| 00:27.21 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 01:41.58 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 05:23.06 | yottanami | I want show my dial plan I run console by asterisk -vvvvvr and I have this error : No such command 'show dialplan' (type 'core show help show dialplan' for other possible commands) |
| 05:27.04 | *** join/#asterisk r00f (~r00f@bba163826.alshamil.net.ae) |
| 05:28.36 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 05:31.40 | TazzNZ | yottanami: dialplan show |
| 05:38.15 | yottanami | TazzNZ: What about "file formats show" ? |
| 05:39.39 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 05:49.14 | yottanami | I copied my welcome message in .wav and .gsm format but I have this error yet : Sep 22 09:23:59 WARNING[25263] file.c: File agent-pass does not exist in any format |
| 05:49.15 | yottanami | Sep 22 09:23:59 WARNING[25263] file.c: Unable to open agent-pass (format ulaw): No such file or directory |
| 05:51.51 | yottanami | here is my full error that said file does no exist http://dpaste.com/1Y2HBNT/ |
| 06:08.49 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 06:24.23 | ChannelZ | Make sure it has read permission. |
| 06:24.36 | ChannelZ | And that it's in the right place. |
| 06:25.39 | yottanami | ChannelZ: It is 777, I also changed my dial plan to ael-default and I got error " Unable to open demo-instruct (format 0x4 (ulaw)): No such file or directory" |
| 06:26.42 | *** join/#asterisk Draecos (~Draecos@106-69-31-78.dyn.iinet.net.au) |
| 06:27.18 | yottanami | ChannelZ: That there is "demo-instruct.gsm" "/usr/share/asterisk/sounds/en_US_f_Allison" |
| 06:28.28 | ChannelZ | well unless your language is set to "en_US_f_Alison" it's not going to play anything out of there |
| 06:28.50 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:9d3f:6c64:4e5a:ab34) |
| 06:32.06 | yottanami | ChannelZ: Where should I set it ? |
| 06:32.20 | yottanami | but my recorded files can not play too |
| 06:32.42 | yottanami | can it be because of func_ and format_ modules ? |
| 06:35.07 | *** join/#asterisk r00f (~r00f@bba134458.alshamil.net.ae) |
| 06:38.01 | ChannelZ | Well you can have bunk files yes. The wavs need to be 16-bit 8khz mono. How did you make the gsm? |
| 06:38.43 | ChannelZ | And generally they are in wherever your spool dir is in sounds/en |
| 06:39.55 | yottanami | ChannelZ: I used full address and it fixed, tnx |
| 06:39.58 | ChannelZ | or sorry the data directory not spool. |
| 06:40.11 | yottanami | I create gsm with sox |
| 06:41.43 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 06:54.09 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 06:54.19 | snadge | why would i have to restart asterisk for it to find an audio file ? |
| 06:54.39 | snadge | [2014-06-02 16:53:39] WARNING[32184]: file.c:953 ast_streamfile: Unable to open ogm/09536044/livinghealth (format 0x8 (alaw)): No such file or directory |
| 06:54.46 | snadge | that's the error im getting.. but the file exists |
| 06:55.27 | snadge | one of the guys just told me their usual fix is to restart the server.. and im thinking.. that doesn't sound right to me |
| 07:01.21 | *** join/#asterisk CeBe (~CeBe@brsg-4dbbee78.pool.mediaWays.net) |
| 07:01.28 | ChannelZ | because it's not |
| 07:01.34 | *** join/#asterisk jmls (~jmls@77.107.171.82) |
| 07:02.03 | ChannelZ | and does that whole path exist inside the sounds/en dir, and asterisk have read permission to the entire thing? |
| 07:03.25 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:05.08 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 07:13.44 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 07:13.59 | yottanami | I have a simple dialplan (just exten => s,1,Background(welcome) ) |
| 07:13.59 | yottanami | in some incomming cals it play background but in some other I got this error http://dpaste.com/0ZC7GA5/ |
| 07:14.50 | yottanami | should I change 8000 to newrock or newrock to 8000 ? |
| 07:18.22 | *** join/#asterisk iulhk (iulhk@182.189.119.229) |
| 07:21.17 | ChannelZ | I dunno, what is the peer's actual name? |
| 07:21.25 | ChannelZ | (8000 probably) |
| 07:21.59 | ChannelZ | Whatever is trying to make the call is using a separate Username in its authentication string. You really should fix the peer. |
| 07:26.35 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
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| 09:07.32 | *** join/#asterisk jaflong (5bec7504@gateway/web/freenode/ip.91.236.117.4) |
| 09:11.49 | jaflong | hi, when I do a Dial() asterisk sends a re-invite. How can re-invites be stopped. |
| 09:12.19 | jaflong | I tried directmedia=yes but it still sends re-invites |
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| 09:20.55 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:31.02 | Zogot | jaflong: don't you want reinvites? otherwise the phone will hangup after the session timeout |
| 09:35.38 | jaflong | Zogot: I am not sure . Let me explain what I want to do. |
| 09:40.09 | jaflong | I have a video device calling to asterisk. This includes the video attribues in the sdp. When the call hit asterisk, I use the Dial() in the dialplan for this call. However asterisk sends a re-invite and in the sdp of the reinvite is only audio attributes and the video part is left out |
| 09:41.33 | jaflong | I think a solution will also be if the reinvite sdp support video |
| 09:43.06 | *** join/#asterisk workingcats (~workingca@212.122.48.77) |
| 09:43.29 | jaflong | so I am lokking for direct rtp seesion between the the devices that calls asterisk and the device asterisk calls at the end of the Dial(), or if the reinvites supports video |
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| 10:10.57 | dan_j | Hi. I'm just testing out asterisk 12. Where does libjansson come from? I'm looking for a yum repo with it |
| 10:13.41 | wdoekes | dan_j: if I google libjansson, the main site turns up #1 |
| 10:13.50 | wdoekes | no yum repo's though |
| 10:14.17 | eirirs | yumyum |
| 10:14.54 | wdoekes | although rpmfind.net shows a few |
| 10:19.49 | jaflong | "The current versions of ASterisk cannot reinvite the video stream" This is noted in the asterisk for dummies book. Is this still the case at present |
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| 11:19.00 | jaflong | is there a way I can execute another app after executing confBridge |
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| 12:01.10 | *** join/#asterisk As001 (~uros@82.117.198.142) |
| 12:01.34 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:02.19 | As001 | Hello my Asterisk see one agent as always busy in all queues. How can I change his statate in queue show output to not in use as he is not busy. I would like to avoid restarting entire asterisk. |
| 12:03.09 | As001 | logout agent did not help, removing from queues and adding again did not help. |
| 12:03.28 | [TK]D-Fender | Show us, and give us proper details on what you're running |
| 12:04.54 | As001 | Asterisk version 1.8.27, I have a lot of queues and agents in them, and one agent is Busy in queue show command output. |
| 12:05.07 | As001 | and he is not even logged in. |
| 12:05.30 | yottanami | I installed Asterisk & dahdi from debian package manager but there is no zapata.conf, should I install zaptel from source? |
| 12:06.04 | Chainsaw | yottanami: Why would there be a zapata.conf on dahdi? It's no longer called zaptel. |
| 12:06.08 | [TK]D-Fender | As001: SHOW US |
| 12:06.54 | As001 | what to show you queue show line ? Agent/501 (realtime) (Busy) has taken no calls yet |
| 12:07.17 | As001 | he somehow stayed in Busy state I don't know how. |
| 12:08.25 | [TK]D-Fender | Shows us the actual queue dump, logout attempts, etc..... |
| 12:08.25 | yottanami | Chainsaw: I just want to add outbuilding call,So I think my book is old! should I config dahdi.conf ? and how ? |
| 12:09.17 | [TK]D-Fender | DAHDI replaced Zaptel. |
| 12:09.34 | [TK]D-Fender | You shouldn't have a zaptel.conf / zapata.conf any more |
| 12:11.11 | yottanami | [TK]D-Fender: should I change /etc/dahdi/xpp.conf file for creating out bounding calls ? |
| 12:11.30 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 12:11.56 | [TK]D-Fender | What is this term you're using? "out bounding calls " |
| 12:12.04 | [TK]D-Fender | And what are you calling out using? |
| 12:25.56 | [TK]D-Fender | Yup, it's a "no-show" Monday... |
| 12:28.33 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-59-195.user.veloxzone.com.br) |
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| 13:05.03 | *** join/#asterisk `mc (~mc@c-71-235-50-162.hsd1.ct.comcast.net) |
| 13:05.07 | `mc | mornin |
| 13:06.36 | `mc | I am trying to investigate why a user's call wasn't transferred successfully. After pressing "1" the person was supposed to be transfer to a conference with another person but it just was dead silent. The other person got the call notification to connect to the conference but there was no one there..I want to understand how I can troubleshoot the cause |
| 13:08.39 | [TK]D-Fender | What is this transfer you're referring to? How does this relate to some sort of conference? |
| 13:11.30 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 13:11.41 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:11.41 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:12.18 | Katty | morning |
| 13:12.25 | Chainsaw | Good morning Katty. |
| 13:12.28 | [TK]D-Fender | Katty: Mew. |
| 13:12.35 | Katty | hugs Chainsaw and [TK]D-Fender |
| 13:12.40 | Chainsaw | :) |
| 13:19.09 | `mc | [TK]D-Fender: Forgive me but I didn't developer this app. I know how its supposed to work and I know it uses meetme() and its Asterisk 11 LTS Certified ;) |
| 13:19.30 | `mc | [TK]D-Fender: I'm trying to isolate what the problem is first, not so much in code but in the logs |
| 13:20.11 | file | raises eyebrow |
| 13:20.21 | [TK]D-Fender | `mc: well your "app" sounds to be doing something specific and unless you can drill the the code, we have no idea how it's supposed to work, what it's doing or how. |
| 13:20.57 | [TK]D-Fender | `mc: We need more to go on |
| 13:22.47 | *** join/#asterisk yottanami (~yottanami@5.52.79.23) |
| 13:24.48 | *** join/#asterisk chezgi (~abbas@2.180.59.97) |
| 13:24.57 | yottanami | [TK]D-Fender: Sorry I disconnected and cound not find out the answer about dahdi |
| 13:24.58 | yottanami | for out bounding should I change /etc/dahdi/xpp.conf ? What others should I change for a ZAP ? |
| 13:25.05 | `mc | [TK]D-Fender: what can I provide you? |
| 13:26.01 | chezgi | hello |
| 13:26.59 | chezgi | "manager show events" in console only prints this events: OriginateResponse ParkedCallGiveUp ParkedCallTimeOut |
| 13:27.09 | chezgi | how can i enable other events? |
| 13:27.27 | [TK]D-Fender | yottanami: You haven't told us what you're using.... |
| 13:27.58 | yottanami | [TK]D-Fender: I use Astersik on Debian I installed Asterisk from Debian repository |
| 13:28.31 | chezgi | can anybody help me? |
| 13:28.39 | [TK]D-Fender | `mc: You need real details about what this app of yours is calling and how. |
| 13:29.06 | `mc | [TK]D-Fender: i know the flow and what should happen |
| 13:29.22 | [TK]D-Fender | `mc: We need specifics. |
| 13:29.39 | `mc | ok, which specifics do you need? |
| 13:29.53 | [TK]D-Fender | `mc: Exactly how it is interacting with * to do what it has to do. |
| 13:30.07 | [TK]D-Fender | `mc: And provide debug and code to match. |
| 13:30.16 | *** join/#asterisk SuperNull (~YoMommaEa@24-148-101-238.ip.mhcable.com) |
| 13:30.29 | yottanami | [TK]D-Fender: I use a NewRock USB GateWay |
| 13:30.29 | chezgi | i am using asterisk 11.9.0 from fedora 20 |
| 13:30.43 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:31.11 | [TK]D-Fender | yottanami: What exact model? |
| 13:31.21 | SuperNull | am i correct to say if an ata has a regtime of one hour and i set maxexpiry=60 it will force it to register every minute ? |
| 13:32.04 | yottanami | [TK]D-Fender: NewRock HX411 |
| 13:34.24 | [TK]D-Fender | yottanami: That looks like a SIP GATEWAY, not a USB gateway. |
| 13:37.41 | chezgi | [TK]D-Fender: can you help me? |
| 13:39.48 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:39.49 | yottanami | [TK]D-Fender: I am sorry It is not USB |
| 13:40.23 | [TK]D-Fender | yottanami: If it's a SIP gateway as it appears then you have no need for DAHDI to use it at all. |
| 13:40.56 | BeachBall | looks at [TK]D-Fender ... Hug time |
| 13:41.03 | BeachBall | ;D |
| 13:41.44 | yottanami | [TK]D-Fender: YEs it is SIP, I wrote http://dpaste.com/1SW6DMX/ in extensions.conf what other should I do for outgoing ? |
| 13:42.09 | yottanami | [TK]D-Fender: OUTBOUNDTRUNK=Zap/1 |
| 13:42.58 | [TK]D-Fender | yottanami: that is not a DAHDI device. |
| 13:43.08 | [TK]D-Fender | yottanami: stop trying to use it as though it were |
| 13:43.15 | [TK]D-Fender | yottanami: this is a SIP DEVICE |
| 13:46.07 | yottanami | [TK]D-Fender: Can you help me more about out bounding on SIP devices ( or any document ) |
| 13:46.14 | [TK]D-Fender | ~book |
| 13:46.14 | 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 |
| 13:46.18 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
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| 13:48.20 | *** join/#asterisk Johnny5 (~Johnny5@nat/digium/x-gahjhwjiiebszscd) |
| 13:49.44 | *** part/#asterisk chezgi (~abbas@2.180.59.97) |
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| 13:57.04 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:57.10 | yottanami | [TK]D-Fender: Can You help me which part of Astersik books talks about this ? because the out bounding section does not use SIP device, I just want to create a route like this http://tarvandco.com/blog/wp-content/uploads/2013/12/12-10-2013-10-14-45-AM.png ( in Elastix ) |
| 13:58.28 | [TK]D-Fender | yottanami: It works like just about any other SIP device |
| 13:59.05 | [TK]D-Fender | SIP is SIP. |
| 13:59.55 | [TK]D-Fender | yottanami: You showed raw dialplan before, and this iws a GUI config page for Elastix's chopped up implementation of FreePBX |
| 14:00.08 | [TK]D-Fender | yottanami: SO you shouldn't be comparing the process for hand-coded with thqat of a GUI |
| 14:00.44 | yottanami | [TK]D-Fender: I did not use GUI, It was a sample for outbounding that I found it, but I dont want to use Elastix |
| 14:01.46 | [TK]D-Fender | yottanami: You swhouldn't be looking at that at all then |
| 14:01.53 | yottanami | [TK]D-Fender: I want to know What should I set instead of "exten => _9N.,1,Dial(Zap/${EXTEN:1})" for outgoing calls |
| 14:02.00 | [TK]D-Fender | yottanami: Go read the book. You don't seem to understand any of the basics yet |
| 14:02.41 | [TK]D-Fender | Dial(Zap/${EXTEN:1}) <- and this isn't right on ANY system. |
| 14:03.43 | yottanami | [TK]D-Fender: you mean Dial(SIP/MY_SIP}) can helps ? |
| 14:04.12 | [TK]D-Fender | yottanami: Go read the book. there is no shortcut to learning the basics of setting up your devices & dialplan |
| 14:04.19 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:04.19 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:05.20 | yottanami | [TK]D-Fender: I read Dialplan parts of Jim Van Meggelen, Jared Smith, Leif Madsen-Asterisk_ The Future of Telephony-O'Reilly Media (2007) but all the samples was about Zap for outgoing, can you introduce a book or a section of this book that is more important ? |
| 14:05.38 | [TK]D-Fender | Look at any SIP provider sample |
| 14:05.43 | [TK]D-Fender | They are mostly the same |
| 14:06.05 | [TK]D-Fender | Dial(SIP/peername/numbertodial....) |
| 14:06.58 | [TK]D-Fender | And you shouldn't be using the 2007 book. |
| 14:07.38 | yottanami | [TK]D-Fender: Is any online free book about that ? |
| 14:07.59 | [TK]D-Fender | [09:46][TK]D-Fender~book |
| 14:08.00 | [TK]D-Fender | [09:46]infobotAsterisk: 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 |
| 14:08.02 | [TK]D-Fender | [09:46][TK]D-Fender^^^^^^^^^^^^^^ |
| 14:10.46 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-ymgfbhpqciqzizow) |
| 14:11.48 | yottanami | [TK]D-Fender: Thanks allot |
| 14:15.06 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 14:27.37 | *** join/#asterisk solmsted (~solmsted@pool-71-176-247-26.rcmdva.fios.verizon.net) |
| 14:33.34 | *** join/#asterisk phpboy (~shane@196.46.187.254) |
| 14:34.24 | phpboy | Hi, I'm trying to use WebRTC... SIPML5 client... I'm able to register with my asterisk server but when I place a call I get this error on the asterisk console... [Jun 2 17:26:58] WARNING[8351][C-0000000a]: chan_sip.c:10512 process_sdp: Rejecting secure audio stream without encryption details: audio 4060 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 |
| 14:35.04 | *** part/#asterisk yottanami (~yottanami@5.52.79.23) |
| 14:35.59 | phpboy | I think this is a tls / srtp issue but I cannot be sure. Has anybody ever done this before and experienced similar issues? |
| 14:39.47 | [TK]D-Fender | It's clearly complaining about a mismatch between expect4ed vs received encryption.... |
| 14:39.59 | *** join/#asterisk iulhk (iulhk@182.189.29.167) |
| 14:40.03 | [TK]D-Fender | One side expects and the other doesn't. result = not happy |
| 14:41.22 | *** join/#asterisk igcewieling (~ewieling@ip98-183-26-100.pn.at.cox.net) |
| 14:47.05 | *** part/#asterisk igcewieling (~ewieling@ip98-183-26-100.pn.at.cox.net) |
| 14:57.49 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 14:59.00 | phpboy | [TK]D-Fender: but the question is where/how do I fix this |
| 14:59.11 | phpboy | not even sure what I am or am not looking for |
| 14:59.36 | [TK]D-Fender | phpboy: The client is saying "no encrption", your side is expecting it... change one of them so they match |
| 15:02.42 | phpboy | that's the thing, I've not completely disabled encryption on the asterisk server and still I get this error |
| 15:03.21 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:f74:21b:63ff:fe31:8426) |
| 15:14.27 | phpboy | :| |
| 15:14.43 | phpboy | I think I need to go back to the drawing board with this WebRTC stuff |
| 15:15.16 | [TK]D-Fender | Sorry, looks like I got the sides reversed |
| 15:16.49 | anonymouz666 | phpboy: lo, I've got the same warning here. |
| 15:16.58 | anonymouz666 | I just started to play with webrtc stuff |
| 15:17.38 | file | http://www.joshua-colp.com/webrtc-let-me-get-right-on-that/ |
| 15:20.47 | phpboy | [TK]D-Fender: I believe I have everything for secure connections set up and complied but my asterisk doesn't seem to want to play nice |
| 15:21.26 | phpboy | anonymouz666: it's very painful :( |
| 15:21.35 | [TK]D-Fender | you just said you set * to NOT be secure. |
| 15:21.50 | file | SRTP is required for WebRTC. |
| 15:21.55 | [TK]D-Fender | "I have everything for secure connections set up" |
| 15:22.22 | [TK]D-Fender | "I've not completely disabled encryption on the asterisk server" <- loks like a double-negative... |
| 15:22.54 | phpboy | [TK]D-Fender: It was and then I changed it to NOT secure and not it's back to secure |
| 15:22.55 | [TK]D-Fender | and an ambiguous "not completely" |
| 15:23.20 | [TK]D-Fender | "and not it's back to secure" <- still hard to follow wording... |
| 15:23.25 | phpboy | Apparently Webrtc will only work with secure (TLS and SRTP) * |
| 15:23.29 | [TK]D-Fender | fix your description and provide backup |
| 15:23.52 | anonymouz666 | phpboy: yeah, file just said that ! |
| 15:23.54 | anonymouz666 | heh. |
| 15:24.03 | anonymouz666 | let's config then! |
| 15:24.07 | file | WebRTC does not mandate any signaling, so the statement about it needing TLS is incorrect |
| 15:24.10 | anonymouz666 | and then make the first webrtc call |
| 15:24.49 | anonymouz666 | yeah we could use jingle xmpp if we want but we want trouble, let's go with SIP! |
| 15:25.16 | phpboy | file: if that's true I have definitely tried it with secure config options and standard (non secure) config options and still I get this same error |
| 15:25.24 | phpboy | WARNING[8351][C-0000000a]: chan_sip.c:10512 process_sdp: Rejecting secure audio stream without encryption details: audio 4060 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 |
| 15:25.48 | phpboy | anonymouz666: what all have you done and what have you tried to resolve this error? |
| 15:26.16 | anonymouz666 | phpboy: I need to setup SRTP before starting to cry here again |
| 15:26.39 | phpboy | anonymouz666: srtp install is very simple... let me know if you want a quick guide |
| 15:30.39 | anonymouz666 | need to install the libs |
| 15:31.26 | phpboy | ok |
| 15:31.46 | anonymouz666 | 1.4.4 from yum |
| 15:31.51 | anonymouz666 | I am afraid if this is old |
| 15:35.31 | phpboy | bad idea |
| 15:35.38 | phpboy | 11.*+ is what you want |
| 15:35.43 | phpboy | compile it manually |
| 15:35.58 | anonymouz666 | 1.4.4 is the libsrtp |
| 15:36.02 | anonymouz666 | not ast version |
| 15:36.36 | phpboy | ah ok |
| 15:37.02 | phpboy | I've got 1.4.5 |
| 15:37.15 | phpboy | probably won't be much difference between the two |
| 15:41.21 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 15:57.41 | anonymouz666 | phpboy: same error. |
| 15:58.28 | anonymouz666 | libsrtp loaded, tried the rtcweb breaker, avpf, encryption... |
| 16:01.37 | phpboy | me too |
| 16:01.55 | phpboy | what seems weird though is how long it takes to setup the call |
| 16:03.38 | phpboy | SIP debug not telling me much more than I already know |
| 16:03.40 | phpboy | any ideas? |
| 16:04.37 | anonymouz666 | I don't have ANY ideas. |
| 16:07.23 | phpboy | this sucks |
| 16:07.41 | anonymouz666 | phpboy: need to google and see if other people already saw this |
| 16:08.24 | phpboy | I've found tons of people that have had this issue but no solutions |
| 16:09.04 | anonymouz666 | so we are lucky |
| 16:09.56 | anonymouz666 | are you using SIPML5? |
| 16:13.00 | phpboy | yep |
| 16:13.57 | Katty | nope. |
| 16:14.11 | Katty | tinkers with phpboy's source. |
| 16:14.15 | *** join/#asterisk navaismo (~navaismo@187-178-254-98.dynamic.axtel.net) |
| 16:15.24 | anonymouz666 | phpboy: what asterisk 11 version? |
| 16:15.34 | anonymouz666 | here it is the 11.6.0 |
| 16:16.18 | anonymouz666 | https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 16:16.31 | anonymouz666 | tried that, still no lucky. |
| 16:16.33 | anonymouz666 | same error. |
| 16:17.57 | anonymouz666 | phpboy: If i tried from a normal phone to chrome, works. |
| 16:18.01 | phpboy | What I don't understand is in sipml5 nothing 'security' related has been set |
| 16:18.31 | phpboy | but when I try make a call it tries to use security |
| 16:18.33 | phpboy | :\ |
| 16:19.22 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:19.49 | Katty | hi tony |
| 16:19.54 | navaismo | plain webrtc need encryption=yes and avpf |
| 16:23.09 | phpboy | I have both enabled |
| 16:23.34 | navaismo | are you using chrome 35? |
| 16:23.36 | phpboy | and still it doesn't work |
| 16:24.21 | navaismo | becasue chroe 35 use dtls-srtp |
| 16:24.34 | navaismo | and you need to patch-configure asterisk for that |
| 16:24.37 | phpboy | chrome: 35.0.1916.114 m .... firefox: 29.0.1 |
| 16:24.55 | phpboy | oh lovely, another patch |
| 16:24.58 | phpboy | lol :P |
| 16:25.29 | mjordan | if you can't handle patching things, dealing with source, and generally working with a moving target, you should not be using WebRTC. |
| 16:25.31 | phpboy | I think it may be easier to proxy this using freeswitch |
| 16:25.35 | mjordan | go for it. |
| 16:26.08 | anonymouz666 | now I understand the mjordan's quote |
| 16:26.16 | phpboy | Me too |
| 16:26.21 | mjordan | could not care less. |
| 16:26.59 | phpboy | I'm looking to WebRTC as a solution to a problem I'm experiencing at a client |
| 16:27.13 | phpboy | but a moving target solution is more of a hack than a solution |
| 16:27.14 | anonymouz666 | phpboy: I think you are going to have 2 problems |
| 16:27.19 | navaismo | yeah why peoplo blame asterisk since webrtc is very inmature |
| 16:27.20 | phpboy | ye? |
| 16:27.53 | phpboy | I'm blaming the client in this case (WebRTC) not * |
| 16:28.04 | phpboy | just hoping * has some sort of answer for me |
| 16:28.16 | anonymouz666 | I am blaming nobody, just trying to understand how stuff works |
| 16:28.53 | phpboy | anonymouz666: it looks like the client is trying to setup a secure connection of some sort and * hasn't been set up for said secure connection. |
| 16:29.12 | anonymouz666 | phpboy: navaismo just explained what happens with chrome 35 |
| 16:29.12 | phpboy | this DTLS patch may be the answer to that question |
| 16:29.38 | phpboy | it seems firefox may be experiencing the same problem |
| 16:29.55 | phpboy | I'll play with it some more tomorrow (it's 18h30) here |
| 16:30.02 | phpboy | brain is tired now |
| 16:30.32 | anonymouz666 | but... what do you think guys about webrtc? it will in near future with a default behaviour? |
| 16:30.43 | anonymouz666 | *usable |
| 16:31.18 | anonymouz666 | if google keeps changing stuff, it will not be a good idea try to follow every path |
| 16:32.21 | phpboy | :| |
| 16:32.29 | phpboy | I don't think it's google specifically |
| 16:32.47 | phpboy | I think it's a RFC thing for webrtc |
| 16:33.51 | jameswf | outside of chan_dahdi*conf and realtime where would chan_dahdi pull configs from |
| 16:34.03 | Qwell | users.conf? |
| 16:35.01 | anonymouz666 | users.conf is a mistery to me. never used. don't even know what is for |
| 16:38.58 | mjordan | Qwell: -1 for mentioning the abomination |
| 16:39.11 | Qwell | mjordan: I will make it up in tacos. |
| 16:39.19 | mjordan | Qwell: gracias. |
| 16:40.25 | jameswf | yeah it was.... who uses that |
| 16:40.43 | jameswf | glares at tzafrir |
| 16:43.26 | Qwell | jameswf: I was right? Huh. First time for everything. |
| 16:44.06 | jameswf | Qwell: in my [8?9?+] years I have never seen/used that file |
| 16:45.20 | [TK]D-Fender | ~users.conf |
| 16:45.20 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 16:45.22 | navaismo | anonymouz666, phpboy there is battle about standards and codec, now a new branch emerged called ORTC so i guess there is no rfc sooner. Your ptions so far are to use a media gateway or pacth-configure asterisk |
| 16:45.27 | [TK]D-Fender | Ah... still there... |
| 16:46.06 | pabelanger | users.conf is a flaming pile of sh1t |
| 16:46.07 | pabelanger | classy |
| 16:47.09 | mjordan | [TK]D-Fender: nice |
| 16:47.10 | [TK]D-Fender | pabelanger: users.conf was one of the iron-pyrite bricks used to pave the road to hell.... |
| 16:47.31 | mjordan | that sounds like something Qwell would write |
| 16:47.46 | Qwell | takes a bow |
| 16:48.46 | Qwell | PS, my scapegoat timer has expired. :p |
| 16:48.58 | mjordan | ha |
| 16:49.05 | mjordan | by quite a long measure, too |
| 16:49.08 | Qwell | YOU HAD YOUR CHANCE. :D |
| 16:49.35 | mjordan | Hey, I'm proud of that one. A wart is a wart by any measure |
| 16:49.37 | file | gives Qwell a burrito |
| 16:49.43 | file | of BLAME |
| 16:49.53 | mjordan | feels a blamestorm coming on |
| 16:50.06 | Kobaz | i dont understand why after all this time working with sip, i still run into problems like this |
| 16:50.17 | Kobaz | okay, so i have an audiocodes trying to register |
| 16:50.23 | Kobaz | and i got my happy little sip debufs |
| 16:50.30 | Kobaz | debugs... |
| 16:50.52 | Kobaz | so I get a register: Looking for c30010-151 in default (domain 192.168.100.175) |
| 16:50.53 | Qwell | MMO debuff: SIP, causes -10 intellect. |
| 16:51.10 | Kobaz | and then: <--- Transmitting (NAT) to xxx -> SIP/2.0 404 Not Found |
| 16:51.37 | Kobaz | > sip show peers like c30010-151 c30010-151/c30010-151 |
| 16:51.40 | Kobaz | it's definitely found |
| 16:51.45 | Kobaz | why the hell is it sending back a 404 |
| 16:51.55 | mjordan | do you have an extension c30010-151@default? |
| 16:52.19 | Kobaz | no. i'm not using context default |
| 16:52.38 | [TK]D-Fender | that call is |
| 16:52.43 | Kobaz | ooooh |
| 16:52.49 | mjordan | and that's not typically logged for a REGISTER request... |
| 16:53.06 | mjordan | that looks like what I would expect to see for an inbound INVITE request |
| 16:53.26 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 16:53.35 | Kobaz | that's responding to OPTIONS |
| 16:53.46 | Kobaz | the register actually hmm |
| 16:53.48 | Kobaz | is not going through |
| 16:54.05 | Kobaz | i think the audiocodes is bailing on registering because of the 404 on options |
| 16:55.00 | Kobaz | so what i should have said... clicking on register and then get.... |
| 16:55.06 | Kobaz | not actually receiving a reg just yet... ugh |
| 16:56.25 | anonymouz666 | navaismo: did you try the webrtc2sip? |
| 16:56.56 | navaismo | i use that without dtls sonner i will start to configure it |
| 17:00.21 | Kobaz | okay |
| 17:00.22 | Kobaz | yay fixed it |
| 17:00.31 | Kobaz | it was on the audiocodes side |
| 17:00.41 | Kobaz | had the username associated to the wrong port |
| 17:01.40 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:01.45 | *** join/#asterisk mirela666 (~mirko.bra@82.117.215.158) |
| 17:03.16 | navaismo | thanks file for immortalized the webrtc words that why im one of your (hidden) groupies |
| 17:03.36 | file | mjordan said it, I just put it in an unofficial place |
| 17:03.44 | *** join/#asterisk elbriga (~elbriga@177.220.177.165) |
| 17:03.51 | navaismo | yep i was here when he wrote it |
| 17:05.28 | navaismo | now i can share that tweet together the tweet of Iñaki Baz |
| 17:11.37 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 17:15.29 | *** join/#asterisk litn (~blice@alrig.ht) |
| 17:15.50 | litn | hey guys, we have some lines that are less than ideal, they drop calls here or there. I was wondering if there is some setting I can do so that it doesn't drop the call if the connection gets choppy? |
| 17:16.00 | litn | like a timeout setting or number of dropped packets, I don't know |
| 17:17.32 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
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| 17:33.17 | dan_j | Has anyone got Panasonic SIP Phones connected to their asterisk box and noticed that the quality figures are higher than other make/model phones on the same network? |
| 17:35.13 | navaismo | no |
| 17:37.13 | dan_j | weird. ive got panasonic phones connected to my asterisk box, and consistantly, the response time is 100ms higher than other phones behind the same router. |
| 17:37.37 | *** join/#asterisk wsnw (warrenw@barfly.xmission.com) |
| 17:38.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
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| 17:45.45 | dan_j | litn: usually calls are dropped by the sip trunk provider and you can't control that. most often, calls are dropped when either server doesnt receive a SIP dialog packet at the start of the call. Asterisk will retry 5 times to get the packet through, and then give up and drop the call. |
| 17:46.26 | dan_j | litn: From a call quality point of view, have you tried g729? You may need to buy a license for that, but i've found it helps with poor connections. |
| 17:46.42 | litn | dan_j: I reached out to the provider who said they didn't see anything weird on their end and requested a tcpdump. But it seems like the dropped calls are occuring more to some users than others (on the same network) |
| 17:46.42 | *** part/#asterisk LiuYan (~stephen@unaffiliated/asenr) |
| 17:46.50 | file | dan_j, the quality time also includes processing time by the device in question - some devices will place the priority of certain messages lower |
| 17:48.05 | dan_j | file: yes, i realised that. i just found it interesting. makes it a little hard to compare network connections for quality if the phones are responding slower. |
| 17:48.25 | dan_j | file: btw, is it possible to increase the quality required to be classed as OK? |
| 17:48.39 | dan_j | can i simply do quality=5000 ? |
| 17:48.49 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:50.55 | [TK]D-Fender | first, that's "qualify", second, that's the amount of time for * to get a response", not a frequency. |
| 17:52.00 | dan_j | yes, i know that its the amount of time to get a response. |
| 17:52.12 | dan_j | when i said increase, i meant increase the amount of time |
| 17:52.18 | dan_j | to get a response. |
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| 18:59.16 | *** join/#asterisk warc (~warc@d51A490D7.access.telenet.be) |
| 19:02.15 | warc | Hi everyone, I am designing a project and was wondering if asterisk is the right tool for it. In particular, I was wondering if asterisk can be used to develop an application that can send SMS's to users and can interact with users over USSD. Does anyone have an idea if this is possible and where I can find more information on the subject? |
| 19:03.18 | *** join/#asterisk fireman_biff (~biff@208.0.98.13) |
| 19:04.07 | fireman_biff | Hi all, Am I correct in believing that when my PRI detects a yellow alarm that the fault is definitely with the provider? |
| 19:06.04 | [TK]D-Fender | warc: I've seen something about chan_dongle support for USSSD, but that's about it. And since * isn't natively an SMS platform it probably isn' the best starting point for you |
| 19:06.08 | navaismo | not always |
| 19:06.44 | warc | Fender, thank you for the information. I'll check out some other platforms |
| 19:07.06 | *** join/#asterisk yottanami (~yottanami@5.52.12.12) |
| 19:07.52 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 19:08.47 | yottanami | I want create outbound, here is my trunk http://www.dpaste.com/1JJC8M5/ should I change context ? or any item? |
| 19:13.20 | [TK]D-Fender | yottanami: First that implies that your device registers to *, second you didn' |
| 19:13.43 | [TK]D-Fender | 't specifiy peer-leve codecs which you really should do. |
| 19:13.53 | [TK]D-Fender | And that it just the peer... not the dialplan that uses it |
| 19:13.59 | [TK]D-Fender | so go prove that your peer has registered |
| 19:14.09 | [TK]D-Fender | If that hasn't happened, forget about dialping out. |
| 19:16.43 | [TK]D-Fender | You should also never use the context "default" ever |
| 19:16.56 | *** join/#asterisk jwr_ (~quassel@205.196.167.54) |
| 19:17.13 | [TK]D-Fender | your [general] should point to some dead-end unless you have need of accepting un-authed calls, and your peers hsould certainly never point to that place. |
| 19:31.09 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 19:33.15 | yottanami | [TK]D-Fender: sorry I did not underestand, Is it wrong that my device registers to *? |
| 19:35.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:35.44 | yottanami | [TK]D-Fender: My dialplan is here http://www.dpaste.com/0X49KN7/ |
| 19:36.59 | yottanami | [TK]D-Fender: How can I prove that peer has registered ? It works for incoming calls |
| 19:38.19 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:41.42 | [TK]D-Fender | "sip show peers" <- you'll either see an IP listed.. or not |
| 19:42.03 | [TK]D-Fender | exten => _9N.,1,Dial(SIP/newrock//${EXTEN:1}) <- you did a double // |
| 19:42.07 | [TK]D-Fender | remove one |
| 19:43.50 | yottanami | [TK]D-Fender: Here is my sip show peers output http://www.dpaste.com/2EKH000/ I think it is registered, am I right? |
| 19:45.04 | yottanami | [TK]D-Fender: What should I use for context instead of default ? |
| 19:45.37 | [TK]D-Fender | You look like you have MULTIPLE things registered to it... |
| 19:45.46 | [TK]D-Fender | you should use ANOTHER context. |
| 19:45.59 | [TK]D-Fender | Doesn't matter what else you call it so long as the name is legal |
| 19:49.22 | yottanami | [TK]D-Fender: Multiple? Should I remote something ? |
| 19:49.45 | [TK]D-Fender | loks like you set up 2 profiles on your gateway |
| 19:51.53 | yottanami | [TK]D-Fender: Do you mean "newrock/newrock" and "0/newrock" ? |
| 19:52.00 | [TK]D-Fender | yes |
| 19:53.02 | yottanami | [TK]D-Fender: "0/newrock" is for FXO on gateway I connected a regular phone to it |
| 19:53.17 | [TK]D-Fender | those should not be the same peer |
| 19:53.28 | [TK]D-Fender | the FXO & FXS should be different accoutns |
| 19:53.40 | [TK]D-Fender | they are completely separate |
| 19:58.09 | *** join/#asterisk jonno11 (~jonno11@amigopod.rave.ac.uk) |
| 19:58.54 | yottanami | [TK]D-Fender: Here is my full sip.conf should I define some other thing like [newrock] ? |
| 19:59.59 | yottanami | [TK]D-Fender: FXS use 777 number and I get it with exten => 777,1,Goto(mainmenu,s,1) |
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| 20:00.59 | *** join/#asterisk mirela666 (~mirko.bra@95.180.121.169) |
| 20:01.21 | [TK]D-Fender | you should make a separate peer for each FXS port |
| 20:01.38 | [TK]D-Fender | And stop over-associated dialplan lines to SIP peers |
| 20:02.06 | [TK]D-Fender | Each peer should point to a proper context containing things that peer should be able to dial. |
| 20:02.12 | *** part/#asterisk warc (~warc@d51A490D7.access.telenet.be) |
| 20:05.17 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 20:17.11 | *** join/#asterisk yottanami (~yottanami@5.237.111.21) |
| 20:17.32 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 20:18.54 | yottanami | [TK]D-Fender: sorry again I disconnected ! so accept my apologizes to repeat the question |
| 20:18.54 | yottanami | <PROTECTED> |
| 20:18.54 | yottanami | <PROTECTED> |
| 20:19.20 | [TK]D-Fender | Yes you need a new user & pass for the FXS |
| 20:20.22 | yottanami | [TK]D-Fender: What about stoping over-associated dialplan lines to SIP peers ? how should I wrote it ? |
| 20:20.58 | *** join/#asterisk imox (~imox@p57A96DBC.dip0.t-ipconnect.de) |
| 20:22.14 | [TK]D-Fender | just fix the rest and test it |
| 20:25.11 | *** part/#asterisk fireman_biff (~biff@208.0.98.13) |
| 20:26.57 | *** join/#asterisk mirela666 (~mirko.bra@95.180.121.169) |
| 20:30.26 | *** join/#asterisk CeBe (~CeBe@port-92-206-103-120.dynamic.qsc.de) |
| 20:31.40 | [TK]D-Fender | checkout time, BBL |
| 20:36.10 | yottanami | What should I set for permit for allowing all ip addresses ? It can not accept 0.0.0.0 |
| 20:38.38 | TazzNZ | yottanami: you can't ask someone to "do your work for you" |
| 20:38.47 | Qwell | TazzNZ: He can ask. |
| 20:39.06 | TazzNZ | ok - sorry - yes he can, but should expect it :) |
| 20:40.31 | TazzNZ | shouldn't* |
| 20:41.42 | yottanami | TazzNZ: You mean I should not ask my questions about Astersik during learning that ? |
| 20:42.52 | yottanami | <PROTECTED> |
| 20:43.26 | TazzNZ | no yottanami - You are asking "What context should I put X into" - that is totally for you to decide. As for the permit - only you would know what IP's you *can* allow |
| 20:43.43 | TazzNZ | you need to help us, help you |
| 20:44.55 | TazzNZ | you need to at least appear to come up with some of the solution |
| 20:45.40 | yottanami | TazzNZ: Sorry maybe it is because of my poor English I don't want limit IPs, I want allow ANY ip addresses |
| 20:45.40 | yottanami | anyone from anywhere |
| 20:47.22 | Qwell | Why are you setting the permit/deny options, if you don't want to prohibit anyone from accessing it? |
| 20:49.13 | TazzNZ | Qwell: did you used to hang out in the PvPGN channel ? |
| 20:49.25 | TazzNZ | didn't* (geez - typing is shocking today) |
| 20:51.40 | Qwell | never heard of it |
| 20:51.55 | yottanami | Qwell: When I remove allow/deny I have Wrong password error when I want to connect from another IP range but in local it is ok ! |
| 20:52.52 | TazzNZ | Qwell: hhmmm - I could have sworn we have crossed paths in the past.... |
| 20:53.09 | Qwell | People know me. I'm kind of a big deal. |
| 20:53.25 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 20:53.33 | TazzNZ | yottanami: are you trying to register an end point to your asterisk ? |
| 20:53.48 | TazzNZ | Qwell: you a server/network op ? |
| 20:54.05 | Qwell | TazzNZ: I get around. |
| 20:54.18 | TazzNZ | right :) will leave it at that |
| 20:54.56 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 20:56.33 | yottanami | TazzNZ: I am registering a remote softphone |
| 20:57.25 | TazzNZ | yottanami: put host=dynamic in the phone's sip.conf section |
| 21:05.24 | yottanami | TazzNZ: Thanks it works |
| 21:05.32 | yottanami | I want to create outboundig call I create this dial plan http://www.dpaste.com/3YTEGFX/ and include it in [default] |
| 21:05.32 | yottanami | and here is my sip.conf but I got this error http://www.dpaste.com/2S9FQTE/ |
| 21:05.50 | *** join/#asterisk LemensTS (~T16@8.33.19.98) |
| 21:06.12 | LemensTS | Is there any options to the Park CMD? exten => 700,n,Park |
| 21:06.30 | LemensTS | core show application park ... does not give me any information |
| 21:06.30 | TazzNZ | but D-Fender already warned against using default ? |
| 21:07.24 | LemensTS | My timeout is 45seconds instead of the 120seconds I have defined in res_parking.conf. I am thinking its because I am using manual Park cmd instead of including parking? |
| 21:10.34 | LemensTS | Cancel that. Looks like res_parking.conf was put in asterisk 12. Im on asterisk 11. But installed asterisk 12 first then down graded and left old config files still in place. |
| 21:10.39 | *** part/#asterisk LemensTS (~T16@8.33.19.98) |
| 21:10.59 | *** join/#asterisk SpaceInvaders (~SpaceInva@adsl-74-235-60-95.clt.bellsouth.net) |
| 21:13.03 | SpaceInvaders | Hi! I want voicemail (on a server, preferably that I can access via my computer) and I'd like to add an extension (or secret code that rings my phone to kill marketing calls). I'd also like my voicemail accessible remotely via phone. It there a pre-packaged solution I should be looking at? What I've been reading on Asterisk makes it seem more like a platform for development. |
| 21:14.01 | SpaceInvaders | I'm downloading the Oreilly Asterisk The Definitive Guide 3rd edition. I just wanted to know if I was looking in the right places |
| 21:14.01 | TazzNZ | SpaceInvaders: imho, Asterisk can do what you want, but is not the solution you are looking for |
| 21:14.25 | SpaceInvaders | I was wondering. Can you make a few recommendations? Or throw me a pointer? |
| 21:17.31 | pjensen00 | Are you hoping for an "out of the box" solution? |
| 21:18.16 | TazzNZ | yottanami: I need to see more of extension.conf and sip.conf |
| 21:18.21 | SpaceInvaders | I'm ok with having to do extremely minor customization or I can compile code. |
| 21:18.23 | TazzNZ | to be able to help |
| 21:18.56 | SpaceInvaders | I was hoping for something that runs on Linux so I could run it on one of my existing servers |
| 21:19.31 | SpaceInvaders | I had a friend that had his computer answering his phone for him in the 80s and I can't believe its still so hard to do |
| 21:20.07 | TazzNZ | SpaceInvaders: it's not - I have that currently - but I also have SIP phones (or an ATA to connect my old analog phone) |
| 21:20.51 | yottanami | TazzNZ: Here is my full extension.conf and full sip.conf is here http://www.dpaste.com/152EF57/ |
| 21:21.55 | TazzNZ | you missing extension.conf |
| 21:21.56 | SpaceInvaders | I'll change all my phones to voip if I have to XD |
| 21:22.44 | TazzNZ | SpaceInvaders: and you need to convert your incoming line to SIP as well (either via an adaptor or asking your provider if they will do SIP) |
| 21:23.03 | SpaceInvaders | probably via adapter after I split off my DSL |
| 21:23.33 | SpaceInvaders | so if I convert to SIP I can have my computer answer my phone and take messages and ring my phone in the house only if someone presses the right number? |
| 21:23.42 | yottanami | TazzNZ: sorry here is extension.conf http://www.dpaste.com/1FFKWM2/ |
| 21:23.45 | TazzNZ | SpaceInvaders: correct |
| 21:24.09 | SpaceInvaders | sweet. So is that something prepackaged for Asterisk -- like free code I can download? |
| 21:24.17 | SpaceInvaders | or is it "built in"? |
| 21:24.21 | TazzNZ | yottanami: you missing "incal" |
| 21:24.24 | TazzNZ | to start with |
| 21:24.30 | TazzNZ | SpaceInvaders: it's all built in |
| 21:24.33 | drmessano | ~asterisk |
| 21:24.33 | infobot | Asterisk is an open source telephony toolkit, or #asterisk on irc.freenode.net, or http://www.asterisk.org/ |
| 21:24.38 | SpaceInvaders | oh sweet |
| 21:24.41 | TazzNZ | SpaceInvaders: what Linux boxes do you have around ? |
| 21:25.22 | SpaceInvaders | I have a couple of Dells with i3 processors that would be handy to use |
| 21:25.28 | drmessano | Asterisk is a toolkit, and while it can be ENHANCED via APIs and other neato junk, it's a swiss army knife already |
| 21:25.31 | TazzNZ | what distros :) |
| 21:25.37 | SpaceInvaders | I just have to check on room/form-factor for hardware |
| 21:25.46 | yottanami | TazzNZ: It was default and I changed it to incal but I dont know where should I add it |
| 21:25.51 | SpaceInvaders | OH sorry! I'm on F20 across the board |
| 21:25.54 | SpaceInvaders | Fedora 20 |
| 21:26.23 | TazzNZ | yottanami: in extension.conf, you need to add a section called ical |
| 21:26.27 | TazzNZ | incal* |
| 21:27.02 | TazzNZ | and you need to move your sip devices into another context |
| 21:27.14 | TazzNZ | this sounds a lot like what you have been told already.... |
| 21:27.29 | TazzNZ | SpaceInvaders: "yum install asterisk" should do the trick |
| 21:27.43 | TazzNZ | do that - get a softphone and start playing with asterisk |
| 21:27.53 | TazzNZ | then look at converting your phone line etc |
| 21:28.13 | SpaceInvaders | what are you running for hardware? Not the computer but the telephony card? Or will it work with just a modem? |
| 21:28.36 | TazzNZ | SpaceInvaders: I use the digium hardware |
| 21:28.39 | SpaceInvaders | and I have a handy soft phone :) |
| 21:28.49 | TazzNZ | my brother uses Shagoma |
| 21:29.26 | SpaceInvaders | I've been a fan of digium since I saw them in the late 80s and was impressed by a demo and their capabilities |
| 21:29.27 | TazzNZ | I can't recall a modem that will work with Asterisk |
| 21:29.53 | SpaceInvaders | Are my needs simple? Can I get away under $1000 for a card? under $500? |
| 21:29.54 | paulc | SpaceInvaders: You can use an FXO+FXS ATA to do what you want.. Obihai have one, Grandstream do an "alright-ish" one (503? 703? I can't remember) |
| 21:29.56 | TazzNZ | more like 90's :D |
| 21:30.09 | Qwell | more like 00's |
| 21:30.19 | SpaceInvaders | wow sweet :D |
| 21:30.53 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:30.57 | paulc | SpaceInvaders: http://www.amazon.com/OBi110-Service-Bridge-Telephone-Adapter/dp/B0045RMEPI would do the trick |
| 21:31.24 | TazzNZ | geepers - that is cheap |
| 21:31.37 | SpaceInvaders | and http://www.neweggbusiness.com/Product/Product.aspx?gclid=COCDgOOR3L4CFWYV7AodrGIAgw&Item=9B-33-617-002&nm_mc=KNC-GoogleBiz&cm_mmc=KNC-GoogleBiz-_-pla-_-VoIP-_-9B-33-617-002&ef_id=UjzJNAAAASHfp3JU:20140602213057:s ? |
| 21:31.38 | paulc | Plug your phone line into the FXO port, your phone(s) into the FXS port, and point both ports to Asterisk.. it'll answer the phone, play whatever, get digits, etc, then transfer the call to your phone(s) if you tell it to.. |
| 21:31.45 | SpaceInvaders | and that one at newegg? |
| 21:31.55 | SpaceInvaders | yeah! $47.99!!!! |
| 21:31.59 | paulc | Same product. |
| 21:32.17 | SpaceInvaders | I thought that was the same but wanted to verify |
| 21:32.20 | SpaceInvaders | thank you |
| 21:33.02 | SpaceInvaders | Does Asterisk connect to the the Obihai OBI110 via ethernet to control it? |
| 21:33.15 | SpaceInvaders | meaning no "internal" hw required on the server? |
| 21:33.30 | TazzNZ | SpaceInvaders: connects via ethernet, but it doesn't "control" it |
| 21:33.30 | paulc | $40 for the Grandstream equivalent: http://www.amazon.com/GrandStream-HT503-1-FXS-Analog-Telephone/dp/B002H29TGA |
| 21:33.35 | paulc | although I'd take the Obihai any day |
| 21:33.42 | TazzNZ | there is a bit of configurations that you need to do on the hardware |
| 21:33.59 | paulc | SpaceInvaders: Yes - the ATA is a SIP end point, so it's on ethernet - your server doesn't need any extra hardware installed inside |
| 21:34.10 | SpaceInvaders | thanks ! |
| 21:34.56 | SpaceInvaders | I'm gonna start like suggested. I'll install Asterisk on my server and use a soft phone to become familiar and play with it. Then I'll look at picking up a OBI110 |
| 21:35.13 | TazzNZ | good idea imho :) |
| 21:35.24 | TazzNZ | ~book |
| 21:35.24 | 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 |
| 21:35.31 | TazzNZ | ah - there it is |
| 21:35.35 | TazzNZ | read that too SpaceInvaders |
| 21:35.38 | SpaceInvaders | THANKS!! I was on the 3rd. This one is more current |
| 21:40.03 | SpaceInvaders | one more question -- for the OBI110 (shown in the links we discussed) is there anything I need to know regarding performance or will it be more than sufficient for a 2-line home (non-commercial use)? |
| 21:40.44 | TazzNZ | SpaceInvaders: I doubt you getting that many calls that you need to worry about performance :) |
| 21:40.53 | TazzNZ | and you only have 1 line |
| 21:41.03 | TazzNZ | it should be fine |
| 21:41.13 | SpaceInvaders | do I have to purchase 2 OBI110 units if i have 2 lines? |
| 21:41.19 | SpaceInvaders | I have 2 linees |
| 21:41.41 | TazzNZ | I think it only have 2 ports |
| 21:41.43 | TazzNZ | so yeah |
| 21:41.45 | [TK]D-Fender | SpaceInvaders: as in 2 completely separate sets of copper than can each be on separate calls? |
| 21:42.24 | SpaceInvaders | Yes, like in the 1960s I have a *fancy* house with 2 lines, two separate numbers, two people can be on separate calls at the exact same time |
| 21:42.34 | SpaceInvaders | 2 real land lines :D |
| 21:42.43 | TazzNZ | SpaceInvaders: then you need 2 units |
| 21:42.44 | SpaceInvaders | im running my own museum here |
| 21:42.48 | TazzNZ | lol |
| 21:42.49 | SpaceInvaders | ahh ok |
| 21:43.26 | SpaceInvaders | and asterisk will control both--do voice mail messages reside on the server? |
| 21:43.32 | TazzNZ | yip |
| 21:43.51 | TazzNZ | Asterisk can do 100's if need be |
| 21:44.09 | SpaceInvaders | ok so the hw (the OBI110) is just the connectivity and Asterisk is all the feature/funciton/control (i love component systems like that) |
| 21:44.24 | TazzNZ | yip |
| 21:45.16 | SpaceInvaders | lights a candle, leans over his new copy of Asterisk: The Definitive Guide |
| 21:45.42 | TazzNZ | hope that is a long candle :) |
| 21:46.22 | yottanami | TazzNZ: about "you need to move your sip devices into another context" Do you mean I should create a context for each item in sip.conf ? |
| 21:46.45 | TazzNZ | not each, just all of them - and it will be in extension.conf |
| 21:46.54 | TazzNZ | you referance them in sip.conf |
| 21:47.00 | TazzNZ | using context=<bla> |
| 21:47.27 | paulc | SpaceInvaders: Yeah - the OBI110 lets you connect one (external) phone line, plus one "phone" (or multiple phones in parallel, like inside your house).. So 2 phone lines? Get 2 of them.. BUT! You'll then be able to access either line from any phone connected to either device, with a bit Asterisk magic in the mix :-) |
| 21:48.15 | TazzNZ | also, incoming calls can "hunt" a free phone |
| 21:49.27 | SpaceInvaders | paulc that makes sense and is kinda what I was anticipating...once Asterisk gets its hands on a live line you can do what you want with it |
| 21:50.01 | SpaceInvaders | like for instance for my wife when calls come in from autodialers she wont get them any more unless they know the secret code to punch... like "1" lol |
| 21:50.46 | paulc | SpaceInvaders: Yup, exactly right. There's a privacy manager that you can use, get people to record their names and screen calls etc.. block dodgy caller IDs in future.. all sorts.. lots you can do - just a case of how you glue it all together :-) |
| 21:51.23 | SpaceInvaders | oooh you just gave me an idea... when her uncle calls he won't have to put in a secret code if he calls from his home phone number :D |
| 21:51.30 | yottanami | TazzNZ: now I got this error for outcall http://www.dpaste.com/2M0BG98/ |
| 21:51.51 | paulc | SpaceInvaders: Yup - whitelisting is totally possible too |
| 21:51.55 | TazzNZ | yottanami: I need to see what you have changed |
| 21:52.03 | TazzNZ | so I need sip.conf and extension.conf again |
| 21:52.07 | SpaceInvaders | once I have it set up and running I'll have to come back here n share what Ive done and ask about more ideas like that. |
| 21:52.26 | TazzNZ | SpaceInvaders: get to the section about astdb.....then you can have some real good fun ! |
| 21:53.10 | SpaceInvaders | LOL this should be called party-in-a-box |
| 21:53.41 | yottanami | TazzNZ: I did not edit extension I just changed default to incal http://www.dpaste.com/1FFKWM2/ |
| 21:54.17 | SpaceInvaders | what's the current ata recommendation? Is there one that's better? More stable? longer lasting? |
| 21:55.00 | SpaceInvaders | ugh... I need to read the OBi110 specs. If it has one I'll feel stupid |
| 21:55.19 | *** join/#asterisk jql (~awinters@74.123.82.143) |
| 21:55.33 | paulc | SpaceInvaders: I'm running off to a meeting.. I've played with the Grandstream HT503 and it worked well. But for pedigree, the Obihai is great - it's from the guys who were previously Linksys/Sipura, and they've always put out a quality product. |
| 21:55.50 | paulc | For the price of the Obi's - hard to go wrong really |
| 21:56.10 | SpaceInvaders | Yep I just found the specs for the Obi. It looks like that's really all I will need for hw (at 1 per line) |
| 21:56.13 | TazzNZ | I agree with paulc - I have a few linksys/cisco's in my past, they are good, but $$$ |
| 21:56.16 | SpaceInvaders | thank you paulc !! |
| 21:56.29 | SpaceInvaders | thank you too, TazzNZ |
| 21:57.12 | TazzNZ | yottanami: I still can't see the context incal |
| 21:57.27 | TazzNZ | did you do a "core reload" after your changes ? |
| 21:57.56 | yottanami | TazzNZ: http://www.dpaste.com/0C2MCR8/ http://www.dpaste.com/1M8V6EE/ |
| 21:58.48 | TazzNZ | yottanami: right - the problem then seems to be with the trunk |
| 21:59.25 | yottanami | TazzNZ: You mean [newrock] ? |
| 21:59.32 | TazzNZ | 34493472 <-- is that a valid number ? |
| 21:59.43 | TazzNZ | yottanami: yes |
| 22:00.14 | TazzNZ | bbib - meeting |
| 22:00.15 | yottanami | TazzNZ: Yes it is |
| 22:01.37 | yottanami | TazzNZ: Now the error is http://www.dpaste.com/0BXS83A/ |
| 22:06.09 | yottanami | TazzNZ: It some times detect the 9 at first some times no http://www.dpaste.com/2YJY6SW/ |
| 22:11.05 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 22:14.35 | [TK]D-Fender | yottanami: "sip set debug on" <--- show a new call |
| 22:16.25 | yottanami | [TK]D-Fender: http://www.dpaste.com/14F9B9F/ |
| 22:17.32 | [TK]D-Fender | SIP/2.0 404 Not Found To: <sip:64493472@192.168.1.88>;tag=1401747321116-1 |
| 22:17.44 | *** join/#asterisk navaismo (~navaismo@187-178-254-98.dynamic.axtel.net) |
| 22:17.45 | [TK]D-Fender | Your gateway does not know what to do with the number you gave it |
| 22:17.51 | [TK]D-Fender | You need to set up your gateway |
| 22:20.22 | yottanami | [TK]D-Fender: Wy it does not remov the 9 at start of number ? |
| 22:20.42 | [TK]D-Fender | To: <sip:64493472@192.168.1.88>;tag=1401747321116-1 <-- do YOU see a 9 in front? |
| 22:21.10 | [TK]D-Fender | <PROTECTED> |
| 22:21.56 | yottanami | [TK]D-Fender: Yes. now What should I set in my gateway ? |
| 22:22.22 | [TK]D-Fender | you should look at the settings you ahve to make for it to accept those #'sa you are sending it. |
| 22:22.26 | [TK]D-Fender | Read the manual |
| 22:23.09 | yottanami | [TK]D-Fender: ok tnx |
| 22:27.56 | *** join/#asterisk TodWulff (~TodWulff@unaffiliated/todwulff) |
| 22:28.05 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 22:29.48 | *** join/#asterisk remoford (~remoford@c-68-52-35-32.hsd1.tn.comcast.net) |
| 22:30.46 | remoford | does anyone have experience with the dynamic range compression settings? |
| 22:32.43 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 22:33.42 | remoford | to explain, ive got a conference call with a large volume spread amongst participants |
| 22:33.55 | remoford | i used to just turn it up to where i could understand the softest people and let the loud people clip |
| 22:34.19 | remoford | but now i need to take the call from the car and the loud people are going to blow out my speakers |
| 22:34.30 | [TK]D-Fender | or you could just bump the gain on THEIR channel..... |
| 22:34.44 | remoford | not being able to get this feature in android, all i can think to do is setup an asterisk server to pipe the call through and compress |
| 22:34.53 | [TK]D-Fender | there is no compression option in * |
| 22:35.11 | remoford | i dont have any ability to adjust anything ont eh bridge number? |
| 22:35.12 | remoford | none? |
| 22:35.32 | [TK]D-Fender | you can adjust the gains of each channel individually |
| 22:35.48 | [TK]D-Fender | "core show function VOLUME" |
| 22:35.52 | remoford | http://doxygen.asterisk.org/trunk/chan_dahdi.conf.html |
| 22:36.06 | remoford | no i dont have asterisk already |
| 22:36.13 | remoford | im not running the conference call on asterisk |
| 22:36.18 | [TK]D-Fender | that what good is DAHDI? |
| 22:36.28 | remoford | i want to setup asterisk and route my call through it |
| 22:36.32 | remoford | and do compression there |
| 22:36.59 | remoford | i would be willing to pay for a service to do this but have been unable to find such |
| 22:37.03 | [TK]D-Fender | * doesn't do compression |
| 22:37.19 | remoford | rxdrc: dynamic range compression for the rx channel. Default: 0.0 |
| 22:37.24 | remoford | so what are these settings for? |
| 22:38.18 | [TK]D-Fender | Hrm.. that is indeed news to me.. |
| 22:38.34 | [TK]D-Fender | So you'd have to run your calls through a DAHDI card to use this... |
| 22:39.02 | remoford | i guess? |
| 22:39.10 | remoford | i dunno im at my wits end to solve this problem |
| 22:39.18 | remoford | looking for any solution |
| 22:39.41 | [TK]D-Fender | Where is the conference actually being run off of? |
| 22:39.53 | remoford | i guess i could buy two cards and wire them directly to each other |
| 22:40.04 | remoford | i have no idea, corporate something |
| 22:40.15 | remoford | its a black box i have no privlidged access to |
| 22:40.22 | [TK]D-Fender | So far if you just want to sit in the middles sounds like you need a 2-pot card |
| 22:40.25 | [TK]D-Fender | port |
| 22:40.41 | remoford | i guess that rules out running asterisk in a vm |
| 22:40.48 | remoford | i would need a dedicated machine |
| 22:41.43 | [TK]D-Fender | pretty much... |
| 22:41.50 | [TK]D-Fender | There is one other possible solution. |
| 22:42.09 | [TK]D-Fender | You could try app_jack and use a JACK powered compression tool. |
| 22:43.17 | remoford | i find it hard to belive im the first person to want to do this |
| 22:43.24 | remoford | but google isnt giving me any love |
| 22:44.25 | *** join/#asterisk MauriceM_ (~MauriceM_@66-193-40-64.static.twtelecom.net) |
| 22:47.33 | remoford | uh, looks like the jack project page has been taken down |
| 22:49.14 | [TK]D-Fender | JACK still exists |
| 22:49.25 | [TK]D-Fender | you mean the * app to interface with it? |
| 22:49.49 | remoford | i mean "The jackaudio.org website is temporarily shutdown due to a deep hack by the leeches who post pharmaceutical spam. " |
| 22:49.56 | remoford | taking the documentation with it |
| 22:50.30 | remoford | therefore making finding/writing a jack powered compression tool difficult |
| 22:51.55 | [TK]D-Fender | Might be some ready-made plugins for it... |
| 22:52.08 | [TK]D-Fender | Never did any real playing around personally, but I kow there's stuff out there... |
| 22:52.16 | [TK]D-Fender | it's an options depending on how much time you have. |
| 22:53.19 | remoford | very little |
| 22:53.28 | remoford | but not solving the problem isnt an option |
| 22:54.03 | remoford | i really wish it were something you could just do in android directly |
| 22:54.28 | remoford | but the only thing ive found requires rooting the phone, everything else is music only |
| 22:54.43 | [TK]D-Fender | ok, it's all I have to suggest for this.. the DAHDI approach may work and you'd only have to take in 1 line if you want to use the other end as VoIP, etc |
| 22:54.44 | remoford | i guess they dont like people playing with streams off the baseband |
| 22:55.10 | remoford | id need both ends as voip |
| 22:55.21 | [TK]D-Fender | No, it's simply a lack of app.... I'm pretty sure a DSP app could be written to accomodate this... just that it hasn't been made yet |
| 22:55.48 | [TK]D-Fender | So you're looking to "loop" the connection just to let the card do its dirty-work? |
| 22:57.23 | remoford | ive got a cellphone on one end and a black box bridge number on the other end |
| 22:57.36 | remoford | anything that gets compression accomplished between the two satisfies my requirement |
| 22:57.49 | remoford | i was thinking if asterisk can do it by hook or by crook |
| 22:57.58 | [TK]D-Fender | This might do it for you... |
| 22:58.02 | remoford | i could rent a vm, subscribe to a voip service |
| 22:58.05 | remoford | and route the call through it |
| 22:58.23 | [TK]D-Fender | no point in that... might as well go direct since you have to have you card in there |
| 22:58.34 | remoford | ? |
| 22:58.50 | remoford | i dont understand what you mean by go direct |
| 22:58.51 | [TK]D-Fender | why have another intermediary server? |
| 22:59.15 | [TK]D-Fender | Or are you running on the hopes of a JACK-based solution? |
| 22:59.33 | remoford | that is currently my only hope obiwan |
| 22:59.46 | remoford | well that and rooting the cellphone and hoping this cynaogen app from 2010 works |
| 23:00.29 | [TK]D-Fender | Well.... you could by a guitar compression pedal and just wire that into your cell-phone's headset jack :) |
| 23:00.48 | [TK]D-Fender | http://cms.rolandus.com/assets/images/products/gallery/cs_3_top_gal.jpg |
| 23:00.55 | remoford | seriously? |
| 23:00.59 | remoford | thats a fantastic idea |
| 23:01.02 | [TK]D-Fender | Desperate situations call for hackish musician solutions :p |
| 23:01.09 | remoford | i would make a bluetooth mitm compressor |
| 23:01.18 | remoford | that might totally work |
| 23:01.35 | remoford | and be a damn sight less expensive |
| 23:01.39 | [TK]D-Fender | You could use the headset jack out, then use a 1/8 BT transmitter to go out.... |
| 23:01.43 | [TK]D-Fender | uber-hack, and portable... |
| 23:01.48 | [TK]D-Fender | FUGLY shit though |
| 23:02.13 | [TK]D-Fender | FrankenGuyver |
| 23:02.31 | remoford | but exactly what i need |
| 23:02.33 | remoford | omg |
| 23:02.43 | remoford | i could kiss you |
| 23:02.46 | remoford | but i wont |
| 23:02.54 | [TK]D-Fender | For which I thank you |
| 23:04.37 | [TK]D-Fender | time to head out for a bit... back later. |
| 23:04.53 | remoford | thanks man |
| 23:05.23 | [TK]D-Fender | np |
| 23:49.45 | *** join/#asterisk jonno11 (~jonno11@86.28.150.71) |
