| 00:00.18 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:27.09 | *** join/#asterisk blee (~blee@72.189.166.247) |
| 00:31.14 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 01:03.25 | *** join/#asterisk fullstop (~fullstop@107.170.53.155) |
| 01:48.42 | *** join/#asterisk aness (~aness@cpe-74-68-148-41.nyc.res.rr.com) |
| 01:52.21 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:32.38 | *** join/#asterisk linuxfool (~james@DHCP-149-228.resnet.ua.edu) |
| 02:43.51 | *** join/#asterisk ahklerner (~ahklerner@unaffiliated/ahklerner) |
| 02:55.29 | *** join/#asterisk D30 (~deo@222.127.13.226) |
| 03:02.44 | *** part/#asterisk ahklerner (~ahklerner@unaffiliated/ahklerner) |
| 03:57.35 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-nzjkgqvewdmlputh) |
| 04:17.14 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 04:28.41 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 04:30.59 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 04:32.51 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 04:48.00 | *** join/#asterisk lkthomas (~lkthomas@n218250150146.netvigator.com) |
| 04:48.02 | lkthomas | hey guys |
| 04:48.04 | lkthomas | I got Span 1: Channel 0/23 got hangup request, cause 102 |
| 04:48.17 | lkthomas | is it problem from telco ? |
| 04:54.10 | WIMPy | {0x66, "Recovery on timer expiry"}, |
| 04:54.44 | WIMPy | But that doesn't tell you where it happened. That's in the location bits. |
| 04:55.39 | *** join/#asterisk aross42 (~aross@h46-190.reznet.ucalgary.ca) |
| 05:11.15 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 05:13.32 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 05:14.06 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:1dd0:221:6aff:feb8:e0b2) |
| 05:15.26 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 05:15.51 | lkthomas | interesting |
| 05:15.59 | lkthomas | if I use music on hold, connection cut in 3 seconds |
| 05:16.13 | *** join/#asterisk oesteve (~oscar@145.Red-88-15-97.dynamicIP.rima-tde.net) |
| 05:25.46 | lkthomas | ahha |
| 05:25.52 | lkthomas | music on hold is extra service |
| 05:26.00 | lkthomas | Telco provider said it cost money |
| 05:27.34 | lkthomas | is it possible to run MOH without notify provider ? |
| 05:29.25 | ChannelZ | They want to charge you for having MOH on your own system? |
| 05:29.30 | ChannelZ | Dump the provider |
| 05:29.30 | lkthomas | yeah |
| 05:29.37 | lkthomas | somehow they could detect I am using MOH |
| 05:30.13 | lkthomas | how could they detect MOH is in use or not |
| 05:35.34 | ChannelZ | Are you using directmedia? |
| 05:35.41 | lkthomas | what's that |
| 05:37.07 | ChannelZ | Well without directmedia, someone calls you, which comes in from your provider SIP -> Asterisk. Then Asterisk makes a call to your device. When you answer, there are 2 connections happening: Provider -> Asterisk and Asterisk -> You |
| 05:37.49 | ChannelZ | With directmedia, if Asterisk doesn't need to listen to the media for any reason, when you answer Asterisk can essentially tell your provider to connect directly to your phone |
| 05:38.30 | lkthomas | provider SIP? |
| 05:38.32 | ChannelZ | I might be not thinking this through correctly, but offhand I think that's the only way the provider could see that you put a call on hold. |
| 05:38.33 | lkthomas | we are using IDAP |
| 05:38.39 | *** join/#asterisk zerick (~eocrospom@190.118.28.252) |
| 05:38.59 | ChannelZ | I have no idea what IDAP is |
| 05:40.02 | lkthomas | or, PRI |
| 05:40.31 | ChannelZ | oh... hmmm |
| 05:40.50 | lkthomas | I don't understand how does PRI provider know I am running MOH, LOL |
| 05:41.29 | ChannelZ | me either |
| 05:41.47 | ChannelZ | Unless they are literally detecting waveforms that sound like music |
| 05:42.09 | lkthomas | it's interesting that no matter what MOH I put, it still cut in 3 seconds |
| 05:42.13 | ChannelZ | What if you play a song in your IVR? Does it hang up then? |
| 05:42.15 | lkthomas | unless I use default ring |
| 05:45.55 | *** join/#asterisk JeroenBo (~JeroenBo@541BE122.cm-5-4d.dynamic.ziggo.nl) |
| 05:47.41 | lkthomas | within SIP server MOH works fine |
| 05:48.44 | *** join/#asterisk oesteve (~oscar@145.Red-88-15-97.dynamicIP.rima-tde.net) |
