 | *** join/#asterisk ghostlin_ (~ghostline@50-205-197-158-static.hfc.comcastbusiness.net) |
| 21:06.38 | klausskelvin | I got it, thanks advance |
| 21:08.08 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 21:09.32 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:16.56 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 21:25.50 | ctpdump | is there a database (csv?) with location codes for the second part of the 10 digit number? eg: 123-XXXX-1234 - to get a more precise location rather than just the big area code defined by the first 3 digits? |
| 21:27.44 | ctpdump | I meant three digits ;) |
| 21:27.55 | ctpdump | eg: 509.497 is Bentoncity, WA |
| 21:35.22 | *** join/#asterisk averythomas (~averythom@2607:5300:60:2d42::1) |
| 21:38.15 | *** join/#asterisk cyford (junkmail@c-73-207-183-115.hsd1.ga.comcast.net) |
| 21:45.08 | wdoekes | ctpdump: my google-fu says your american plan is initially ordered by digits 2..4, not 4..6: http://www.convertit.com/Go/ConvertIt/Reference/Telephone_Area_Codes.ASP |
| 21:45.53 | ctpdump | wdoekes: thanks, I think I found the correct terminology which would be NPA-NXX |
| 21:46.11 | ctpdump | http://www.cnac.ca/co_codes/co_code_status.htm |
| 21:46.17 | ctpdump | I was after the Canadian actually |
| 21:46.22 | ctpdump | and they have an official nice csv file |
| 21:50.12 | wdoekes | didn't know Canadians would assume that everyone was Canadian.. that's new |
| 21:59.48 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 22:06.53 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 22:11.51 | *** part/#asterisk karamell (~karamell@2001:470:1f1b:1b7::ffff) |
| 22:15.13 | *** part/#asterisk mjordan (mjordan@nat/digium/x-lstpscrsgcjajuur) |
| 22:37.29 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 22:40.06 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 22:50.34 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 22:53.19 | *** part/#asterisk kharwell (kharwell@nat/digium/x-lclkrhorwketuced) |
| 22:59.22 | *** join/#asterisk F2Knight (~f2knight@2601:7:2200:3b4:a5c0:9452:c509:b88e) |
| 23:00.55 | F2Knight | Q: I have a high volume call server, we are trying to record all calls. but some of them are being cut off. e.g. a 40 min call may only get the first 2 min or 30 min and just randomly gets cut off. This is only the recordings not the actual call. Any suggestions? |
| 23:08.22 | *** join/#asterisk pEYEd (~kvirc@ool-43561899.dyn.optonline.net) |
| 23:09.21 | pEYEd | how can I initiate SIP from behind a firewall if I don't have the ability to forward a port at the router? |
| 23:15.38 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 23:16.17 | [TK]D-Fender | "initiate SIP" is a bad kind of vague |
| 23:16.51 | [TK]D-Fender | And Initiating from behind a firewall (I presume you just mean **NAT** here) ... is never the issue |
| 23:17.07 | [TK]D-Fender | You're also vague about which end you're referring to |
| 23:25.09 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 23:30.22 | *** join/#asterisk eschmidbauer__ (~eschmidba@unaffiliated/eschmidbauer) |
| 23:35.00 | *** join/#asterisk RackSmurf (~jdaniel@unaffiliated/techsmurf) |
| 23:35.56 | *** join/#asterisk roentgen_ (~none@openvpn/community/support/roentgen) |
| 23:38.03 | *** join/#asterisk Champi_ (Champi@damn.e-leet.be) |
| 23:38.50 | *** join/#asterisk luckman212_ (~luckman21@unaffiliated/luckman212) |
| 23:38.50 | *** join/#asterisk CeBe (~CeBe@port-92-200-62-22.dynamic.qsc.de) |
| 23:40.34 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 23:44.11 | *** join/#asterisk sarlalian (~sarlalian@107.170.239.102) |
| 23:45.51 | *** join/#asterisk sarlalian (~sarlalian@107.170.239.102) |
| 23:49.16 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 23:59.30 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 23:59.37 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |