| 00:00.58 | redotis | damn |
| 00:01.06 | redotis | I can't get confbridge to work |
| 00:01.40 | ChannelZ-Wk | redotis: how so |
| 00:01.55 | navaismo | ChannelZ-Wk, http://pastebin.com/cvbbK8qq |
| 00:02.52 | redotis | http://hastebin.com/laxewivada.coffee |
| 00:02.52 | snadge | ok so no installation recommendations :P |
| 00:03.13 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 00:03.35 | ChannelZ-Wk | redotis: module load app_confbridge |
| 00:04.34 | redotis | Unable to load module app_confbridge |
| 00:04.34 | redotis | Command 'module load app_confbridge' failed. |
| 00:05.18 | ChannelZ-Wk | What version of asterisk? |
| 00:06.24 | redotis | 11.5.` |
| 00:06.27 | navaismo | ChannelZ-Wk, this is with firewall off -->http://pastebin.com/iKyYMMRK |
| 00:06.28 | redotis | 11.5.1 |
| 00:07.03 | *** join/#asterisk Pullphinger (~Pullphing@c-24-13-69-42.hsd1.il.comcast.net) |
| 00:07.39 | ChannelZ-Wk | navaismo: so let's see the output of iptables -L INPUT -v -n with the firewall up, and what port range do you have in rtp.conf |
| 00:07.56 | ChannelZ-Wk | redotis: did you build it yourself? |
| 00:08.29 | redotis | no |
| 00:08.34 | redotis | It's AsteriskNOW |
| 00:09.15 | ChannelZ-Wk | Hmm. Well I don't know then.. if the module ain't there, it ain't there |
| 00:09.28 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 00:09.57 | redotis | wtf |
| 00:09.58 | redotis | great |
| 00:10.12 | redotis | You'd figure it would come with AsteriskNOW |
| 00:10.36 | ChannelZ-Wk | I don't know much about NOW.. if it's freepbx, are they still MeetMe-centric? I have no idea |
| 00:11.19 | redotis | you got me |
| 00:12.25 | ChannelZ-Wk | well you can double-check looking in /usr/lib/asterisk/modules/ (or maybe not, if they're somewhere else in NOW) for app_confbridge.so |
| 00:12.26 | navaismo | ChannelZ-Wk, the output --->http://pastebin.com/TKTn0xZt ports in rtp.conf are 19000 to 20000 |
| 00:15.32 | ChannelZ-Wk | navaismo: You don't have nf_conntrack_sip or nf_nat_sip loaded do you? Is this a firewall script? |
| 00:16.06 | redotis | hah |
| 00:16.07 | redotis | it's there |
| 00:16.32 | ChannelZ-Wk | redotis: did you type-o then, or is it listed as noload or something in modules.conf? (not sure if that prevents a manual load, just a thought) |
| 00:16.54 | redotis | usr/lib64/asterisk/app_confbridge.so |
| 00:17.02 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.18) |
| 00:17.47 | ChannelZ-Wk | navaismo: also the IP you're calling from isn't being caught in your "blockhosts" table is it? |
| 00:18.14 | navaismo | nope, if its in the blockhost i cant reach the server |
| 00:18.22 | navaismo | so blockhosts is not the issue |
| 00:19.05 | redotis | wtholy fuck |
| 00:19.09 | redotis | i have no clue |
| 00:19.58 | navaismo | redotis, check the full log maybe there is the cause why isnt loading the module |
| 00:20.29 | redotis | where is the log |
| 00:21.58 | ChannelZ-Wk | /var/log/asterisk/messages usually |
| 00:22.15 | ChannelZ-Wk | though you should see verbose on the console |
| 00:22.27 | navaismo | they need to check at full log |
| 00:22.30 | navaismo | HE* |
| 00:23.02 | ChannelZ-Wk | assuming it's on |
| 00:23.40 | redotis | ok |
| 00:23.45 | redotis | it's something in that file |
| 00:23.47 | redotis | you need |
| 00:23.53 | redotis | i copied the sample over and it works |
| 00:24.08 | redotis | so there is some kind of bare minimum that must be in it |
| 00:24.48 | navaismo | well i just want to see if the full log tell you the cause about the module load error |
| 00:25.10 | ChannelZ-Wk | http://pastebin.com/4t8gGKfL |
| 00:25.35 | ChannelZ-Wk | wondering why the console isn't showing the load error. Come to think of it your first paste of the 'core show application' looked odd/bare. |
| 00:29.24 | redotis | so looks like the log isn't on |
| 00:29.27 | redotis | how do i turn it on |
| 00:30.11 | redotis | Why is the module not loading when i reload asterisk |
| 00:30.46 | ChannelZ-Wk | well is it still failing on your config? |
| 00:31.11 | ChannelZ-Wk | If not, autoload in modules.conf probably isn't on, or it's set to noload that one |
| 00:31.17 | redotis | it loads when i type module load app_confbridge |
| 00:31.33 | ChannelZ-Wk | so check your /etc/asterisk/modules.conf |
| 00:31.47 | redotis | i renamed it to modules.conf.old |
| 00:31.57 | redotis | after I initially had the problem |
| 00:32.24 | ChannelZ-Wk | well.. that's one problem |
| 00:32.41 | ChannelZ-Wk | I dunno what autoload defaults to. No I suppose. |
| 00:32.46 | redotis | yeah it's working now |
| 00:32.57 | redotis | that i renamed modules.conf |
| 00:33.17 | redotis | ok so apparently you have to have modules.conf with autoload=yes |
| 00:33.26 | redotis | and a bare minimum in confbridge.conf of |
| 00:33.47 | redotis | [default_bridge] |
| 00:33.47 | redotis | type=bridge |
| 00:33.49 | redotis | ok |
| 00:33.50 | redotis | thanks guys |
| 00:36.33 | redotis | how do i enable logging |
| 00:36.35 | redotis | last question |
| 00:36.41 | redotis | since it's obviously important |
| 00:36.45 | redotis | :) |
| 00:38.02 | navaismo | got to go see you guys thanks to all |
| 00:40.31 | ChannelZ-Wk | logger.conf |
| 00:41.00 | ChannelZ-Wk | see if you have console => notice,warning,error at minimum |
| 00:41.18 | ChannelZ-Wk | There's probably a commented-out 'full => ...' line that you can uncomment as well |
| 00:42.18 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 00:43.20 | redotis | lol |
| 00:43.23 | redotis | there's no logger.conf |
| 00:43.27 | redotis | i started from scratch |
| 00:43.48 | redotis | so i should just put console => notice,warning,erro |
| 00:43.53 | redotis | in it |
| 00:43.56 | Katty | howdy |
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| 00:44.57 | redotis | I see the sample |
| 00:45.02 | redotis | thanks...i |
| 00:45.07 | redotis | I'll use that one |
| 00:45.13 | redotis | thank you guys |
| 00:45.14 | redotis | later |
| 00:45.29 | *** part/#asterisk redotis (~redotis@nc-184-3-102-139.dhcp.embarqhsd.net) |
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| 02:35.40 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.1 (2013/08/27), 10.12.3 (2013/08/27), 1.8.23.1 (2013/08/27), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 05:43.37 | tulga | is there any asterisk solution to enable iphone voicemail section? because my country not support by apple, and voicemal tab not working. |
| 05:48.19 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 05:59.31 | [TK]D-Fender | tulga: what "voicemail tab"? |
| 05:59.44 | [TK]D-Fender | Asterisk has nothng to do with the operation of an iPhone |
| 06:08.05 | *** join/#asterisk Dovid (~Dovid@float-gw.core-fw-t.mypbxmanager.net) |
| 06:12.00 | tulga | [TK]D-Fender: iphone phone app has 5 tabs like favorite, recent, contact, keypad and voicemail |
| 06:12.22 | [TK]D-Fender | tulga: Asterisk has no relationship with the dialer app |
| 06:12.53 | *** join/#asterisk peektoseen (~peektosee@193.105.235.1) |
| 06:13.05 | peektoseen | hi all! |
| 06:13.06 | tulga | [TK]D-Fender: I mean if install asterisk with voicemail feature, is it possible to connect this voicemail tab to asterisk |
| 06:13.38 | peektoseen | how asterisk know, what peers unreachable? |
| 06:14.07 | [TK]D-Fender | tulga: There is no "connect" |
| 06:14.23 | Penguin | I still don't know what a voicemail tab is. |
| 06:14.27 | tulga | [TK]D-Fender: ok, only operator and apple |
| 06:14.29 | [TK]D-Fender | tulga: Asterisk will not in any way reconfigure your iPhone |
| 06:14.46 | Penguin | peektoseen: SIP OPTIONS packets |
| 06:14.49 | peektoseen | I have unreachable/reachable host blinking, but peers ping is less than 1ms |
| 06:15.17 | Penguin | ping is ICMP. SIP OPTIONS is a much higher level of the OSI model. |
| 06:15.19 | peektoseen | if Options doesn't recive - host unreachable? |
| 06:15.32 | [TK]D-Fender | Host is CONSIDERED unreachable |
| 06:15.45 | Penguin | If the device does not respond with something, anything, it will be considered unreachable. |
| 06:15.51 | Penguin | Most devices respond with something like a 404. |
| 06:16.01 | Penguin | But it's a response, so asterisk considers it good. |
| 06:16.22 | [TK]D-Fender | It is a method of A: seeing if they are alive, and B: keeping UDP traffic forwarded on remote NAT's to ensure incoming call can make it |
| 06:17.30 | peektoseen | ok, I try sniff traffic for OPTIONS |
| 06:17.47 | peektoseen | thank you all |
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| 06:49.13 | *** join/#asterisk BKhan (ca7d90e3@gateway/web/freenode/ip.202.125.144.227) |
| 06:49.20 | BKhan | Hi |
| 06:49.52 | BKhan | I have some issue call got drop in queue after 10 minutes. Please advise |
| 06:50.03 | peektoseen | I have a many time REGISTER message from peer, but asterisk doesn't answert to it, or aswer with 500(server error) or 401(unauth) :( |
| 06:50.04 | peektoseen | http://pastebin.com/raw.php?i=BR5nQ93X |
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| 07:01.01 | *** join/#asterisk bombev (~User@73.109.Globcom.Net) |
| 07:01.43 | bombev | helllo :) is there application or function where I will be able to record data about the call into file |
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| 07:07.57 | *** join/#asterisk kazak1377 (c169f90a@gateway/web/freenode/ip.193.105.249.10) |
| 07:08.15 | kazak1377 | hello everybody again. |
| 07:08.31 | kazak1377 | Is there any kind of arrays in asterisk? |
| 07:08.32 | Penguin | And that was how I hacked kazak1377's asterisk. |
| 07:08.39 | Penguin | Oh, hi there kazak1377. |
| 07:09.02 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:41cf:761a:1fd0:bb9e) |
| 07:09.10 | Penguin | What kind of arrays? What are you trying to do? |
| 07:10.13 | kazak1377 | Penguin: i have an shell script, that returns something like this: |
| 07:10.16 | kazak1377 | file1 |
| 07:10.18 | kazak1377 | file2 |
| 07:10.22 | kazak1377 | file3 |
| 07:10.28 | kazak1377 | so on so on. |
| 07:10.49 | Penguin | I'm with you so far. |
| 07:11.14 | kazak1377 | I need to loop ControlPlayback for those files |
| 07:12.22 | kazak1377 | so, how would be assigned an multi-line ${SHELL()} result? |
| 07:14.40 | wdoekes | kazak1377: I'd replace the linefeeds with comma's. you can "loop" over the result with CUT |
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| 07:17.02 | kazak1377 | wdoekes: hm... nice idea. Thanks) |
| 07:22.36 | bombev | Penguin do you know such application where I can record data about the call |
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| 07:30.06 | *** join/#asterisk TobSnyder (~schneider@146-52-43-241-dynip.superkabel.de) |
| 07:31.03 | TobSnyder | is it still true that asterisk does not support multiple SIP registrations for one SIP account? e.g. when having a desktop and a notebook, each one running a SIP client and wants to register with same credentials at asterisk so that incoming call will ring both= |
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| 07:50.07 | wdoekes | TobSnyder: that is correct |
| 07:50.26 | TobSnyder | hmpf :( |
| 07:50.28 | wdoekes | with chan_sip at least. it might be possible with chan_pjsip in asterisk 12 |
| 07:50.58 | TobSnyder | well I am running a quite old version based on Elastix |
| 07:51.14 | wdoekes | can't you put the desktop and notebook in a callgroup? |
| 07:51.17 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 07:51.25 | wdoekes | and ring them both on incoming call? |
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| 07:58.30 | BKhan | hi please advise : I have some issue call got drop in queue after 10 minutes |
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| 08:26.18 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 08:26.25 | v0lZy | hi |
| 08:26.50 | v0lZy | I was wondering if anyone could help me identify the root cause of my problem... |
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| 08:27.54 | v0lZy | Audio quality is really bad for calls incoming through ITSP. Ringing tones are clear, MOH is clear, and if the callee places the caller on hold and then resumes the conversation, the sound quality magically becomes OK |
| 08:28.37 | v0lZy | happens only for incoming calls, outgoing calls dont pose a problem |
| 08:31.12 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 08:31.25 | Blashyrkh | how can i see what debug and verbose levels are set in asterisk cli? |
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| 08:41.29 | bulkorok | Blashyrkh: I "think" if you log in to CLI it tells you that right after it comes up |
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| 08:50.41 | Neoti | hi all |
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| 08:59.53 | gnu_d | Hi, I'm trying to disable something in the code, it's not letting Firefox to do a call, in the c source file: chan_sip.c, there is a line with log: "Rejecting secure audio stream without encryption details:", this line is being executed if =crypto is not present, as I read Firefox doesn't send that line, is there a proper way to workaround it and why this line =crypto is used for ? |
| 09:04.59 | *** join/#asterisk Prosouth__ (~sabayonus@62-2-198-100.static.cablecom.ch) |
| 09:05.17 | Neoti | gnu_d: i had the same prob i disabled the secure tls sll thing in modules.conf file... |
| 09:05.48 | gnu_d | Neoti: I don't need to hack the code ? |
| 09:06.24 | Neoti | i just removed the modual and restarted asterisk and it worked... |
| 09:06.56 | Neoti | i did not have the same prob with firefox.. the prob was with a actual phone and asterisk rejected the call ... so i disableed the mod simple... |
| 09:07.21 | Neoti | i will be enabling it again as i need to do secure calls using certs .... :( |
| 09:07.30 | *** join/#asterisk ketas (~ketas@135.216.191.90.dyn.estpak.ee) |
| 09:08.07 | gnu_d | Neoti: I can't find tls stuff inside the modules.conf file. |
| 09:10.22 | peektoseen | I have more than 10 SIP peers connected to asterisk. Each peer have qualify=yes . In tcpdump traffic I doesn't see any OPTIONS package. Why asterisk dont send OPTIONS? |
| 09:11.21 | gnu_d | Neoti: is this the one: tlsenable=yes ? - But this key is located in http.conf |
| 09:12.05 | kaldemar | peektoseen: do you see them in asterisk with sip debug? |
| 09:12.52 | kaldemar | peektoseen: and what does your "sip show peers" say? |
| 09:14.39 | peektoseen | kaldemar: i see them in ngrep - sniffer tool. |
| 09:16.32 | peektoseen | kaldemar: sip show peers |
| 09:16.34 | peektoseen | http://pastebin.com/raw.php?i=8HS2wHtt |
| 09:17.26 | kaldemar | peektoseen: looks like your asterisk sends the qualify packets just fine. |
| 09:19.52 | kaldemar | peektoseen: if you don't see the packets with tcpdump, that's an error on tcpdump usage on your part. |
| 09:20.37 | peektoseen | kaldemar: in 'sip debug' mode I also dont see any OPTIONS |
| 09:21.43 | peektoseen | only REGISTER, 200 OK, 401, INVITE, etc |
| 09:24.14 | peektoseen | here is the log: http://pastebin.com/raw.php?i=FwfUu5RN |
| 09:24.41 | kaldemar | peektoseen: that's not sip debug. |
| 09:24.57 | peektoseen | I know it |
| 09:25.12 | kaldemar | but it shows that peer have become reachable. which again tells that it has sent qualify packets and gotten responses. |
| 09:25.12 | peektoseen | whait a moment... |
| 09:30.49 | *** join/#asterisk mirela666 (~mirko.bra@93-87-236-177.dynamic.isp.telekom.rs) |
| 09:31.22 | peektoseen | kaldemar: here is a sip debug: http://pastebin.com/raw.php?i=QJuhvs4L |
| 09:32.45 | Neoti | gnu_d: give me a sec to find the code |
| 09:32.55 | gnu_d | Neoti: thanks |
| 09:34.39 | Neoti | gnu_d: in mod*.conf add "noload => res_srtp.so" should work..... :) |
| 09:36.46 | kaldemar | peektoseen: and? |
| 09:38.47 | gnu_d | Neoti: alas, same error: Rejecting secure audio stream without encryption details: audio 49889 RTP/SAVPF 109 0 8 101 |
| 09:54.37 | *** join/#asterisk mirela666 (~mirko.bra@93-87-236-177.dynamic.isp.telekom.rs) |
| 10:04.53 | gnu_d | Where do I write the CHANNEL(secure_bridge_signaling) .... line ? |
| 10:11.10 | Neoti | gnu_d: did you do something like service asterisk restart or asterisk -rx "reload" etc ? |
| 10:13.57 | gnu_d | Neoti: I just kileld the asterisk process, and then start it again |
| 10:22.46 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 10:23.02 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:41.15 | *** join/#asterisk mirela666 (~mirko.bra@93-87-236-177.dynamic.isp.telekom.rs) |
| 10:43.03 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 10:45.04 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 10:46.04 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 10:46.07 | v0lZy | Hi |
| 10:47.19 | v0lZy | Can anyone offer any explanation as to why setting to allow only alaw would cause jitter/noise over all phones hooked up to my asterisk, but if those phones pickup the call and then put it on hold and resume it, jitter/noise goes away? |
| 10:47.35 | v0lZy | also, sound is perfectly clear on outgoing calls |
| 10:47.44 | v0lZy | this only happens when receiving from 1 ITSP |
| 10:50.35 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-37-249.dynamic.qsc.de) |
| 10:54.25 | *** join/#asterisk kazak1377 (c169f90a@gateway/web/cgi-irc/kiwiirc.com/ip.193.105.249.10) |
| 10:54.32 | davlefou | hi, |
| 10:54.56 | davlefou | regexten seems have no effect on my asterisk 1.8. |
| 10:58.33 | *** join/#asterisk velus (~velus@unaffiliated/velus) |
| 10:58.53 | velus | hello is anyone about? |
| 11:00.08 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 11:01.29 | *** join/#asterisk kresp0 (~kresp0@1.Red-83-50-237.dynamicIP.rima-tde.net) |
| 11:09.27 | davlefou | v0lZy, i don't undstand your problem. |
| 11:10.11 | Neoti | gnu_d: in the phone/software also turn off srtp..... |
| 11:10.32 | Neoti | gnu_d: i know in snoms you can disable this ... |
| 11:10.42 | Neoti | gnu_d: unless you have it working |
| 11:10.59 | velus | brb |
| 11:13.50 | v0lZy | davlefou: its a weird problem :D |
| 11:14.24 | v0lZy | davlefou: I have 2 telephon service providers... 1 goes through copper ISDN lines, the other goes through internet |
| 11:15.01 | v0lZy | the one that comes in on ISDN is connected to a gateway, so that it fits in with VOIP (gateway translates analog to voip) |
| 11:15.27 | v0lZy | the one that comes in on the internet goes through a firewall and NAT. |
| 11:15.49 | v0lZy | the router has its own itnerface dedicated for voip communication... so WAN <-> VOIP subnet |
| 11:16.07 | v0lZy | ISDN gateway and asterisk are in the same subnet |
| 11:16.29 | v0lZy | so, path for calls from ITSP is WAN->VOIP->SWITCH->asterisk |
| 11:16.50 | v0lZy | path for calls from standard telephony is ISDN->Gateway->same SWITCH as above-> asterisk |
| 11:16.52 | v0lZy | now |
| 11:17.05 | v0lZy | when i receive calls from ISDN lines, everything is peachy |
| 11:17.29 | v0lZy | when i receive calls from WAN, i get 2 issues |
| 11:17.58 | v0lZy | issue no. 1: if a callee answers the ringing phone with his phone, he hears static/ |
| 11:18.48 | v0lZy | if the callee then presses the transfer button, caller is played music on hold and doesnt hear any static... the callee then cancels the transfer (doesnt transfer the caller), and there is no more static |
| 11:19.43 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 11:19.51 | v0lZy | issue 2: if the callee isnt at his desk and another user picks up the callee's ringing extension, that user can not hear the caller, but the caller can hear the user. |
| 11:20.12 | v0lZy | on internal calls and on ISDN lines lines, all this works without issue |
| 11:23.53 | v0lZy | the jitter/static problem goes away if i enable ulaw and gsm codecs... |
| 11:24.37 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 11:37.59 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 11:39.21 | *** join/#asterisk mirela666 (~mirko.bra@93-87-236-177.dynamic.isp.telekom.rs) |
| 11:39.58 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 11:40.43 | lnb_ | anyone here good with syntax for extensions? Have an issue where after enterning numbers like a pin, should hear specific recordings. Instead most times hear the female pbx voice 'goodbye' |
| 11:40.57 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:40.59 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:41.01 | lnb_ | acts like it skips the lines |
| 11:42.06 | velus | hello i have asterisks now with freepbx isntalled and need help setting it up can anyone help me please# |
| 11:46.28 | *** join/#asterisk Dpunkt (~Dpunkt@p5DE5F660.dip0.t-ipconnect.de) |
| 11:53.40 | bacobart | velus: try #freepbx |
| 11:53.59 | velus | ty |
| 11:57.38 | v0lZy | lnb_: show us the relevant part of the dialplan |
| 11:57.45 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 11:57.47 | *** part/#asterisk velus (~velus@unaffiliated/velus) |
| 11:58.53 | lnb_ | v0lZy: there are several lines above but i dont want to flood the channel... |
| 11:59.07 | lnb_ | v0lZy: exten => *2000,n,Goto(startup) |
| 11:59.24 | lnb_ | v0lZy: exten => *2000,n(good),ExecIf($["${stat}"="1"]?Playback(custom/you-have-successfully-punched-in)) |
| 11:59.38 | lnb_ | v0lZy: exten => *2000,n,ExecIf($["${stat}"="2"]?Playback(custom/you-have-successfully-punched-out)) |
| 11:59.43 | v0lZy | lnb_: use bpaste.net |
| 11:59.55 | lnb_ | ok |
| 12:01.23 | lnb_ | v0lZy: http://pastebin.ca/2461144 |
| 12:02.14 | lnb_ | what's happening is it is skipping the ?playback of the two wav files (depending on login or log out) |
| 12:02.34 | lnb_ | the cli shows its being executed but no one hears it |
| 12:03.04 | lnb_ | they only hear the 2nd last line .. playback(goodbye) |
| 12:03.56 | v0lZy | since they are custom files |
| 12:04.05 | v0lZy | im gonna guess you didnt encode them correctly |
| 12:04.17 | v0lZy | try playing some of the standard files |
| 12:04.24 | v0lZy | see if it works then |
| 12:04.36 | v0lZy | if thats the case, then your custom files need to be properly encoded for asterisk to play them |
| 12:04.54 | lnb_ | they are encoded right.. |
| 12:05.11 | lnb_ | you-have-successfully-punched-out.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 12:05.28 | lnb_ | same with other wav file |
| 12:07.44 | v0lZy | just try with other files |
| 12:07.48 | v0lZy | to see if they get skipped to |
| 12:07.49 | v0lZy | if they do |
| 12:07.59 | v0lZy | then maybe instead of execif use gotoif |
| 12:08.13 | v0lZy | and setup some spaghetti code to make it work |
| 12:10.10 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 12:15.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:16.04 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 12:21.00 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 12:24.09 | lnb_ | v0lZy: is it possible this is the problem: |
| 12:24.11 | lnb_ | -- Launched AGI Script /var/lib/asterisk/agi-bin/timeclockinit.php |
| 12:24.12 | lnb_ | [2013-10-02 08:23:18] ERROR[11754][C-00000b8c]: utils.c:1187 ast_carefulwrite: write() returned error: Broken pipe |
| 12:24.54 | lnb_ | those errors are right before where it would play the wav file |
| 12:25.04 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 12:33.57 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 12:54.23 | bombev | hello where is the music on hold directory ? |
| 12:59.57 | v0lZy | lnb_: sorry, dont know, above my knowledge |
| 13:00.42 | [TK]D-Fender | bombev: wherever you pointed it to |
| 13:04.31 | v0lZy | [TK]D-Fender: Hi! Could I trouble you about a weird issue i'm havnig. My ITSP suggest theres something wrong between asterisk and the phones, although if my users dont hear the people calling-in through the ITSP, wouldnt that imply a problme on the ITSP side? |
| 13:04.38 | v0lZy | fact is, if my users call out, everything works ok |
| 13:04.55 | v0lZy | if my users get caled in, i get one way audio AFTER pickup. |
| 13:05.30 | v0lZy | (the phone thats being called gets 2 way audio, but a call that is picked up is just 1 way audio where external caller gets my users voice, but my users dont get theirs. |
| 13:05.39 | [TK]D-Fender | [09:04]v0lZy[TK]D-Fender: Hi! Could I trouble you about a weird issue i'm havnig. My ITSP suggest theres something wrong between asterisk and the phones, although if my users dont hear the people calling-in through the ITSP, wouldnt that imply a problme on the ITSP side? <- EVERY piece along the way could be at fault |
| 13:07.06 | v0lZy | [TK]D-Fender: When I get a call through the ITSP line, if the person thats being called picks up the phone, everything works ok. If another users picks up the ringing extension from another phone, 1 way audio for the user who can only project their voice, but doesnt hear the caller |
| 13:07.13 | [TK]D-Fender | And as a general rule... providers tech works fine. The users' pieces of these pictures are almost always the point at fault |
| 13:08.00 | v0lZy | [TK]D-Fender: What i find puzzling is this, if i set all my phones to use only 1 codec (alaw), and i set my asterisk to report only alaw, this all works, however, I get static on the phone |
| 13:08.06 | Katty | hello good morning how to asterisk please |
| 13:08.35 | v0lZy | [TK]D-Fender: and this static stuff IS strange in the sense that if my user then puts the caller on hold for a second, then resumes the conversation, the static goes away |
| 13:08.59 | v0lZy | [TK]D-Fender: however, in no situations do i ever get any difficulties with any sound quality or anything else when my users are the ones that dial out. |
| 13:10.55 | v0lZy | [TK]D-Fender: One thing i did notice according to core set verbose 9 output, when the pickup is made, i get 'zombies' |
| 13:11.28 | [TK]D-Fender | v0lZy: Time for a 180 in your approach... |
| 13:12.02 | *** join/#asterisk kresp0 (~kresp0@1.Red-83-50-237.dynamicIP.rima-tde.net) |
| 13:12.09 | v0lZy | which would be? |
| 13:13.46 | [TK]D-Fender | v0lZy: You are giving a huge story and not showing what is actually happening. We can't tell if anything funny is happening or if you've misconfigured your side. |
| 13:13.57 | [TK]D-Fender | v0lZy: In sort, all talk, no debug. |
| 13:14.21 | [TK]D-Fender | v0lZy: If you'd like to try to fix it, just show the calls and configs |
| 13:14.38 | v0lZy | [TK]D-Fender: with sip debug? |
| 13:14.57 | [TK]D-Fender | v0lZy: You have voice issues... so do you think SIP is important there? |
| 13:15.12 | Katty | fender. |
| 13:15.18 | Katty | be nice, dear. |
| 13:15.29 | [TK]D-Fender | Katty: I am... I haven't sworn in channel in AGES! |
| 13:15.57 | [TK]D-Fender | Katty: I'm prodding neurons! I'm popping tags! |
| 13:16.02 | v0lZy | [TK]D-Fender: well, its audio problems so im guessing RTP |
| 13:16.12 | [TK]D-Fender | Katty: THIS IS FUCKING AWESOME! |
| 13:16.23 | [TK]D-Fender | resets the "Swear Counter" |
| 13:16.45 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:16.48 | v0lZy | hm |
| 13:16.49 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:16.50 | [TK]D-Fender | DOESN'T have to increase the "Verbal Hostility" one however! |
| 13:17.05 | v0lZy | i just reset my switch and sure enough, i cant reach my phones anymore.. |
| 13:17.11 | [TK]D-Fender | v0lZy: SIP ***negotiates*** RTP |
| 13:17.31 | [TK]D-Fender | v0lZy: Who says it's negotiated right? That filure is the #1 screwup for 1-way voice |
| 13:19.25 | v0lZy | hm, yeah |
| 13:19.40 | v0lZy | but now i have a weird issue where for some reason, i cant even reach my phones anymore |
| 13:19.46 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 13:19.50 | [TK]D-Fender | v0lZy: From where? |
| 13:19.55 | v0lZy | from ITSP |
| 13:20.07 | v0lZy | i can ping my phone |
| 13:20.13 | [TK]D-Fender | v0lZy: As in? |
| 13:20.25 | v0lZy | i can dial out with my phone |
| 13:20.28 | v0lZy | but i cant dial in |
| 13:20.48 | v0lZy | [TK]D-Fender: gimme a sec to draw this |
| 13:20.50 | [TK]D-Fender | v0lZy: For all of this story you aren't giving any technical details at all |
| 13:21.05 | v0lZy | [TK]D-Fender: I'll give an overview, one moment |
| 13:22.17 | [TK]D-Fender | wonder why 2 lines of text to describe where things are requires a drawing. |
| 13:23.43 | *** join/#asterisk ickmund (uid11004@gateway/web/irccloud.com/x-epfpyhauatjqcchb) |
| 13:25.46 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 13:26.13 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 13:27.53 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 13:31.41 | v0lZy | [TK]D-Fender: http://picpaste.com/mynet-3md40ccX.png |
| 13:32.03 | v0lZy | thats how things are configured |
| 13:32.56 | v0lZy | whenever I call out, regardless if im using ITSP or ISDN TSP, everything works. |
| 13:33.04 | v0lZy | whenever I'm called in through ISDN TSP, everything works |
| 13:34.06 | v0lZy | whenever i am called in through ITSP, if the extension thats being called is the extension that answers, everything works (except at the moment, i for some reason cant reach any extensions.. i can ping the phones the managed switch, the pbx etc... but for some reason, dialing in trhough ITSP, i cant reach the phones) |
| 13:34.18 | [TK]D-Fender | v0lZy: the top 4 IP's you list should have been /32 |
| 13:34.51 | [TK]D-Fender | v0lZy: I have no idea what that giant block in the MIDDLE is. |
| 13:34.54 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 13:34.56 | v0lZy | router :D |
| 13:35.00 | [TK]D-Fender | v0lZy: And I don't see configs or debug. |
| 13:35.05 | v0lZy | [TK]D-Fender: erm, i wanted to indicate that they are in the same subnet |
| 13:35.29 | v0lZy | wan is my wan link, voip is my subnet for voip, lan is for my computers... |
| 13:35.35 | v0lZy | im just trying to illustrate, bare with me please |
| 13:35.44 | [TK]D-Fender | v0lZy: You could have said that in one sentence instead of a huge story then a 10-minute delayed drawing |
| 13:36.05 | [TK]D-Fender | v0lZy: And after all that you've still skipped actual technical details. |
| 13:36.44 | v0lZy | everything generally works up until the point where the extension that is called is not the extension that answers.... so if phone B picksup phone A's call, the caller (coming in through ITSP) hears phone B's audio, but phone B doesnt hear the caller's audio. However, if phone A would answer, audio is OK both ways. |
| 13:37.13 | v0lZy | What I dont understand is, why pickup kills incoming audio for the person that does the pickup |
| 13:37.38 | v0lZy | and this only happens with ITSP, doesnt happen when people dial eachother within the local setup |
| 13:37.46 | v0lZy | and it doesnt hapepn for calls received via ISDN TSP... |
| 13:38.45 | [TK]D-Fender | v0lZy: It seems I have not gotten through to you and therefor cannot help. |
| 13:38.53 | v0lZy | Now the fact that before i reset the switch, i could call into phone A through ITSP, and now after i did the reset, I cant... puzzles me |
| 13:39.09 | [TK]D-Fender | moves on to other matters |
| 13:39.13 | v0lZy | [TK]D-Fender: u did get to me, but im dealing with 2 issues at the same time |
| 13:39.23 | v0lZy | [TK]D-Fender: I cant run a sip debug |
| 13:39.30 | v0lZy | [TK]D-Fender: because im not reching my pbx |
| 13:39.44 | v0lZy | [TK]D-Fender: im in asterisk cli now, and nothing is triggered at all when im making a call... |
| 13:40.27 | *** join/#asterisk serafie (~erin@nat/digium/x-pfkchdkntmukbcnk) |
| 13:40.56 | v0lZy | gonna go swap this for a dumb switch, see if its the switch or something |
| 13:40.58 | v0lZy | hold on |
| 13:46.28 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-qwbjghrzehtopkpk) |
| 13:46.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:51.18 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:54.52 | *** join/#asterisk tristero (~al.f.zero@unaffiliated/transfinite) |
| 13:56.28 | Katty | waves to mjordan |
| 13:56.31 | tristero | I have a bunch of patterns in extensions.conf for outgoing calls (local, long distance, 911, etc.) Now I want to allow any of them to be prefixed by *82 to allow callerID. How can I do that without duplicating all the existing patterns? |
| 13:59.00 | [TK]D-Fender | tristero: You have to. |
| 13:59.25 | [TK]D-Fender | tristero: There is no regex for a variable prefix in *'s dialplan pattern matching |
| 14:00.35 | v0lZy | darn |
| 14:00.41 | v0lZy | its not the switch thats at fault |
| 14:00.45 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:00.45 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:00.48 | v0lZy | and its not a phone setting thats at fault |
| 14:01.01 | v0lZy | because i DIDNT touch the phones, certianly not all of them |
| 14:01.04 | Katty | waves to sruffell |
| 14:01.04 | v0lZy | so its either asterisk |
| 14:01.06 | v0lZy | or firewall |
| 14:01.34 | sruffell | hello |
| 14:01.43 | Katty | sruffell: how're you this morning dear |
| 14:01.56 | sruffell | not too bad. Yourself? |
| 14:02.30 | SuperNull | is there no way to Verbose(,) with color ? if i wanted to output my own color for ease of reading |
| 14:03.11 | Katty | sruffell: good good, still waking up. |
| 14:03.18 | Katty | sruffell: did you enjoy your trip out east? |
| 14:03.35 | sruffell | thinking |
| 14:03.39 | sruffell | what trip was that? |
| 14:03.42 | Katty | raleigh |
| 14:03.55 | sruffell | I think you may have me confused for someone else. I have not taken a trip to Raleigh. |
| 14:04.14 | Katty | that is possible. |
| 14:04.19 | Katty | i'm not even halfway through my soda yet! |
| 14:04.23 | sruffell | lol |
| 14:04.26 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 14:04.32 | v0lZy | [TK]D-Fender: == Spawn extension (AppDirectPickup, DirectPickup, 1) exited non-zero on 'SIP/44-000001a1<ZOMBIE>' |
| 14:04.32 | v0lZy | <PROTECTED> |
| 14:04.51 | v0lZy | 44 is the phone I was calling, 16 is the phone that was picking up from 44 |
| 14:05.03 | v0lZy | 16 doesnt get the caller's audio |
| 14:05.22 | Katty | sruffell: well please disregard my lunacy this morning >.< |
| 14:05.55 | v0lZy | i get the same thing with internal calls |
| 14:05.56 | v0lZy | - Executing [DirectPickup@AppDirectPickup:1] Pickup("SIP/16-000001a5", "44@PICKUPMARK") in new stack |
| 14:05.57 | v0lZy | <PROTECTED> |
| 14:05.57 | v0lZy | <PROTECTED> |
| 14:06.03 | v0lZy | so i suspect this zombie thign has nothing to dow ith anything |
| 14:06.49 | SuperNull | rly. |
| 14:07.08 | wdoekes | SuperNull: NoOp(^[[33;1mtest^[[0m) where ^[ is 0x1b (you can type it using ctrl-V ESC) |
| 14:07.26 | SuperNull | wdoekes .. that is awesome. |
| 14:08.43 | SuperNull | sounds like i might have to use vi for that. :-/ |
| 14:09.05 | wdoekes | SuperNull: http://en.wikipedia.org/wiki/ANSI_escape_code#CSI_codes <-- 1=bold, 33 = 30+yellow |
| 14:12.07 | SuperNull | wtf. how do i generate ctrl+v ESC over ssh ? (winderz here) |
| 14:13.11 | Katty | simply right click into putty |
| 14:13.22 | Katty | with esc |
| 14:13.50 | tzanger | lol someone in .it trying to break into my asterisk server with a SIP registration to test:test |
| 14:13.59 | SuperNull | let me get putty, we use SecureCRT here. |
| 14:15.04 | SuperNull | seriously. |
| 14:15.51 | tristero | [TK]D-Fender: I was wondering if there was any way to have two levels of extensions, one to detect the *82, and then jump somehow to matching the rest? |
| 14:18.22 | SuperNull | screw it, im remapping the keys in SecureCRT to just output escape code off insert button lol. |
| 14:21.34 | *** join/#asterisk davlefouAMD (~david@197.15.67.89) |
| 14:22.24 | Katty | infobot: seen eppigy |
| 14:22.30 | infobot | eppigy <~Dave@snugglenets.com> was last seen on IRC in channel #asterisk, 684d 20h 25m 50s ago, saying: 'oh you fancy huh'. |
| 14:22.48 | Katty | it doesn't feel like 2 years :< |
| 14:23.26 | SuperNull | 2 years is .. pretty much never coming back on IRC terms .. lol |
| 14:23.38 | Katty | you don't know that! |
| 14:23.47 | Katty | but yeah, you're probably right. |
| 14:23.53 | SuperNull | been using irc for 13 + years. |
| 14:24.00 | Katty | infobot: seen jaytee |
| 14:24.01 | infobot | jaytee <~jforde051@unaffiliated/jaytee> was last seen on IRC in channel #asterisk, 315d 16h 20m 1s ago, saying: 'slav3_kitten, know the feeling. hate it when that happens'. |