| 05:50.23 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 05:50.51 | ChannelZ | re: if you play a song as part of your normal IVR -- or even hold your phone up to a speaker -- do they terminate the call? |
| 05:51.33 | lkthomas | I don't have IVR, let me try to do other one |
| 05:51.33 | ChannelZ | (I don't know much about the innards of PRI, if there is any type of notification * sends when you put the channel on hold) |
| 05:55.11 | lkthomas | LOL |
| 05:55.34 | lkthomas | those MOH I could hear is nothing related with my PBX |
| 05:55.38 | lkthomas | those sound doesn' |
| 05:55.44 | lkthomas | doesn't even exists |
| 05:55.56 | lkthomas | looks like they indeed send signal to PRI provider |
| 05:55.57 | ChannelZ | huh? |
| 05:57.49 | ChannelZ | What are the local devices (handsets) you're putting on hold from? |
| 05:57.56 | lkthomas | SIP |
| 05:58.17 | ChannelZ | And what does your console say when you do it? |
| 06:06.51 | *** join/#asterisk sjobke (~quassel@D57D8992.static.ziggozakelijk.nl) |
| 06:09.47 | lkthomas | ExecIf("Local/993239670@from-internal-fda1;2", "1?Set(DIAL_TRUNK_OPTIONS=M(setmusic^All-Tech-ops-Engineer))") in new stack |
| 06:09.57 | lkthomas | I believe dial trunk options include MOH call |
| 06:10.41 | lkthomas | <PROTECTED> |
| 06:10.48 | lkthomas | I got this line not long after dial trunk options |
| 06:16.29 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 06:29.46 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:add6:df91:8b2e:ca3) |
| 06:34.31 | *** join/#asterisk jhlavacek (~jirka@ata35-1-78-208-220-3.fbx.proxad.net) |
| 06:51.29 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 06:56.31 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:01.48 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 07:08.42 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 07:15.19 | *** part/#asterisk Magus` (cgrady@ip70-173-121-137.lv.lv.cox.net) |
| 07:15.55 | *** join/#asterisk znf (~ibm86@unu.card-share.eu) |
| 07:43.47 | *** join/#asterisk giany (~giany@shrek.x83.org) |
| 07:44.31 | giany | hi all, currently with use_q850_reason you can send something like this : Reason: Q.850;cause=19, is there a way I can set a "text" too? |
| 07:48.37 | *** join/#asterisk jetlag (~jetlag@pool-71-168-202-149.cmdnnj.east.verizon.net) |
| 07:50.41 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 07:52.14 | *** join/#asterisk danjenkins (~dan@cpc9-folk2-2-0-cust58.1-2.cable.virginm.net) |
| 07:53.07 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 07:55.45 | *** join/#asterisk danjenkins (~dan@cpc9-folk2-2-0-cust58.1-2.cable.virginm.net) |
| 07:56.13 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:59.21 | *** join/#asterisk wanna (~wanna@194.183.244.5) |
| 08:00.08 | *** join/#asterisk voxter (~voxter@irc.voxter.net) |
| 08:05.42 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 08:07.26 | wdoekes | giany: not with chan_sip, no |
| 08:07.52 | wdoekes | note that use_q850_reason works both ways, it will also attempt to parse incoming reason headers if available |
| 08:08.18 | giany | wdoekes: thing is that provider sends towards asterisk : Reason: Q.850;cause=19;text="no answer from the user" |
| 08:08.38 | giany | but asterisk can only send further Reason: Q.850;cause=19.. so no text |
| 08:10.04 | wdoekes | giany: http://fpaste.org/134466/ |
| 08:10.06 | wdoekes | something like that |
| 08:10.31 | wdoekes | although it will not forward the text, it will add a new text |
| 08:10.49 | *** join/#asterisk areski (~areski@80.174.128.93.dyn.user.ono.com) |
| 08:11.34 | wdoekes | (which would be "User alerting, no answer" for 19) |
| 08:12.16 | giany | that is the Q.850 reason? |
| 08:12.51 | wdoekes | ? |
| 08:14.52 | giany | https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings |
| 08:14.59 | giany | AST_CAUSE_NO_ANSWER |
| 08:14.59 | giany | 19. No answer from user (user alerted) |
| 08:14.59 | giany | OR2_CAUSE_NO_ANSWER |
| 08:14.59 | giany | 480, 483 |
| 08:15.34 | giany | <PROTECTED> |
| 08:35.42 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:50.10 | *** join/#asterisk hurdman (~ygcheny@c3p0.r0b0t.fr) |
| 08:50.28 | hurdman | hi, i'm exactly in the same case like here http://lists.digium.com/pipermail/asterisk-users/2014-May/283288.html |
| 08:50.49 | hurdman | anyone have a sip.conf and an extensions.conf that works with early video media ? |
| 08:51.43 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 08:58.01 | *** join/#asterisk miketo (~Michael@unaffiliated/miketo) |
| 08:59.11 | *** join/#asterisk LiuYan (~root@unaffiliated/liuyan) |
| 09:10.34 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 09:24.51 | *** join/#asterisk CeBe (~CeBe@brsg-4dbb1268.pool.mediaWays.net) |
| 09:48.10 | *** join/#asterisk workingcats (~workingca@212.122.48.77) |
| 09:54.13 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 10:03.21 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 10:28.12 | *** join/#asterisk iulhk (~iulhk@39.34.144.31) |
| 10:30.49 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 10:36.49 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:53.37 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 10:54.39 | *** join/#asterisk cunningpike (~cunningpi@2602:304:6eaf:6280:24fc:2055:8a2c:998c) |
| 11:43.45 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 12:00.22 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:05.56 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:09.04 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 12:13.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:13.28 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 12:13.51 | *** join/#asterisk w3rt (~jasonwert@71.89.137.28) |
| 12:14.35 | *** join/#asterisk jwr__ (~quassel@205.196.167.54) |
| 12:15.00 | *** join/#asterisk iulhk (~iulhk@39.34.144.31) |
| 12:15.30 | iulhk | i am using asterisk-11 , focus just sip to sip call, audio/video, in allowed codecs 'gsm;h264' , Issue peer 1000 calling to 2000 audio call, after ack, at receiver end video screen automatically opened, although caller started audio call, if i just enabled gsm, codec then both-end its audio screen, but in that case i can't start video as video codec not enabled. i want when user starting |
| 12:15.30 | iulhk | video call, it should start video at both-end when user starting audio call then at both end it should be audio call, is it possible ? |
| 12:20.01 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:27.55 | *** join/#asterisk blee (~blee@50-88-4-82.res.bhn.net) |
| 12:30.29 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-34-104.user.veloxzone.com.br) |
| 12:33.08 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 12:33.36 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 12:33.36 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:37.07 | *** join/#asterisk The_Guru (~guru@24-158-143-3.dhcp.kgpt.tn.charter.com) |
| 13:13.40 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 13:15.56 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 13:17.23 | *** join/#asterisk dhardison (uid20219@gateway/web/irccloud.com/x-hrhewrvfrvfgnerz) |
| 13:24.46 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 13:26.12 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 13:33.40 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 13:53.08 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-lyfaelehaozzhamt) |
| 14:03.03 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:17.29 | *** join/#asterisk jsjc (~Adium@90.174.2.71) |
| 14:20.32 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:21.35 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ihxcdsklqwnrrvgm) |
| 14:21.35 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:26.01 | *** join/#asterisk ThatCantBe (~irc@dsl.dyn-206.53.182.105.tbinet.bm) |
| 14:29.19 | *** join/#asterisk W0rmDr1nk (~wormdrink@unaffiliated/wormdrink) |
| 14:30.42 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-apkoozgtqmxgzmon) |
| 14:30.42 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:31.01 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:32.41 | *** part/#asterisk LiuYan (~root@unaffiliated/liuyan) |
| 14:42.19 | Zogot | https://issues.asterisk.org/jira/browse/ASTERISK-23807 anything further with this and things like transports in realtime? there are database tables defined in alembic but if they are not for intended to be used |
| 14:42.59 | Zogot | they perhaps should be dropped from alembic |
| 14:44.48 | newtonr | Bugs should be filed for those that are not used, and then they will be removed eventually. |
| 14:45.03 | newtonr | As is likely the case with ASTERISK-23807 |
| 14:46.59 | hurdman | Any idea of why i don't get my early video media ? http://cylon.r0b0t.fr/zerobin/index.php?95fbf099909ad9c5#6hplYLG40cd5a+jLTitqMPeEXDzU6nXrT7bVHtFbNCo= |
| 14:49.41 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:1dd0:221:6aff:feb8:e0b2) |
| 14:52.20 | *** join/#asterisk dipanjan (671b082b@gateway/web/freenode/ip.103.27.8.43) |
| 14:52.29 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:53.28 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-civfuphucwwddgcx) |
| 14:56.58 | dipanjan | [TK]D-Fender: I upgraded as you suggested yesterday. Still facing the same problem. When a callee disconnects the call when the script is expecting an input then the BYE is not detected. I keep getting this message every 10 seconds: WARNING[15867]: chan_sip.c:4258 __sip_autodestruct: Autodestruct on dialog '05e367c47b0884714dd2b34143cc7dd4@10.208.26.150:5060' with owner SIP/10.208.26.54:5060-0000005c in place (Method: BYE). Resched |
| 14:57.45 | [TK]D-Fender | PASTEBIN the entire call along with your script |
| 14:59.58 | *** part/#asterisk fullstop (~fullstop@107.170.53.155) |
| 15:01.56 | hurdman | thinks that early video media with sip on asterisk has never worked. |
| 15:07.36 | dipanjan | [TK]D-Fender: call: http://pastebin.com/AjuGerCj script: http://pastebin.com/z3deFD3h |
| 15:08.25 | [TK]D-Fender | dipanjan: You did not include SIP debug. pace another call. |
| 15:09.08 | [TK]D-Fender | place* |
| 15:13.11 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:13.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:15.43 | dipanjan | [TK]D-Fender: http://pastebin.com/uU02yytL the intermittent warnings are from the zombie of the previous call |
| 15:18.05 | hurdman | here my full sip debug, my dialplan and my sip.conf... any idea of why i doesn't get any video early media but the video is ok during call ? http://cylon.r0b0t.fr/zerobin/index.php?8b694d1dd6986936#bx/koS9CS/nmR+aaOgnv4hdwo0lhuUZnx5ulbfeXtX4= |
| 15:19.29 | [TK]D-Fender | dipanjan: I'm not seeing anything after the end fo the call including channel dumps, etc./.. |
| 15:20.44 | [TK]D-Fender | dipanjan: $SIG{HUP} = "IGNORE"; <-- and your script is IGNORING HUP |
| 15:23.55 | dipanjan | [TK]D-Fender: removed the line with the HUP IGNORE. Same result |
| 15:28.41 | [TK]D-Fender | Not sure at this point.... hopefully someone with mor AGI/PERL experience can pick up from here. |
| 15:29.30 | hurdman | dipanjan: what was your problem ? |
| 15:29.32 | dipanjan | [TK]D-Fender: OK. That line was the problem. The last call was discarded after the first 10 seconds expired. |
| 15:29.54 | [TK]D-Fender | dipanjan: Good, that was it then... |
| 15:30.02 | dipanjan | [TK]D-Fender: thanks. Did not notice that line was there. Very old code :) |
| 15:30.32 | [TK]D-Fender | First thing you should ahve looked for... |
| 15:30.35 | [TK]D-Fender | hangup handler... |
| 15:31.00 | dipanjan | hurdman: zombie calls remained on when a user disconnected the call when the script expects an input. In other cases hangup was normal. |
| 15:31.00 | [TK]D-Fender | At least now you're also on a more modern version that is supported |
| 15:31.10 | hurdman | dipanjan: ah ok |
| 15:31.12 | dipanjan | [TK]D-Fender: Right. I will remember that. |
| 15:32.06 | dipanjan | Thanks all |
| 15:32.11 | file | I support cheese. |
| 15:36.24 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 15:47.15 | *** join/#asterisk aness (~aness@cpe-74-68-148-41.nyc.res.rr.com) |
| 15:50.47 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 16:13.56 | *** join/#asterisk oesteve (~oscar@145.Red-88-15-97.dynamicIP.rima-tde.net) |
| 16:27.40 | *** join/#asterisk casdude (57ed466d@gateway/web/freenode/ip.87.237.70.109) |
| 16:27.46 | casdude | hi |
| 16:32.58 | *** join/#asterisk oesteve (~oscar@145.Red-88-15-97.dynamicIP.rima-tde.net) |
| 16:33.27 | casdude | I have a seemingly random issue, I receive congestion on an outbound trunk every now and then and the all circuits busy message is played. I have read some forums and think that it maybe due to the system not waiting longer enough prior to giving up |
| 16:33.58 | casdude | is there anyway of me getting the system to wait longer? as there are lines available, i just think the telco is a little slow |
| 16:34.21 | [TK]D-Fender | Go look and prove what's actually happening. |
| 16:35.38 | [TK]D-Fender | "Some forum" doesn't really mean anything. They aren't seeing your call. |
| 16:36.59 | The_Guru | i've heard forum trolls are a great scarcity nowadays... becoming extinct |
| 16:37.31 | [TK]D-Fender | that's the exact sort of thing a troll would say... |
| 16:37.37 | The_Guru | :) |
| 16:38.30 | casdude | I have fairly hard skin I can take it |
| 16:38.57 | casdude | sorry for the lack of back ground explanation Fender |
| 16:39.27 | *** join/#asterisk blee (~blee@142.196.167.138) |
| 16:40.22 | The_Guru | from the earlier explanation my recommendation would be to increase the number of channels. |
| 16:41.12 | casdude | thanks Guru lines are fine, seems to be random, no particular load |
| 16:41.14 | casdude | issues |
| 16:41.32 | casdude | i receive outisbusy and Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 1. With this being a seemingly random event i took it was not a configuration issue |
| 16:41.38 | [TK]D-Fender | from the earlier explanation my recommendation would be go show us the actual call so we aren't playing guessing games about what's really happening based on a very loose description. |
| 16:41.43 | [TK]D-Fender | PASTEBIN is your friend. |
| 16:41.45 | [TK]D-Fender | ~pb |
| 16:41.45 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:41.47 | [TK]D-Fender | ^^^ |
| 16:41.54 | casdude | nps on it |
| 16:42.01 | The_Guru | i really like Fenders recommendation |
| 16:42.22 | [TK]D-Fender | likes Reality-based approaches. |
| 16:44.01 | casdude | http://pastebin.com/2JX8weUL |
| 16:44.33 | casdude | here we are thanks for your help |
| 16:49.31 | *** join/#asterisk workingcats (~workingca@212.122.48.77) |
| 16:50.43 | [TK]D-Fender | casdude: Looks like network congestion. |
| 16:50.53 | casdude | umm |
| 16:50.59 | [TK]D-Fender | casdude: Or possibly an oddly-reported "busy" |
| 16:51.05 | [TK]D-Fender | casdude: PRI, correct? |
| 16:51.08 | casdude | PRI |
| 16:51.24 | [TK]D-Fender | "pri debug span X" <- |
| 16:51.30 | [TK]D-Fender | Go look at the actual comms |
| 16:51.41 | casdude | k |
| 16:51.44 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 16:53.29 | casdude | thanks for your help |
| 16:56.49 | hurdman | pompompom, snif, nothing works :'( |
| 16:56.51 | The_Guru | line #82 looked real spooky. must be getting close to Halloween. |
| 16:59.09 | The_Guru | anyone know of any way cool reminder scripts for like a doctors office? i'm way too poor for cepstral or to actually pay for one. |
| 17:01.27 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 17:09.55 | *** join/#asterisk smellis_werk (~smellis@2001:470:c452:2::9) |
| 17:12.11 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 17:22.46 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:23.30 | [TK]D-Fender | Festival for TTS |
| 17:27.39 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 17:31.40 | *** join/#asterisk blee (~blee@50-88-4-82.res.bhn.net) |
| 17:36.19 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:59.03 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 17:59.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 18:03.18 | *** join/#asterisk CeBe (~CeBe@brsg-4dbb1268.pool.mediaWays.net) |
| 18:30.23 | *** join/#asterisk al_nz1 (~bigal.nz@203-173-134-2.dsl.dyn.ihug.co.nz) |
| 18:31.12 | al_nz1 | I can see from both a packet race and debug that my cisco phone is getting UNAUTHORISED when trying to register - I have double checked the password in the cnf file and the web gui on PIAF. |
| 18:31.25 | al_nz1 | Is there another way from the CLI on the server to verify extension passwords? |
| 18:31.33 | hurdman | al_nz1: can you show your sip.conf ? |