| 14:24.01 | SuperNull | maybe longer. |
| 14:24.13 | Katty | pouts |
| 14:24.25 | SuperNull | weird last messages tho.. |
| 14:24.30 | Katty | jaytee was so much fun. he fed twinkies to elephants. |
| 14:25.42 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 14:26.27 | *** join/#asterisk velus-universe (~velus@unaffiliated/velus) |
| 14:27.05 | wdoekes | SuperNull: you could append it to a file and then just "copy-paste" it from there (using the keyboard) |
| 14:27.20 | wdoekes | printf '\x1b\n' >> /etc/asterisk/extensions.conf |
| 14:28.43 | wdoekes | if you're using vi, it's a matter of hovering over it and doing 'x', now you can 'p' it wherever you want |
| 14:30.38 | Qwell | wdoekes: what is this sorcery? |
| 14:31.50 | file | not my sorcery. |
| 14:32.12 | wdoekes | SuperNull wanted ansi colors in his NoOp/Verbose |
| 14:32.17 | Qwell | oh |
| 14:32.33 | Qwell | I thought you were pasting with printf. My mind was blown for a bit there. |
| 14:32.56 | wdoekes | SuperNull had trouble doing ctrl-V ESC |
| 14:33.02 | wdoekes | hence the workaround |
| 14:33.57 | SuperNull | wdoekes was trying to do a shell_func .. just to test. |
| 14:34.12 | SuperNull | failed. keeps bitching it needs parens... when it doesnt. |
| 14:35.51 | Qwell | Katty: Has squirrelcam been shutdown, like pandacam? |
| 14:36.17 | Katty | checks |
| 14:36.29 | Katty | oh. |
| 14:36.32 | Katty | sec |
| 14:36.35 | Katty | remotes into server |
| 14:36.51 | Qwell | is it actually down? I totally made that up to be funny. |
| 14:37.03 | Katty | it is off air, yes. |
| 14:37.08 | Qwell | damn, I'm good |
| 14:37.14 | Katty | you sure are |
| 14:37.20 | Qwell | twss? |
| 14:37.28 | Katty | back online! |
| 14:37.30 | Katty | infobot: crittercam |
| 14:37.30 | infobot | i heard crittercam is Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
| 14:37.44 | *** join/#asterisk Prosouth__ (~sabayonus@62-2-198-100.static.cablecom.ch) |
| 14:37.54 | Katty | but no, the government shutdown has not affected my squirrels. |
| 14:37.55 | *** join/#asterisk Defraz (~Defraz@209.141.122.71) |
| 14:38.11 | Katty | i've had a couple friends cancel for one of my parties, due to not going to drill tho :< |
| 14:38.34 | *** join/#asterisk PLMg (PLMg@78.96.151.225) |
| 14:39.16 | PLMg | hey, what was the command to see number of recieved calls of an extension? |
| 14:40.06 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
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| 14:50.06 | [TK]D-Fender | [10:15]tristero[TK]D-Fender: I was wondering if there was any way to have two levels of extensions, one to detect the *82, and then jump somehow to matching the rest? <- you don't have to duplicate the full contents... just the initial match and the GOTO the other portion of your dialplan for actual processing |
| 14:51.17 | jeev | iax has been really funky all of last night, this has happened before. it goes reachable, 26ms, then unreachable, 10xx ms. for just a few seconds, then comes back. i can't get icmp to react this way |
| 14:55.22 | SuperNull | wdoekes i keep getting .. pbx_load_config: No closing parenthesis found? ' NoOp(at ..... wtf. |
| 14:59.28 | [TK]D-Fender | SuperNull: You might want to fix your syntax errors... |
| 15:00.43 | SuperNull | working on not doing retard stuff.. one sec.. lol |
| 15:02.25 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-nmqddouiaxybriiv) |
| 15:02.25 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:07.04 | jeev | i had qualify=25000 on an iax peer, i changed it to 50000, now it doesn't lag off.. it seems like a fix for the disconnection but it's just a bandaid, right ? |
| 15:08.31 | SuperNull | [TK] http://imgur.com/kdZKRpe |
| 15:09.21 | [TK]D-Fender | SuperNull: YaY aNsI! |
| 15:09.30 | SuperNull | ... it doesnt work. |
| 15:09.38 | SuperNull | as i said bitches of no closing parens. |
| 15:12.44 | SuperNull | maybe it works on 10+ not 1.8? |
| 15:17.07 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:17.51 | SuperNull | any of you guys have this working ? or is this just a 'should work ' type of thing.. |
| 15:22.06 | [TK]D-Fender | SuperNull: What... shoving ANSI into dialplan lines? |
| 15:22.50 | SuperNull | yeah.... |
| 15:22.56 | SuperNull | it doesnt like the semi-colon.. |
| 15:22.57 | SuperNull | lol |
| 15:22.58 | SuperNull | wtf. |
| 15:23.42 | SuperNull | ex: exten => s,n,NoOp(^[[33);1mtest^[[0m .... with garbage on the end will load.. YET exten => s,n,NoOp(^[[33;)1mtest^[[0m wont. |
| 15:24.07 | SuperNull | some how the semi-colon is fubaring something... which kills the idea lol |
| 15:26.13 | SuperNull | if i remove the semi-colon only it works. |
| 15:26.23 | SuperNull | sorry.. it loads. it doesnt work of course. |
| 15:27.01 | [TK]D-Fender | <PROTECTED> |
| 15:27.28 | SuperNull | ? |
| 15:27.37 | [TK]D-Fender | what's not clear? |
| 15:27.42 | [TK]D-Fender | you have stuff AFTER the ) |
| 15:27.52 | SuperNull | irrelative to it loading.. |
| 15:27.56 | SuperNull | i can remove it so you can see. |
| 15:28.13 | [TK]D-Fender | Not necessarily irrelevent to its looking like a fail waiting to happen |
| 15:29.35 | SuperNull | removed the crap. it still bitched. |
| 15:30.04 | wdoekes | SuperNull: escape the semi with a backslash |
| 15:30.21 | [TK]D-Fender | Ah yes... ; as COMMENT delimiter |
| 15:30.22 | SuperNull | let me try that. |
| 15:30.27 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 15:30.41 | [TK]D-Fender | thus commenting out the closing quote |
| 15:30.41 | SuperNull | yeah but .. wtf it thinks comment mid 'string' (this is why asterisk should require " " or ' ') |
| 15:30.47 | *** join/#asterisk hypknight (~hypknight@unaffiliated/hypknight) |
| 15:30.52 | wdoekes | (sorry, I use static realtime db for my dialplan, I don't have comments there) |
| 15:31.01 | [TK]D-Fender | SuperNull: Asterisk wouldn't know data types if they ran up and bit it in the butt :p |
| 15:31.16 | SuperNull | yet it can deal with string manipulation. |
| 15:32.16 | [TK]D-Fender | SuperNull: EVERYTHING is dumb text. When you get down to it the raw concept of "math" is more of a hack |
| 15:32.29 | SuperNull | agreed. |
| 15:33.00 | SuperNull | asterisk dialplan for 'advanced' things is always quirky .. |
| 15:33.19 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 15:34.47 | SuperNull | that worked.. yey. |
| 15:37.52 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-111-87.user.veloxzone.com.br) |
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| 15:58.14 | SuperNull | anyone every have calls drop at EXACTLY 15 minutes? odd... considering the configuration should be identical and ast versions identical. |
| 16:02.52 | *** join/#asterisk Galaxor (~Galaxor@208.67.250.157) |
| 16:03.39 | Galaxor | If I log in to asterisk via sip, and I make a call to another sip client, does all the traffic go through the asterisk machine, or do the sip clients negotiate with each other? |
| 16:03.55 | [TK]D-Fender | Galaxor: Depends how you configured * |
| 16:04.21 | *** join/#asterisk navaismo (~navaismo@189.241.90.55) |
| 16:05.10 | Galaxor | [TK]D-Fender: Oh okay, good. It *can* be configured the way I want, then. |
| 16:05.24 | [TK]D-Fender | Galaxor: Dependsing on the precise circumstances maybe |
| 16:05.49 | Galaxor | What happens if both clients are behind nats? |
| 16:05.57 | Galaxor | Can asterisk punch holes for them? |
| 16:06.35 | [TK]D-Fender | Don't bet on it |
| 16:11.30 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 16:12.03 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 16:13.11 | qakhan | can we put space and . in veriable values, like a=1234 i want to setup a= 1 . 2. 3. 4 |
| 16:13.28 | [TK]D-Fender | qakhan: Yes |
| 16:14.06 | qakhan | which function i can use for this |
| 16:14.42 | [TK]D-Fender | qakhan: What do you mean "function"? You just asked if you could spaces and dots in a variable. YES, you CAN. So go SET YOUR VARIABLE |
| 16:14.50 | [TK]D-Fender | qakhan: SET <- |
| 16:17.47 | qakhan | ok let me explain. i am getting date from caller in veriale date=10022013 and tts is speacking to callers what he entered. |
| 16:19.54 | wdoekes | qakhan: sooo... you don't want "10 02 2013", you want Say(10) Say(02) ... |
| 16:20.05 | wdoekes | Say(${var:0:2}) ... ? |
| 16:20.16 | qakhan | yes wdoekes |
| 16:21.00 | wdoekes | reminds me of ${LENGTH(${CUT(mystring,charImLookingFor,1)})} |
| 16:21.12 | *** part/#asterisk velus-universe (~velus@unaffiliated/velus) |
| 16:21.36 | wdoekes | while you wanted CUT in the first place, not charindex |
| 16:22.15 | qakhan | no i want to put . in date |
| 16:22.50 | qakhan | like 10. 02. 2013. |
| 16:24.43 | navaismo | use SET or ask caller one by one and use 3 different variables to store day month& year |
| 16:25.05 | navaismo | and then one global joining the 3 vars |
| 16:32.58 | [TK]D-Fender | [12:17]qakhanok let me explain. i am getting date from caller in veriale date=10022013 and tts is speacking to callers what he entered. |
| 16:33.09 | [TK]D-Fender | [12:22]qakhanno i want to put . in date [12:22]qakhanlike 10. 02. 2013. |
| 16:33.30 | [TK]D-Fender | qakhan: So set a NEW variable with the chopped up bits of the entered variable and dots. |
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| 16:40.27 | gnu_d | Neoti: Any other suggestions, also have you used jsSOP ever ? |
| 16:40.35 | gnu_d | SIP* |
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| 17:16.06 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 17:16.07 | v0lZy | hi |
| 17:16.18 | v0lZy | anyone around? I have an interesting problem (resolved my previous issues btw) |
| 17:16.41 | v0lZy | my ITSP sums up the billing based on IP |
| 17:17.10 | v0lZy | we now have a tennant that wants their own bill from ITSP rather than from us... but they want to use same asterisk box as we are using |
| 17:17.24 | v0lZy | now i can configure virtual ip's on my box |
| 17:17.32 | v0lZy | on the router i mean |
| 17:17.53 | v0lZy | but im hoping theres an easy way to force asterisk to use a different public ip |
| 17:18.23 | v0lZy | im thinknig along the lines of adding a virtual interface on top of the one that i already have |
| 17:18.49 | v0lZy | and then im trying to figure out how to force registration through a different public ip. |
| 17:20.07 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 17:23.23 | [TK]D-Fender | v0lZy: Forget about multi-homing like that. |
| 17:24.46 | v0lZy | [TK]D-Fender: can asterisk multihome on its own at all? |
| 17:24.51 | v0lZy | or must i trick it with a proxy or something |
| 17:25.44 | v0lZy | I was hoping to do some policy routing based on IP... like if asterisk is using IP 192.168.1.3 route request throug this public ip... if its doing from 192.168.1.4, route through other public ip.. |
| 17:27.34 | v0lZy | [TK]D-Fender: what are the alternatives? |
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| 17:29.57 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 17:30.08 | qakhan | Thank you guys |
| 17:31.00 | qakhan | i did exten => 1,n(setdate),Set(month=${date:0:2}) |
| 17:31.04 | qakhan | exten => 1,n,Set(day=${date:2:2}) |
| 17:31.08 | qakhan | exten => 1,n,Set(year=${date:4:8}) |
| 17:31.45 | boom^time | qakhan, what if they don't enter two digits for a month or day? |
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| 17:37.05 | *** join/#asterisk joesmoe (~joesmoe@admins.phreefilez.com) |
| 17:54.51 | [TK]D-Fender | boom^time: well I guess he should check that. which he's been told several times over the past few days of course. |
| 17:56.05 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 18:07.27 | qakhan | boom^time that is my next question. how we can check correct date is being entered |
| 18:07.49 | qakhan | i meant correct format. MM/DD/YYYY |
| 18:08.44 | boom^time | I'd start with a len check on your date variable to make sure it's 8 digits |
| 18:10.03 | boom^time | then after you chop it up do some simple comparisons GotoIf($[${month}>12]?invalid-month) |
| 18:11.06 | boom^time | I'd probably end it all with a nice verbal verification, "You chose January 1st, 1992. Is this correct?" |
| 18:11.27 | [TK]D-Fender | Seems to have a real problem with the idea of LOOKING at the value |
| 18:11.50 | [TK]D-Fender | Can't tell that month 13 is BAD? |
| 18:12.06 | [TK]D-Fender | No concept of GotoIf? (yes, he's used it plenty before) |
| 18:12.37 | [TK]D-Fender | Not even sure what is a valid date or not? |
| 18:12.42 | [TK]D-Fender | Serious problems..... |
| 18:16.06 | qakhan | accordig to boom^time description it looks nice and do able |
| 18:16.46 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 18:17.02 | qakhan | but there should be a function or app in dialplan which can check as we can do in php or other languages |
| 18:17.35 | Penguin | I'd prefer to be asked to input month first, then be asked for the day, then asked for the year. I would not want to enter it all as one string. |
| 18:18.33 | paulc | Good TUI design would prompt you for day, month, year separately, but allow experienced/repeat users to DTMF through the whole entry sequence seamlessly :-) |
| 18:18.41 | Penguin | If the month is entered at one digit, transform it to two. Same for the day. Ask for four digit year. |
| 18:19.39 | [TK]D-Fender | [14:16]qakhanbut there should be a function or app in dialplan which can check as we can do in php or other languages <- no need |
| 18:20.13 | paulc | +1 for a 4 digit year.. and -1 for any request to "enter XX digits, followed by the pound key" - fixed lengths don't need terminating, right?! |
| 18:20.16 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 18:20.18 | Penguin | GotoIf() does a pretty good job of checking. |
| 18:21.37 | Penguin | I think for an IVR, you really only need Read(), GotoIf(), and Playback() or BackGround(). |
| 18:21.37 | qakhan | but its takes time |
| 18:22.16 | qakhan | i am using Cepstral TTS for IVR |
| 18:22.51 | Penguin | Cepstral doesn't make IVRs. It only plays sounds. You still need the regular applications to make the IVR. |
| 18:23.02 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 18:23.39 | lnb_ | i have a dialplan that is failing. There are two possible recordings that the caller hears. In either event, the caller only hears the freepbx 'goodbye'. http://pastebin.ca/2461223 any help is greatly appreciated |
| 18:24.08 | *** join/#asterisk serafie (~erin@nat/digium/x-qqayllxsobxhxgkt) |
| 18:24.23 | [TK]D-Fender | [14:21]qakhanbut its takes time <- how many milliseconds is "too much" for you? |
| 18:24.53 | qakhan | [TK]D-Fender i am ok with it but you know callers |
| 18:26.16 | qakhan | ok i do some work on month day and year check. i will share it with you guys |
| 18:28.34 | [TK]D-Fender | qakhan: You have not even done the check and have a problem. Nor any basis to even guess that one could exist. |
| 18:32.30 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 18:32.56 | paulc | lnb_ Can you pastebin your dialplan for *2000? |
| 18:37.46 | qakhan | what strftime function does? |
| 18:40.28 | Penguin | What did "core show function STRFTIME" say it does? |
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| 19:20.02 | lnb_ | paulc: http://pastebin.ca/2461240 |
| 19:22.05 | lnb_ | paulc: if you were to dial that extension, you would hear 'enter your userid, enter your password. then upon correct credentials you would hear 'you have successfully punched in. but it doesnt play the successfully part, it just does the (Goodbye) |
| 19:22.52 | lnb_ | that code was used on an old elastix server and sent to me to work on 1.8 Asterisk |
| 19:23.51 | *** join/#asterisk toddf (todd@2001:470:817c:10:207:e9ff:fe10:a36c) |
| 19:23.59 | *** join/#asterisk rbd_ (~rbd@cpe-076-182-043-018.nc.res.rr.com) |
| 19:24.23 | rbd_ | hey guys...asterisk 11.4.0 ... where do I find MP3Player ... is there still an addons package for asterisk where it can be compiled from? |
| 19:26.02 | Penguin | If you end up having to recompile anyway, you may as well upgrade to current 11. |
| 19:27.29 | rbd_ | Penguin: eh...this is the version packaged with unimrcp that we use....I could try upgrading independently...but this is what they include |
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| 19:28.10 | rbd_ | ah, I see that app_mp3 is included in the base 11 source |
| 19:34.31 | lnb_ | trying to dial an extension, cli shows: WARNING[12378][C-00000cba]: ccss.c:1000 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead. |
| 19:34.36 | lnb_ | what is that ? |
| 19:35.40 | navaismo | macros are deprecated and replaced by subfunctions||subroutines |
| 19:35.51 | paulc | lnb_ From your pastebin, you're using GotoIf but then you're calling Playback directly, instead of jumping to a priority/extension etc. Also, it looks like your ${stat} isn't either of the values you're testing for, because your console log shows it as a 0 each time |
| 19:37.50 | lnb_ | i had before ExecIF |
| 19:40.52 | paulc | Use GotoIf.. then instead of Playback, go to a labelled priority that plays the right prompt, then Goto end or wherever afterwards |
| 19:41.51 | lnb_ | paulc: i will change it right now and and core reload |
| 19:42.34 | paulc | lnb_: "dialplan reload" would be enough.. |
| 19:44.01 | lnb_ | paulc: done |
| 19:44.24 | lnb_ | can a sound file be sent to paste bin? |
| 19:44.57 | lnb_ | paulc: does same thing |
| 19:45.13 | lnb_ | after entering credentials, hear (Goodbye) |
| 19:45.33 | paulc | lnb_: Do me a pastebin of your dialplan now, together with the console output, and I'll take a look for you |
| 19:45.49 | paulc | (I still half think your "stat" variable isn't being set properly) |
| 19:47.00 | lnb_ | any idea what should go there? |
| 19:48.17 | paulc | I'm rusty on "setting dialplan variables from within an AGI script" (I usually use CURL to pass stuff to/from web services).. but show me your dialplan + console output and we can verify/confirm that that is indeed the problem.. |
| 19:49.12 | lnb_ | you saw the dialplan :) |
| 19:56.06 | paulc | Yes, and I told you to change the GotoIf to use labels, instead of Playback.. did you make those changes? |
| 19:58.30 | *** part/#asterisk rsaffi (~rsaffi_@187.109.36.11) |
| 19:58.36 | lnb_ | labels? |
| 19:58.45 | lnb_ | no i changed it back to ExecIF |
| 19:58.50 | lnb_ | paulc: http://pastebin.ca/2461250 |
| 19:59.14 | lnb_ | i will put the dialplan up again (current one) |
| 20:00.11 | lnb_ | paulc: http://pastebin.ca/2461251 |
| 20:00.56 | lnb_ | it says it ran the playback , but i am not hearing it on the phone |
| 20:02.22 | paulc | lnb_: no, it's not.. it's telling you it didn't jump to the playback, because the digit before the question mark is 0.. you're falling through both GotoIf's |
| 20:02.43 | paulc | lnb_: See http://pastebin.ca/2461252 - I rewrote your dialplan a bit, to do both what I was saying you should do, and give you some extra debugging. |
| 20:03.10 | paulc | Tell me what your "stat=" line says when you run it.. if it says "stat=...." with nothing between the dots, your AGI isn't setting the return variable correctly. |
| 20:05.12 | lnb_ | one sec phone call |
| 20:20.22 | lnb_ | paulc: how do i run the stat line? |
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| 20:29.23 | paulc | lnb_: I don't understand the question. Take a look at the diaplan - it's going to NoOp(stat=..${stat}..) - so you can see exactly what your variable is being set to. I'm betting a dollar it's empty. |
| 20:30.35 | lnb_ | paulc: i am just going to install now what you sent to pastebin |
| 20:30.38 | lnb_ | one sec |
| 20:34.54 | [TK]D-Fender | heading home, BBIAB |
| 20:35.50 | lnb_ | paulc: http://pastebin.ca/2461259 |
| 20:38.34 | paulc | lnb_ go look at your output at line 44.. and line 49.. your script is passing back a 3, but your dialplan doesn't know what to do with it.. |
| 20:39.29 | lnb_ | that should be a 2? |
| 20:40.22 | lnb_ | wait a sec |
| 20:40.24 | lnb_ | that mean |
| 20:40.25 | lnb_ | s |
| 20:40.33 | navaismo | maybe missing ${} |
| 20:40.34 | lnb_ | the .php file has the wrong number? |
| 20:40.46 | lnb_ | i did write this stuff |
| 20:41.00 | navaismo | at line 15 |
| 20:41.04 | lnb_ | as i said before it was sent to me by old IT guy |
| 20:41.13 | lnb_ | did NOT write this i mean |
| 20:41.24 | *** part/#asterisk navaismo (~navai_000@189.241.90.55) |
| 20:42.02 | lnb_ | doesn't auth need to be ${auth} |
| 20:43.52 | lnb_ | changed it to ${auth} and now it tells me invalid userid/pin |
| 20:43.58 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:49.57 | *** join/#asterisk DanielSa (~daniel@198.147.23.156) |
| 20:51.47 | *** join/#asterisk navaismo (~navaismo@189.241.90.55) |
| 20:53.11 | paulc | lnb_: I have to go to a meeting, back in 90ish.. I think stat is being set to 3 at the start of the dialplan.. so it definitely looks like the AGI isn't setting it.. I'd do "core show application agi" and see what it says about arguments/parameters.. then look inside the script you're calling for more clues.. |
| 20:55.54 | lnb_ | ok |
| 20:56.09 | lnb_ | paulc: thank you for your help! |
| 21:01.33 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 21:03.01 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:03.43 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 21:04.47 | anonymouz666 | [TK]D-Fender has joined! |
| 21:05.30 | drmessano | He needs a bouncer |
| 21:06.00 | drmessano | I'm surprised he doesn't use ZNC with all the home/work flipping |
| 21:06.28 | drmessano | ZNC is great.. I am actually only here for 11 minutes per day, but it looks like I care.. A LOT |
| 21:06.32 | blizzow | I just created a new SIP extension and enabled voicemail on that extension. I try and enter the mailbox via *97 or *99 and am getting stonewalled. *97 doesn't accept the password that was set. *99 just loops a message saying "to listen to it press 1, to re-record press #" I watch the log and see: "ast_streamfile failed on SIP/55005-0001d257 for /var/spool/asterisk/tmp/55005-ivrrecording,m,en,macro-systemrecording" |
| 21:06.34 | blizzow | /var/spool/asterisk/tmp/ exists and is owned by asterisk:asterisk. I even touched the file as the asterisk user and still get that error in the log. |
| 21:06.57 | blizzow | I tried deleting and re-creating the extension to no avail. |
| 21:07.18 | blizzow | Anyone have suggestions or ideas what might be causing this problem? |
| 21:07.42 | blizzow | Other previously existing extensions seem to work okay. |
| 21:08.42 | [TK]D-Fender | drmessano: Couldn't be bothered most of the time... though maybe not a bad idea for the theory of having to auth repeatedly on crash, etc |
| 21:08.52 | [TK]D-Fender | drmessano: Might get around to setting one up |
| 21:09.31 | [TK]D-Fender | blizzow: PASTEBIN the actual call |
| 21:09.52 | drmessano | [TK]D-Fender, it's annoying to bother setting one up, but useful once implemented. |
| 21:11.30 | *** part/#asterisk DanielSa (~daniel@198.147.23.156) |
| 21:12.18 | [TK]D-Fender | drmessano: Consistent logs and saving channel re-auth (nickserv ghost, etc) might make it worthwhile, especially if it logs things nicely. |
| 21:13.31 | blizzow | [TK]D-Fender: here's the log of the attempt at accessing voicemail. http://pastebin.com/PC0f86D3 |
| 21:15.34 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:18.31 | drmessano | [TK]D-Fender, the only thing I hated about it was that X-Chat and even prior versions of Hexchat displayed the timestamps of the buffer dump on reconnect as inline text with each message. They've apparently figured that out with Hexchat and fixed it |
| 21:18.52 | [TK]D-Fender | blizzow: [17:06]blizzow/var/spool/asterisk/tmp/ exists and is owned by asterisk:asterisk. I even touched the file as the asterisk user and still get that error in the log. <- what file? |
| 21:19.08 | drmessano | Now it looks like theres a real scroll buffer being populated |
| 21:19.10 | [TK]D-Fender | blizzow: Next... what version of FreePBX is that from? |
| 21:20.41 | blizzow | Version - 2.8.1.4 The file mentioned in the log as not existing: /var/spool/asterisk/tmp/55005-ivrrecording |
| 21:21.44 | [TK]D-Fender | blizzow: First that is ancient unsupported junk at this point. |
| 21:21.52 | [TK]D-Fender | blizzow: second, that is NOT the file it is looking for |
| 21:22.14 | [TK]D-Fender | from the looks of it |
| 21:22.19 | [TK]D-Fender | blizzow: Dump the folder |
| 21:23.01 | blizzow | Which folder? |
| 21:24.07 | [TK]D-Fender | [2013-10-02 15:01:23] WARNING[16224] pbx.c: ast_streamfile failed on SIP/55005-0001d304 for /var/spool/asterisk/tmp/55005-ivrrecording,m,en,macro-systemrecording |
| 21:24.09 | [TK]D-Fender | see this? |
| 21:24.19 | blizzow | Yeah, I see that. |
| 21:24.30 | [TK]D-Fender | it is counting the COMMA and everything past as PART of the filename it's looking for |
| 21:25.34 | blizzow | The line just before that seems to indicate that it's only looking for the part prepended to the comma, no? |
| 21:26.05 | [TK]D-Fender | blizzow: DUMP THE FOLDER |
| 21:26.37 | blizzow | /var/spool/asterisk/tmp/ ?? |
| 21:26.50 | [TK]D-Fender | yes |
| 21:26.55 | anonymouz666 | nice... You do not appear to have the sources for the 2.6.32-358.el6.x86_64 kernel installed. |
| 21:27.02 | anonymouz666 | dahdi does not compile under KVM |
| 21:27.26 | drmessano | Without kernel sources it won't |
| 21:27.37 | anonymouz666 | it is there. |
| 21:28.36 | *** join/#asterisk deegen (~deegen@S01060023bee90320.gv.shawcable.net) |
| 21:28.46 | anonymouz666 | it doesn't find the kernel path |
| 21:28.58 | anonymouz666 | in a KVM machine |
| 21:29.33 | blizzow | [TK]D-Fender: then what, just re-create it as the asterisk user? |
| 21:30.17 | [TK]D-Fender | blizzow: "ls -la /var/spool/asterisk/tmp/" |
| 21:31.32 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:32.38 | blizzow | I moved the whole directory, so it's not there. (EG: mv /var/spool/asterisk/tmp /home/phoneuser/varspoolasterisk/tmp) |
| 21:33.16 | anonymouz666 | broken build link. that's it |
| 21:33.31 | [TK]D-Fender | blizzow: And you're wondering why it can't find things there? |
| 21:33.33 | [TK]D-Fender | BRB |
| 21:35.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:37.01 | blizzow | [TK]D-Fender: no, I'm not wondering why asterisk can't find it. It was there before when you told me to "dump the folder". I re-created the /var/spool/asterisk/tmp folder as the asterisk user and it still fails with the same messages. |
| 21:38.43 | blizzow | ls -la /var/spool/asterisk/tmp shows only . and .. with the following permissions: drwxrwxr-x |
| 21:41.35 | *** join/#asterisk navaismo (~navaismo@189.241.90.55) |
| 21:42.47 | [TK]D-Fender | blizzow: So * is looking for a file... that's not there. |
| 21:42.55 | [TK]D-Fender | bliAnd wondering why it is failing.... |
| 21:42.56 | file | is here |
| 21:47.15 | [TK]D-Fender | See, file is HERE, not THERE! |
| 21:50.28 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 21:52.26 | blizzow | Werewolf, no THEREwolf. I'm really confused as to what you're trying to tell me. I dumped the folder per your request. I recreated the folder so asterisk would have a place to create the sound file. Asterisk should take care of creating the temporary voicemail file on it's own. It does for other extensions. |
| 21:53.20 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:53.23 | [TK]D-Fender | blizzow: In your output all I saw is a BACKGROUND. NOt * creating a file there |
| 21:55.48 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-qwbjghrzehtopkpk) |
| 22:00.59 | blizzow | ok, here's logfile output from an extension that works. http://pastebin.com/FatpVgEJ |
| 22:00.59 | blizzow | and here's logfile from the extension that doesn't work: http://pastebin.com/aEaQS970 |
| 22:02.21 | blizzow | to me they look nearly the same, and even throw the same file not found error. |
| 22:04.25 | *** join/#asterisk xzarth_ (~krikkit@88.207.60.255) |
| 22:04.43 | [TK]D-Fender | blizzow: What difference am I supposed to be noting between those?> |
| 22:07.56 | *** join/#asterisk digitall (~digitall@cpc3-hitc7-2-0-cust590.9-2.cable.virginmedia.com) |
| 22:08.10 | blizzow | That's my problem. I can't figure out any difference between the two that would indicate why a new extension can't access voicemail or change their greeting. |
| 22:08.16 | digitall | Evening... |
| 22:08.47 | digitall | Quick headline question... Has anyone here had any experience setting up a DID with Draytel SIP? |
| 22:09.22 | [TK]D-Fender | blizzow: Nowhere is that call VOICEMAIL for either in any case. That is straight up dialplan apps doing OTHER random junk. |
| 22:09.37 | [TK]D-Fender | blizzow: That TMP folder has nothing to do with *'s voicemail that I can see |
| 22:10.04 | [TK]D-Fender | blizzow: And all I see is 1 user answering a prompt, and another hanging up |
| 22:10.17 | [TK]D-Fender | blizzow: And not getting anything concrete as to what constitutes a "failure" |
| 22:10.47 | [TK]D-Fender | blizzow: that is a FreePBX SYSTEM RECORDING. Not VOICEMAIL |
| 22:11.09 | [TK]D-Fender | blizzow: Who said that was the feature code for voicemail? |
| 22:11.52 | [TK]D-Fender | blizzow: The standards are *97 and *98, and those are just the DEFAULTS |
| 22:12.12 | blizzow | [TK]D-Fender: sorry, my misnomer. Yes, I'm trying to record a VM greeting. |
| 22:12.27 | [TK]D-Fender | blitz``: that is NOT how you do it |
| 22:12.47 | [TK]D-Fender | blizzow: You LOG IN .... and use the VM box options that the menu clearly reads to you |
| 22:13.25 | digitall | I have the DID PSTN number ringing my SIP trunk and then getting routed to ringing an extension or to the IVR menu.... but I am trying to work out how to do DDI for all the extensions... |
| 22:14.04 | digitall | ie. How to get the SIP DID to have a section which changes and can be mapped to the internal extension... i.e. 01234 567890 + ext no. |
| 22:14.19 | digitall | so 01234 567890100 for extension 100 etc. |
| 22:14.31 | [TK]D-Fender | digitall: Go re-read the chapter on DIALPLAN BASICS |
| 22:14.33 | [TK]D-Fender | ~book |
| 22:14.33 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:14.35 | [TK]D-Fender | ^^^ |
| 22:14.41 | [TK]D-Fender | digitall: this is basic pattern matching. |
| 22:14.53 | digitall | I KNOW... |
| 22:15.15 | blizzow | That's part of the problem I initially explained. The new extension cannot login with *97 using the VM password set for the extension. Nor am I able to record a greeting by using *99. |
| 22:15.48 | [TK]D-Fender | blizzow: Show us a failure using the proper way |
| 22:16.05 | [TK]D-Fender | blizzow: Oh, and... LEAVE A VM there first <- |
| 22:16.11 | [TK]D-Fender | to initialize the box |
| 22:16.11 | digitall | My problem is a more meta-question... My SIP provider is Draytel and they provide a single DID trunk... with a single PSTN number. I have tried adding digits to the end, but I get a "this number is not provisioned"... but I think that is from their system..... |
| 22:16.47 | digitall | as if I purposely break the linkage on the incoming part of the dialplan, I get a error on the console of unknown extension and hangup. |
| 22:16.55 | [TK]D-Fender | digitall: What do you mean "adding digitss"? They send you ONLY one number? |
| 22:17.08 | [TK]D-Fender | digitall: Show us the call. |
| 22:17.11 | [TK]D-Fender | ~pb |
| 22:17.11 | 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. |
| 22:17.12 | [TK]D-Fender | ^^^ |
| 22:18.19 | digitall | I can't ... that is the point. |
| 22:18.44 | digitall | i.e If I add any digits to the end of the PSTN number, they have provided... no incoming ringing on the trunk. |
| 22:18.48 | [TK]D-Fender | digitall: You say you are "adding digits". What is this supposed to mean? |
| 22:19.11 | [TK]D-Fender | digitall: Adding where? |
| 22:19.20 | WIMPy | digitall: Get a provider that allows you to do that. |
| 22:19.44 | [TK]D-Fender | WIMPy: Things are dangerously vague ... just wait for a little more rope... |
| 22:19.44 | digitall | As I said.... So if my assigned number is ie. 01234 567890, I am trying to get it so I can add an extension at the end... ie. 01234 567890 100 for extension 100 |
| 22:20.15 | [TK]D-Fender | digitall: They send you what the send you. |
| 22:20.15 | WIMPy | Nothing vague for me. |
| 22:20.31 | [TK]D-Fender | WIMPy: You've taken something as implicit here I can't account for yet... |
| 22:20.44 | WIMPy | What? |
| 22:21.01 | WIMPy | A SIP account is not a line with DDI. |
| 22:21.03 | blizzow | [TK]D-Fender: w00t, I think leaving a VM on the mailbox did it. so strange. |
| 22:21.05 | WIMPy | Usually. |
| 22:21.18 | blizzow | thanks for the hemp |
| 22:21.20 | blizzow | *help |
| 22:22.27 | digitall | http://www.draytel.org/plans ... |
| 22:23.02 | WIMPy | only sees number of numbers there, not DDI. |
| 22:24.10 | digitall | The query I have is trying to work out if it is normal for a provider to connect any range with that prefix e.g. 01234 567890 (*) to the trunk... or just a single number e.g. 01234 567890 to the trunk |
| 22:24.34 | WIMPy | Usually only a single number. |
| 22:24.37 | digitall | WIMPy: Yes... I did wonder about that... But all the providers seem to be very vague on their technical detail... It is painful. |
| 22:25.25 | digitall | WIMPy: Suspected as much... Damn. Any suggestions for providers who provide a range i.e. so I can sort out 01234 567 890 XXXX to map to internal extensions for DDI |
| 22:25.40 | digitall | I'm in the UK for reference. |
| 22:25.49 | WIMPy | Country? |
| 22:26.15 | WIMPy | I assume sipgate will have the team or trunk option in the UK as well. |
| 22:27.00 | file | sipgate is shutting down |
| 22:27.06 | WIMPy | hasn't looked too close at that, however. |
| 22:27.13 | WIMPy | Definitely not. |
| 22:27.39 | file | http://www.sipgate.com |
| 22:27.49 | file | oh, UK |
| 22:27.58 | WIMPy | Who is talking about the US? |
| 22:28.31 | file | not I! *cough* >_> |
| 22:28.32 | WIMPy | That have just come with new offerings here. |
| 22:29.15 | WIMPy | They... |
| 22:30.03 | digitall | Hmm... I know BT do SIP trunking services... Their tech support tends to be a PITA to deal with... a kafkaesque nightmare! :) But they are technically quite competent.. |
| 22:31.00 | digitall | IT manager chose this lot as being cheap and cheerful... |
| 22:33.52 | digitall | Has anyone done this kind of thing before and got it working? |
| 22:34.20 | WIMPy | Only with real lines. |
| 22:34.49 | WIMPy | (where things just work) |
| 22:35.40 | digitall | Hmm... FXS/FXO FTW! |
| 22:36.13 | WIMPy | What's that? |
| 22:37.17 | ChannelZ | for the win |
| 22:37.18 | digitall | WIMPy: Sorry... I _really_ have read the documentation... :) http://www.asterisk.org/get-started/glossary |
| 22:37.46 | WIMPy | ChannelZ: I know. I meant the FXS/FXO part. |
| 22:39.06 | ChannelZ | Fucking Xylophone Sound / Farting Xenophobe Opera |
| 22:43.39 | digitall | Foreign Exchange Station / Foreign Exchange Office according to the glossary... as far as I can see, the male/female, master/slave of the PSTN line .... ie. whichever port card you have... Murphy's law says it's the wrong one for the equipment. :) |
| 22:44.08 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 22:44.50 | ChannelZ | I'd have figured you knew that too WIMPy |
| 22:45.44 | WIMPy | Yes, I heard about it in here. But I'm only 43 years old. That was a little before my time. |
| 22:50.20 | WIMPy | But thanks to SIP the analogoue stuff has a bug revival, it seems. |
| 22:56.23 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 22:59.51 | ChannelZ | Like vinyl |
| 23:00.49 | WIMPy | Just that vinyl has something on the positive side. |
| 23:02.58 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 23:04.06 | digitall | Yes... The flexidiscs occasionally selfcombust taking out a hipster! ;-) |
| 23:31.34 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:34.34 | digitall | Hmm... Anyway, thank you for the answers... Time for sleep in my TZ. ttfn. |
| 23:34.47 | *** part/#asterisk digitall (~digitall@cpc3-hitc7-2-0-cust590.9-2.cable.virginmedia.com) |