| 18:31.43 | al_nz1 | I presume they are not stored in clear text anywhere? |
| 18:31.43 | hurdman | en paste your sip debug log ? |
| 18:32.18 | al_nz1 | hurdman: I got a packet dump here insead of SIP lg? |
| 18:32.38 | hurdman | ok |
| 18:32.41 | hurdman | and sip.conf |
| 18:32.58 | al_nz1 | ok - where is sip.conf kept? |
| 18:33.07 | hurdman | cat /etc/asterisk/sip.conf |
| 18:33.08 | al_nz1 | .../etc/ ? |
| 18:33.10 | al_nz1 | ok |
| 18:33.37 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-grknnwscjlnjzrsc) |
| 18:33.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:35.08 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 18:36.36 | al_nz1 | http://pastebin.com/5F4YKjGP |
| 18:36.40 | al_nz1 | sip.conf |
| 18:37.43 | hurdman | arf, there's only include into :) can you paste all the sip**** please ? |
| 18:37.54 | [TK]D-Fender | .... |
| 18:37.57 | [TK]D-Fender | no need |
| 18:38.07 | [TK]D-Fender | do NOT just paste it. |
| 18:38.16 | [TK]D-Fender | you can see the secret on your peer in sip_additional.conf |
| 18:38.33 | [TK]D-Fender | Which will be the same as on the extension in the GUI unless you failed to commit a change |
| 18:39.07 | al_nz1 | [TK]D-Fender: awsome!!! |
| 18:39.16 | al_nz1 | [TK]D-Fender: and the bloody password is differnt |
| 18:39.25 | al_nz1 | I cant work out why it didnt commit the change |
| 18:39.36 | [TK]D-Fender | "Apply" |
| 18:40.06 | al_nz1 | Im sure I did! |
| 18:40.23 | [TK]D-Fender | It looks like you might be wrong. |
| 18:41.01 | al_nz1 | well...I will make another change, apply, and then recheck....but great answer - thanks |
| 18:43.08 | al_nz1 | bugger!~ |
| 18:43.15 | al_nz1 | I am still getting Unauthorised |
| 18:43.50 | hurdman | al_nz1: you're pcap only show a 401 |
| 18:43.50 | hurdman | :/ |
| 18:44.13 | hurdman | perhaps you will get more info into your asterisk console using sip debug ? |
| 18:44.42 | al_nz1 | actually I take it back, I looked at the wrong extension - the password was right in the first place |
| 18:44.49 | al_nz1 | sure sip debug coming up |
| 18:47.14 | hurdman | [TK]D-Fender: keep cool, i'm not an hackers ^^" |
| 18:50.15 | al_nz1 | http://pastebin.com/LnsPDuBy |
| 18:50.21 | al_nz1 | debug |
| 18:54.38 | al_nz1 | brb |
| 18:57.49 | hurdman | al_nz1: i can't see anything that can help sorry. imho, just check all your sip***.conf to see if username/password etc ... are ok, |
| 18:58.33 | hurdman | ( and i never user the web gui, only conf file :/) |
| 19:02.35 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:02.37 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:02.44 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:09.56 | al_nz1 | hurdman: ta |
| 19:10.09 | al_nz1 | [TK]D-Fender : anything you can see? |
| 19:10.26 | [TK]D-Fender | no |
| 19:12.10 | al_nz1 | weiiird |
| 19:12.21 | al_nz1 | [TK]D-Fender: you saw the pic |
| 19:12.24 | al_nz1 | ? |
| 19:12.35 | [TK]D-Fender | super tiny almost unreadable |
| 19:12.38 | [TK]D-Fender | but looked the same |
| 19:12.53 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.12.1 (2014/09/18), 1.8.30.0 (2014/08/19); Standard: Asterisk 12.5.1 (2014/09/18); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 19:16.11 | al_nz1 | got me stumped then |
| 19:17.30 | *** join/#asterisk aross42 (~aross@h46-190.reznet.ucalgary.ca) |
| 19:19.11 | [TK]D-Fender | I haven't seen teh actual failure yet |
| 19:20.33 | al_nz1 | 401 auth error on sevrer |
| 19:22.40 | [TK]D-Fender | pastebin the whole thing. |
| 19:22.53 | [TK]D-Fender | 401 can be a CHALLENGE and not just a refusal |
| 19:30.29 | al_nz1 | [TK]D-Fender: http://pastebin.com/LnsPDuBy |
| 19:30.46 | al_nz1 | yes I am reading and see that 401 and just be a reposnse to send the hashed password? |
| 19:31.46 | [TK]D-Fender | al_nz1: What is your NAT setting in the extension? |
| 19:32.01 | al_nz1 | [TK]D-Fender: never |
| 19:32.09 | [TK]D-Fender | al_nz1: set to YES |
| 19:32.09 | al_nz1 | but I have tried both |
| 19:33.17 | al_nz1 | done. and applied |
| 19:56.07 | al_nz1 | didnt register |
| 19:58.44 | *** join/#asterisk aness (~aness@cpe-74-68-148-41.nyc.res.rr.com) |
| 19:59.31 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 20:01.46 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:17.28 | *** join/#asterisk drale (~drale@unaffiliated/drale) |
| 20:43.03 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 20:58.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:02.18 | *** join/#asterisk oesteve (~oscar@145.Red-88-15-97.dynamicIP.rima-tde.net) |
| 21:09.16 | *** join/#asterisk Zedax (d4e1d545@gateway/web/freenode/ip.212.225.213.69) |
| 21:09.46 | Zedax | hello is there any web gui for current version of asterisk? |
| 21:13.14 | Penguin | FreePBX seems to work with Asterisk. |
| 21:14.09 | Penguin | |
| 21:19.15 | Zedax | Penguin: does it work well if you're not using their distro? |
| 21:19.31 | [TK]D-Fender | No different |
| 21:19.42 | Penguin | FreePBX is just a web application. |
| 21:24.55 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-grknnwscjlnjzrsc) |
| 21:26.57 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 21:28.45 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 21:29.44 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 21:30.17 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 21:37.01 | *** join/#asterisk diametric (~diametric@2604:3400:dc1:43:216:3eff:fe27:bf9d) |
| 21:48.01 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:50.46 | marceloamorim | Hello guys, I`m building my own call record details on mysql and I wish to get the variable for the call starting to ring and when the extension pick up the call |
| 21:51.29 | marceloamorim | like cel and cdr does with those data to mysql |
| 21:51.43 | marceloamorim | is that possible? |
| 21:53.11 | [TK]D-Fender | There is no "starting to ring" |
| 21:53.26 | [TK]D-Fender | There is "before dial", and "upon answer" |
| 21:53.48 | *** join/#asterisk tm1000 (tm1000@2600:3c00::f03c:91ff:fe6e:6894) |
| 21:55.11 | marceloamorim | I should get those "BEFORE DIAL" and "UPON ANSWER" from the cel database or I can get those variables from channel? |
| 21:56.57 | [TK]D-Fender | You could get them from channel |
| 21:57.21 | [TK]D-Fender | I don't know CEL |
| 21:57.47 | marceloamorim | at this moment I`m using app Dial with option "g" to keep going on my dialplan and I `ll put those variables to the mysql |
| 21:58.30 | marceloamorim | do you know which variables we are talking about? |
| 21:59.27 | *** join/#asterisk danjenkins (~dan@mail.hotel-latour.co.uk) |
| 22:00.31 | [TK]D-Fender | No. |
| 22:00.47 | [TK]D-Fender | You just said "the variable" You gave no useful description |
| 22:02.03 | zerick | Is it possible to run Asterisk and FreeSwitch on the same server ? (both running at same time, independently) |
| 22:03.20 | marceloamorim | the variable ( START TO RINGING) and variable ( ANSWER THE PHONE ) |
| 22:03.47 | Penguin | zedax: As I already said: Just don't try to bind them both to the same port(s) on the same IP address(es). |
| 22:04.02 | Penguin | marceloamorim: Those are not variables. |
| 22:05.27 | [TK]D-Fender | ^^ |
| 22:06.35 | marceloamorim | so there is no way to get the same informations that a CDR and CEL? |
| 22:06.43 | marceloamorim | my english is bad, I know that |
| 22:07.05 | marceloamorim | but didn`t you understand what I`m saying? =( |
| 22:07.14 | [TK]D-Fender | marceloamorim: No. |
| 22:07.31 | [TK]D-Fender | WHAT INFORMATION <-- You are not telling us WHAT information you want to store |
| 22:07.43 | Penguin | I don't think there is a variable created at the startup of Dial(). |
| 22:08.04 | marceloamorim | the time |
| 22:08.23 | [TK]D-Fender | before you dial go STORE the time |
| 22:09.22 | [TK]D-Fender | marceloamorim: "core show function STRFTIME" |
| 22:09.23 | marceloamorim | I need the time before start to call, the time the phone start to ring and the time that phone answer the call |
| 22:09.50 | [TK]D-Fender | [18:09]marceloamorimI need the time before start to call, the time the phone start to ring and the time that phone answer the call <- I already told you you could do TWO of these. |
| 22:10.01 | [TK]D-Fender | There is NO option for "start of ringing" |
| 22:10.22 | Penguin | You can only get the time BEFORE you execute Dial(). |
| 22:11.20 | [TK]D-Fender | And right as they answer |
| 22:11.38 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 22:12.09 | alexw | My voicemail has two beeps - "The person at ext 1000 is on the phone please leave a message after the beep, (beep) (beep)" |
| 22:12.42 | marceloamorim | those variables are CHAN_START and ANSWER? |
| 22:13.10 | [TK]D-Fender | no |
| 22:13.14 | [TK]D-Fender | there are no variables |
| 22:13.17 | [TK]D-Fender | stop looking for them |
| 22:13.20 | [TK]D-Fender | they do not exist |
| 22:13.26 | alexw | <SIP/1000-00000005> Playing 'vm-isonphone.gsm' (language 'en'), <SIP/1000-00000005> Playing 'vm-intro.gsm' (language 'en'), <SIP/1000-00000005> Playing 'beep.gsm' (language 'en') |
| 22:13.26 | [TK]D-Fender | LOOK AT THE TIME |
| 22:13.31 | [TK]D-Fender | [18:09][TK]D-Fendermarceloamorim: "core show function STRFTIME" |
| 22:14.48 | Zedax | i get ***Checking for PEAR DB..FAILED [FATAL] PEAR must be installed (requires DB.php). Include path: .:****, i have pear, how do i say the path? |
| 22:15.11 | marceloamorim | maybe you know something that I don`t know, because if I set ${EPOCH} before the dial and set ${EPOCH} after the app DIAL, how can I get those times correctly |
| 22:16.04 | [TK]D-Fender | You don't SET that variable. |
| 22:16.09 | [TK]D-Fender | it is READ-ONLY |
| 22:16.26 | marceloamorim | I don`t need to set, I need to read the time |
| 22:16.34 | [TK]D-Fender | So go read it |
| 22:16.45 | marceloamorim | gez |
| 22:16.48 | [TK]D-Fender | ${EPOCH}: The current UNIX-style epoch (number of seconds since 1 Jan 1970) |
| 22:16.59 | [TK]D-Fender | [18:09][TK]D-Fendermarceloamorim: "core show function STRFTIME" |
| 22:17.13 | marceloamorim | dude, I know THE FUNCTION STRFTIME AND THE EPOCH |
| 22:17.49 | [TK]D-Fender | Then go use them and get the time |
| 22:19.40 | alexw | [TK]D-Fender any idea how to remove the second beep from voicemail <SIP/1000-00000008> Playing 'beep.gsm' (language 'en_AU'? |
| 22:20.15 | marceloamorim | let me try one more time, If I use same => n,Set(time_before_dial=${EPOCH}) and then same => n,DIAL(whatever) and then same => n,Set(time_after_dial)=${EPOCH} this information is useless |
| 22:20.58 | marceloamorim | I need the exactly time that phone answer the call |
| 22:21.48 | [TK]D-Fender | So go do it upon answer |
| 22:23.11 | marceloamorim | I need ppl answering the phone, not the pbx |
| 22:24.30 | [TK]D-Fender | wonders what that is supposed to mean.... |
| 22:24.40 | marceloamorim | app Answer |
| 22:24.52 | [TK]D-Fender | Who said to use that app? |
| 22:24.55 | [TK]D-Fender | I never did |
| 22:25.34 | marceloamorim | so what do you mean with upon answer? |
| 22:27.53 | [TK]D-Fender | "core show application dial" <- you clearly aren't reading the instructions for the apps you are using |
| 22:28.07 | [TK]D-Fender | go set it when the OTHER SIDE ANSWERS |
| 22:28.15 | *** join/#asterisk hrnt (hrnt@wintermute.hrnt.net) |
| 22:29.13 | marceloamorim | ok, next time when I full understand the all applications, I`m back and ask you something else. Thank you |
| 22:29.41 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 23:12.38 | *** part/#asterisk Zedax (d4e1d545@gateway/web/freenode/ip.212.225.213.69) |
| 23:15.16 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 23:16.43 | Dovid | Hi. I am having a problem using FastAgi that if a call is hung up just at the correct time the php script will complete however no Asterisk functions will work. so my problem is that ant SQL statements that are prepaired are ran but anything that is asterisk related (like setting a variable) is not recorded. is there any way around it? |
| 23:20.50 | *** join/#asterisk u0m3 (~u0m3@92.80.106.41) |
| 23:35.49 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 23:38.10 | Dovid | I am having a problem where a call is hung up on but the agi continues. would this be the issue? https://reviewboard.asterisk.org/r/1165/ |
| 23:39.48 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 23:46.06 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 23:52.02 | zerick | How could I call between two extensions that are on different contexts ? |
| 23:53.23 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |