| 00:08.54 | Penguin | xandrix: If sound files do not exist, Playback() cannot play them. |
| 00:09.38 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 00:10.55 | Synx|hm_ | ChannelZ: is there some dtmf debug settings so i can see what asterisk is parsing? |
| 00:17.40 | navaismo | enable it via logger.conf |
| 00:19.29 | Synx|hm_ | thanks |
| 00:20.05 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 00:20.35 | WIMPy | Does anyne have an idea, why an IAX trunk woun't register? |
| 00:21.11 | WIMPy | I've got register=yes, but when I turn on debug, I don't see it sending any registration attempts. |
| 00:21.35 | WIMPy | OTOH it does send pokes and correctely qualifies the peer. |
| 00:24.21 | XandriX | Penguin: oh so thats why it reads the ip in my case char by char i wanted to do the same thing as ip but it read the output of a specific curl command |
| 00:25.10 | navaismo | you need a tts |
| 00:25.42 | XandriX | oooh |
| 00:25.51 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.111) |
| 00:25.52 | XandriX | crap lol ill haveto figure out a way to get one on that machine |
| 00:26.54 | navaismo | http://www.voip-info.org/wiki/view/Text-to-Speech+(TTS) |
| 00:27.39 | Penguin | xandrix: Playback() doesn't read your IP address. |
| 00:28.31 | XandriX | nope it doesnt |
| 00:28.38 | XandriX | bashes head on desk |
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| 00:31.09 | Synx|hm_ | can dtmfmode go in [general] of sip.conf? |
| 00:31.40 | Penguin | Yes, you should define it in the general section. |
| 00:32.23 | Synx|hm_ | damn i do not know what is going on here, even a call from one peer to another they are not getting DTMF |
| 00:32.44 | Synx|hm_ | but DTMF does make it out the WAN to my sip carrier wtf |
| 00:34.08 | Synx|hm_ | and i've turned on dtmf console logging in logger.conf and i dont see anythign in the console :( |
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| 00:46.07 | XandriX | finaly got the curl script i wanted hehe |
| 00:46.14 | WIMPy | I wonder where I got that register= from. |
| 00:46.43 | WIMPy | It's just the usual thing: With things that just work, you tend to forget how to configure them. |
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| 00:54.53 | XandriX | if there are many replacements i take it built in festival sucks ? |
| 00:58.34 | Penguin | What is this "built in festival" you speak of? |
| 00:59.22 | XandriX | i must have read something wrong |
| 00:59.49 | XandriX | yep |
| 00:59.58 | XandriX | <PROTECTED> |
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| 01:04.05 | navaismo | Synx|hm_, did you ran logger reload? |
| 01:04.54 | Synx|hm_ | navaismo: just tried that |
| 01:04.59 | Synx|hm_ | still not seeing dtmf in the console |
| 01:06.26 | navaismo | show us the output of logger show channels |
| 01:07.14 | Synx|hm_ | Console Enabled - DEBUG NOTICE WARNING ERROR DTMF |
| 01:07.35 | XandriX | hrm i see no one whos built a package for a asterisk compatible tts engine for pfsense |
| 01:08.30 | navaismo | Synx|hm_, now do a test call to whatever number and press numbers during the call you may see something like: [Dec 3 19:07:44] DTMF[13451][C-00000000]: channel.c:4062 __ast_read: DTMF end passthrough '2' on PJSIP/5000-00000000 |
| 01:09.34 | Synx|hm_ | navaismo: thats what i am doing, i am calling from one sip peer direct to another right now in a simple test and sending dtmf between them |
| 01:11.50 | Synx|hm_ | navaismo: if instead of calling direct between local peers i send a call to a TFN out my sip carrier i can send DTMF to the IVR and it works however i still dont see anything in the asterisk console |
| 01:15.06 | bkruse | slaps Qwell around a bit with a large trout |
| 01:17.39 | Synx|hm_ | yay mIRC |
| 01:17.53 | *** join/#asterisk nny (~Scott@cpe-066-057-212-252.sc.res.rr.com) |
| 01:18.24 | nny | can I insert a comma in a Set statement like exten => _X,n,Set(answer=${answer},${EXTEN}) comma is part of the string? |
| 01:18.28 | XandriX | misses the trouting |
| 01:18.35 | nny | or do I need to escape it somehow? |
| 01:23.45 | [TK]D-Fender | nny: probably... Set does have a 2nd parameter |
| 01:26.53 | XandriX | Penguin: wich tts engine would you suggest ? espeak ? flite ? festival ? |
| 01:27.22 | Penguin | Probably cepstral. |
| 01:27.45 | XandriX | cepstral ? |
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| 01:32.11 | XandriX | looks nice |
| 01:39.05 | nny | [TK]D-Fender: any way to escape the comma as a part of the variable? |
| 01:39.28 | [TK]D-Fender | nny: I'd try the usual \, |
| 01:39.55 | nny | [TK]D-Fender: will test thanks |
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| 02:59.50 | saliak | I have a remote SIP extension that can call out fine, and receives calls fine, but suffers from the "no audio" problem, when I call from an extension on our local network, out to it (or vice versa). I have RTP and SIP ports opened to it on the NAT that it's behind. My asterisk server is on my firewall. I have RTP ports open to it as well on the external interface (which is what it connects through), and I have nat=yes in the sip.conf. S |
| 02:59.51 | saliak | debug of a call is at http://pastebin.com/TN0xxCEm. My only thought was that the internal call goes from phone to phone, so there needs to be an additional firewall rule that opens RTP from my phone on my internal network to the remote extension? I have a remote extension on a soft phone that works fine through IAX FWIW. is there any better way to test this than trying to make calls to someone at the remote site? |
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| 03:08.33 | XandriX | Penguin: dyou ever use festival ? |
| 03:09.10 | Penguin | I tried it once and found out it sounds terrible. |
| 03:09.30 | XandriX | darn |
| 03:09.42 | XandriX | apprently the person who made the asterisk package for pfsense forgot about tts |
| 03:10.07 | [TK]D-Fender | saliak: Peer audio RTP is at port 192.168.1.101:5004 |
| 03:10.27 | [TK]D-Fender | saliak: * is taking their internal IP for audio., Not good \ |
| 03:10.27 | XandriX | not completely but the only tie in module it has is app_festival nothing for espeak or anything else available from what i see |
| 03:10.33 | [TK]D-Fender | (line 486) |
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| 03:12.04 | XandriX | well atleast i know my code for the weather works its just real funny to hear it spell out cloudy 0 c |
| 03:12.23 | [TK]D-Fender | saliak: You are several revisions behind on your branch. Step 1 : UPGRADE |
| 03:13.15 | saliak | [TK]D-Fender: yeah, that's what I was thinking. isn't nat=yes supposed to fix that? about the internal ip? |
| 03:14.09 | saliak | [TK]D-Fender: is there any reason to not just go to 11.6.0? Or better to stay in 1.8? |
| 03:14.26 | [TK]D-Fender | saliak: Normally yes. There are other options to for comedia, force_rpot. |
| 03:14.58 | saliak | [TK]D-Fender: you think this could just be something that's resolved by upgrading then? |
| 03:15.06 | [TK]D-Fender | I'm a little grey on these others. Try them, then upgrade as there may have been SIP mapping fixes, etc |
| 03:15.07 | [TK]D-Fender | heads out for a while |
| 03:16.04 | XandriX | Penguin: the sound files you mentioned the other day is there a place i can get a zip with like a ton of words in us english as sound files ? |
| 03:17.08 | Penguin | asterisk.org |
| 03:18.22 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 03:19.25 | Penguin | xandrix: Specifically, http://downloads.asterisk.org/pub/telephony/sounds/ |
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| 03:19.49 | Penguin | http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-en-wav-current.tar.gz |
| 03:20.01 | Penguin | http://downloads.asterisk.org/pub/telephony/sounds/asterisk-extra-sounds-en-wav-current.tar.gz |
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| 04:31.28 | showy | hi guys, does exist a digital archive for astricons? |
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| 04:49.02 | ChannelZ | mmmm youtube |
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| 04:54.11 | showy | official channel? |
| 05:00.06 | hebber | For PLC how is your experience with Jitterbuffer? Many of my connections are over low quality ADSL connections. We have used ILBC so far and that was a great improvement. Would like to improve it further. |
| 05:05.28 | hebber | Is it worth trying? |
| 05:05.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.80) |
| 05:16.00 | ChannelZ | well jitter buffer is only applicable to receiving, and adds latency. |
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| 05:24.59 | TeknoJuce | Hi I have a unistim phone setup and cant get the phone to ring but I have voip software in the ring pool as well and it rings when it gets hit. |
| 05:25.01 | TeknoJuce | http://pastebin.com/kC4dPUwA |
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| 06:25.17 | hebber | channelz: then I guess its not worth it - thanks for feedback |
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| 07:04.11 | yottanami | <PROTECTED> |
| 07:04.11 | yottanami | How can I find out which IVR and witch option of that ran ? |
| 07:06.38 | kaldemar | yottanami: look at each line, it says -- Executing [<extension>@<context>:<priority>] |
| 07:22.11 | yottanami | kaldemar: What about IVR? for example there is "ivr-8" Is it the name of IVR? because I did not defaine that |
| 07:23.24 | *** join/#asterisk neider_ru (~neider@82-117-174-016.nts.su) |
| 07:23.32 | neider_ru | Good day, everyone |
| 07:24.29 | michael_work | good morning |
| 07:24.50 | kaldemar | yottanami: ivr-8 seems to be a context in your dialplan. |
| 07:25.34 | michael_work | when it's betetr to use ast_str and when ast_field |
| 07:25.35 | michael_work | ? |
| 07:25.40 | michael_work | better* |
| 07:26.11 | neider_ru | a little question about trunks, all my clients uses only g729, and one peer is on sems, that does not support g729. The problem is in sems, how to make calls on it using alaw, and all other calls using only g729 |
| 07:26.13 | neider_ru | ? |
| 07:26.38 | yottanami | kaldemar: Some other one designed the dialplan from the web interface ( I think It is freepbx ) How can I see dial plan from CLI ? |
| 07:26.44 | michael_work | you just configurate it to use ulaw :) |
| 07:27.03 | michael_work | yottanami, dialplan show $NAME |
| 07:27.12 | kaldemar | yottanami: dialplan show |
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| 07:35.50 | yottanami | kaldemar: The output is very long. Is any way to put it in a file ? |
| 07:36.31 | kaldemar | yottanami: asterisk -rx "dialplan show" > /tmp/ast_dialplan.txt |
| 07:36.46 | michael_work | yottanami, or use |tee command :) |
| 07:36.57 | kaldemar | yottanami: but you already should have your dialplan in files. extensions.conf and all files that are included from it. |
| 07:37.39 | michael_work | yottanami, cd /etc/asterisk && grep $DIALPLAN_NAME\] -r . |
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| 07:42.07 | yottanami | michael_work: It have a big output http://dpaste.com/1492196/ |
| 07:43.00 | michael_work | yottanami, what;'s the name of dialplan |
| 07:43.11 | michael_work | if the name is my_dialplan |
| 07:43.31 | michael_work | all you need to run "grep mydialplan\] -r /etc/asterisk" |
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| 07:43.36 | michael_work | do not forget \] |
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| 07:44.19 | yottanami | michael_work: My problem is I am not sure about which dialplan is running, How can I figure out which one is running ?here is my cli output http://dpaste.com/1492159/ |
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| 07:44.56 | kaldemar | yottanami: you only have 1 dialplan. it tells you where in your dialplan the call is on each line. |
| 07:45.02 | michael_work | yottanami, you have few there |
| 07:45.30 | kaldemar | stop calling a context a dialplan. |
| 07:46.23 | michael_work | who called it ? |
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| 07:47.03 | michael_work | [s@from-zaptel:1] means you in from-zaptel dialplan first line for extension s |
| 07:47.06 | michael_work | : |
| 07:47.08 | michael_work | :) |
| 07:47.19 | kaldemar | context, not dialplan. |
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| 08:35.59 | xytis | Hello, I am stuck with an asterisk bug. Bug is related to ICE session lifetime within an asterisk RTC session. |
| 08:36.17 | xytis | Could I ask for help here, or someone knows where I could get help from PJSIP maintainers? |
| 08:36.27 | xytis | Since I have a rough idea how to fix it, yet can not figure out the right way to do that. |
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| 09:03.14 | ibo23 | hi i'm trying out asterisk 12, does anyone know where the log file for pjsip is located? Or is there a way to see the log output in CLI |
| 09:06.04 | xytis | sip set debug on |
| 09:06.39 | xytis | then everything will get dumped both to console and the log file You have set up. |
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| 09:23.50 | ibo23 | xytis: i'm using pjsip, so sip set debug on won't work |
| 09:30.49 | TeknoJuce | I get no audible ringtone when you dial an ext to a unistim phone, anyone have any help for this scenario? |
| 09:32.38 | TeknoJuce | even tried the destintive ring tone option for the phone with no luck exten => 200,hint,USTM/200@violet1/r43 |
| 09:32.39 | TeknoJuce | <PROTECTED> |
| 09:39.34 | xytis | ibo23: I just checked the source in previous version of asterisk. Unfortunately as far as I can tell, You should recompile pjproject with logging enabled. Then You may trace where the logs go from pj_log function. I thought they share the same logger as sip. |
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| 10:20.01 | ibo23 | xytis: i just found out, i had to set core set debug on, silly me :) |
| 10:20.16 | ibo23 | xytis: core set debug 5 |
| 10:20.21 | ibo23 | xytis: core set verbose 5 |
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| 10:40.29 | jkroon | hi guys, I'm sitting with a situation where x-Lite is just dropping all of my calls, basically I send it INVITE, it comes back with 100 Trying, 180 Ringing, and then in the SDP in the 200 OK I'm not receiving any audio codecs back ... |
| 10:40.52 | jkroon | I'll quickly pastebin the SIP conversation |
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| 12:03.40 | jkroon | http://pastebin.com/bQh6Fkdm |
| 12:05.28 | jkroon | that all looks normal to me. |
| 12:06.08 | jkroon | the scenario is that i've got a client that can't transfer to x-Lite, now, from what I can tell (the above is the outbound to ext 2022 transfer from 2020 usnig the transfer button on eyeBeam) |
| 12:06.32 | jkroon | the only thing that worries me is that I get no audio codecs in the 200 OK packet from x-Lite |
| 12:07.45 | Greenlight | Howdy jaco :) It's not trying to directly bridge the RTP steams or something like that ? |
| 12:11.55 | jkroon | not that I can see no |
| 12:12.06 | jkroon | hmm, but that might be interesting to look at |
| 12:12.07 | jkroon | let's see |
| 12:12.59 | jkroon | well, Greenlight - the portion that bugs me is the packet starting at line 350 |
| 12:13.04 | jkroon | there is no audio codec in there. |
| 12:14.12 | Greenlight | Yea, but then at 378 it decides on alaw |
| 12:14.19 | jkroon | Greenlight, c=IN IP4 192.9.200.230 <-- in the SDP. |
| 12:14.34 | jkroon | but the RTP packet is coming from 192.9.200.24 |
| 12:14.48 | jkroon | directmedia=no might be the culprit |
| 12:15.13 | Greenlight | directmedia=no is correct |
| 12:15.24 | Greenlight | That'll force RTP through Asterisk |
| 12:16.01 | jkroon | yea, but could this whole mess be caused by eyeBeam trying to get asterisk to send audio to the wrong IP in combination with nat=yes? |
| 12:16.57 | Greenlight | Possibly :S |
| 12:18.15 | jkroon | ok, set nat=no |
| 12:18.18 | jkroon | let's see what happens. |
| 12:25.47 | jkroon | Greenlight, .230 is the server, so that was from the transmit ... i need a wakup |
| 12:34.27 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
| 12:35.16 | Greenlight | And this only happens when using the transfer button? They can call normally to eachother otherwise ? |
| 12:35.57 | jkroon | even the #ext# meganisme werk perfectly |
| 12:36.18 | jkroon | if they phone direct yes, it works. |
| 12:36.56 | WIMPy | Good morning |
| 12:37.14 | jkroon | something to do with REFER perhaps somewhere that I'm missing that doesn't get handled properly? |
| 12:37.30 | Greenlight | Yea sounds like it |
| 12:37.44 | jkroon | hmm, not even seeing REFER packets in /var/log/asterisk/full with sip debug on. |
| 12:37.50 | Greenlight | So, it's an attended or blind transfer ? |
| 12:38.08 | jkroon | semi-attended from what I understand. |
| 12:38.29 | Greenlight | So at first the caller is put on hold, and they are speaking to the transfer desination ? |
| 12:38.35 | jkroon | so transfer actually initiates a new call, but then as soon as it gets 180 ringing it completes the transfer |
| 12:38.47 | Greenlight | Right I see |
| 12:39.18 | Greenlight | So, we should be seeing a REFER ? |
| 12:39.25 | jkroon | and the number of NOTIFY packets being sent makes it difficult. |
| 12:39.27 | jkroon | indeed i should |
| 12:39.36 | jkroon | will grab a trace on the call i'm making now again. |
| 12:39.54 | Greenlight | And it's *after* the transfer is completed that the problem exists ? |
| 12:40.06 | jkroon | eyeBeam says transfer cancelled. |
| 12:40.10 | Greenlight | Oh, odd |
| 12:40.14 | jkroon | indeed. |
| 12:40.54 | Greenlight | Hmm.. how about you make say a exten => 5000,1,Playback(testmusic) |
| 12:40.59 | Greenlight | And try transferring to that |
| 12:41.21 | WIMPy | Ok, I've seen transfer cancelled a few times as well. |
| 12:41.40 | jkroon | WIMPy, have you figured out what caused it? |
| 12:42.04 | WIMPy | No. I haven't tried to find out. |
| 12:42.49 | WIMPy | I didn't see any pattern so far as to when or in what situations it happens. |
| 12:43.04 | jkroon | internal calls works just fine |
| 12:43.21 | WIMPy | It looks completely random to me. |
| 12:43.32 | jkroon | well, in this case it's pretty darn reliable. |
| 12:43.39 | jkroon | and there asterisk crashed |
| 12:43.40 | jkroon | :( |
| 12:43.47 | Greenlight | uh oh |
| 12:45.59 | jkroon | 11.6.0 |
| 12:46.02 | jkroon | so it's a problem. |
| 12:46.09 | jkroon | but not the one I've been hunting |
| 12:47.04 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 12:47.47 | Greenlight | So, the crash is unrelated to the transfer issue ? |
| 12:48.48 | jkroon | i think so yes |
| 12:55.01 | *** join/#asterisk grummund_ (~user@unaffiliated/grummund) |
| 12:57.29 | *** join/#asterisk yottanami (~yottanami@178.131.120.234) |
| 13:02.17 | jkroon | hmm |
| 13:03.12 | jkroon | ok, how about an rport issue? |
| 13:03.23 | jkroon | but that would affect normal calls too |
| 13:05.02 | jkroon | ok, i am stumped |
| 13:07.05 | *** join/#asterisk Greenlight (wluke@cpc1-dund9-0-0-cust142.16-4.cable.virginm.net) |
| 13:08.30 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:20.34 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 13:20.55 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
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| 13:29.30 | *** part/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM7cb21b15b251.cpe.net.cable.rogers.com) |
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| 13:58.02 | *** join/#asterisk dar123 (~DAR@94.98.13.38) |
| 13:59.28 | dar123 | hey guyz i want to upgrade dahdi, do i delete everything installed before and install the new version or it can be upgraded? |
| 13:59.53 | Greenlight | Just install the new version, it'll overwrite |
| 14:00.31 | dar123 | thanks |
| 14:01.30 | *** join/#asterisk serafie (~erin@nat/digium/x-rkpuscwivbhcivmw) |
| 14:02.21 | *** join/#asterisk dongola7 (~dongola7@unaffiliated/blair/x-0911782) |
| 14:02.45 | dar123 | do i need to do make config? |
| 14:03.29 | XandriX | i dont know what radio station this place uses for hold music but holycrap |
| 14:03.43 | XandriX | it started with the monster mash now its they blinded me with science |
| 14:05.14 | *** join/#asterisk yaaago (~kresp0@gateway/tor-sasl/kresp0) |
| 14:07.18 | saliak | I'm still trying to debug a no-audio situation between an external sip extension, and an internal sip extension. Things work when calling an outside line with either, but communication between the two doesn't work (call connects then dies). I'm assuming this is a RTP/NAT issue. On the remote extension rtp (10000-20000) is forwarded, and the handset has STUN set for nat traversal. nat=yes in sip.conf. upgraded to latest asterisk (11.4) |
| 14:07.19 | saliak | night. sip debug of failed call at http://pastebin.com/P9MWkXz1. 192.168.1.129 is my local extension LAN IP. 192.168.1.101 is my remote extension LAN IP. 72.195.155.142 is my asterisk server WAN IP. 70.188.147.97 is the WAN IP of the remote. I can't tell where in the SIP debug that things break |
| 14:09.36 | BeachBall | is still here |
| 14:10.02 | BeachBall | OMG WIMPy is undercover! |
| 14:11.16 | *** join/#asterisk kresp0__ (~kresp0@gateway/tor-sasl/kresp0) |
| 14:11.29 | [TK]D-Fender | For saliak's issue, Call placed at 241 with "nat=yes", on line 577 call ACK'd on WAN, but it is taking the IP for media from the 200 and not overriding with peer IP as dialed. |
| 14:11.36 | WIMPy | No, I'm on my chair. |
| 14:12.00 | XandriX | pokes BeachBall |
| 14:12.06 | BeachBall | glares |
| 14:12.13 | XandriX | yay a reaction :P |
| 14:12.23 | XandriX | what part of montreal u from / around ? |
| 14:12.36 | BeachBall | The east part |
| 14:12.50 | XandriX | like ahunsic ? |
| 14:12.58 | BeachBall | little further east than that |
| 14:13.15 | BeachBall | near Gaspe |
| 14:14.06 | BeachBall | kinda north east really |
| 14:14.07 | saliak | [TK]D-Fender: wow. that's impressive to pull that out quickly. so why would that be happening? Ihought nat=yes was supposed to fix that specifically? |
| 14:16.08 | [TK]D-Fender | saliak: This has the appearance of being a "bug", so 'd like a dev to look at it... |
| 14:16.38 | [TK]D-Fender | saliak: Figured I do you the favour of gift-wrapping the backup so you get the best help as fast as possible for it. |
| 14:17.07 | saliak | [TK]D-Fender: ok. how do I submit it? thanks for the wrap |
| 14:17.09 | [TK]D-Fender | DEV's, can someone with chan_sip experience ake a look at this with him? |
| 14:17.30 | [TK]D-Fender | saliak: Just pinging people here... suppose you could make an issue on the tracker, etc... |
| 14:18.11 | saliak | [TK]D-Fender: is there a workaround by any chance? |
| 14:18.20 | Penguin | First thing they'll say is to upgrade your 11.4 to 11.6. |
| 14:18.52 | [TK]D-Fender | User-Agent: Asterisk PBX 11.6.0 |
| 14:18.58 | [TK]D-Fender | he's on 11.6.0 |
| 14:19.11 | Penguin | Told us the wrong thing, then. |
| 14:19.16 | [TK]D-Fender | saliak: Not that I can see.. you're doing things right... |
| 14:19.57 | [TK]D-Fender | Penguin: Guess so... perhaps he upgraded but just copy/pasted from his previous request for help on that version... |
| 14:19.58 | saliak | Penguin: yeah, sorry, got the latest 11 last night. 11.6.0 |
| 14:21.10 | XandriX | BeachBall: oh north shore o.O |
| 14:21.51 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 14:22.37 | [TK]D-Fender | [09:13]BeachBallnear Gaspe <- Where the poulation density is so low it's a fine line between camping and homelesness... |
| 14:22.51 | BeachBall | :{ |
| 14:23.02 | BeachBall | thinks Defender is ready for his daily kick in the nuts |
| 14:23.08 | *** join/#asterisk mjordan (~matt@50-192-47-77-static.hfc.comcastbusiness.net) |
| 14:23.08 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:26.04 | dar123 | wctdm24xxp 0000:04:06.0: Missed interrupt. Increasing latency to 5 ms in order to compensate. <- is this normal |
| 14:26.26 | Greenlight | Missing interrupts is generally bad |
| 14:26.59 | dar123 | i have TDM410P, i will use wctdm24xxp driver correct |
| 14:28.37 | Greenlight | Is the card sharing an IRQ with another device, such as a network card or raid controller ? |
| 14:36.30 | XandriX | [TK]D-Fender: gaspe is awesome small communities are the best we got welcomed by the entire town when me and my buddy went down there for a 2 week vacation |
| 14:37.20 | [TK]D-Fender | XandriX: Of course... with so few faces around they figured you must be part of a rescue party :p |
| 14:37.34 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 14:37.39 | XandriX | lol |
| 14:37.49 | XandriX | [TK]D-Fender: that was a good one i must admit :P |
| 14:38.08 | [TK]D-Fender | OMG PEOPLE! WE'RE SAVED! |
| 14:38.35 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 14:39.32 | XandriX | [TK]D-Fender: where you from ? |
| 14:39.41 | [TK]D-Fender | XandriX: Montreal. |
| 14:41.20 | BeachBall | LMAO |
| 14:41.23 | BeachBall | bahahahaha |
| 14:43.51 | XandriX | bashes his head on the desk |
| 14:44.01 | XandriX | [TK]D-Fender: beer ? :P |
| 14:45.17 | *** join/#asterisk GTXComm (~John@72.135.4.232) |
| 14:47.40 | BeachBall | has anyone heard of a website that you can signup and they will stress test your network - DDoS you. |
| 14:47.48 | BeachBall | :/ |
| 14:48.37 | dar123 | i don't have IRQ conflicts |
| 14:49.48 | carrar | isn't that #teen on efnet? |
| 14:49.56 | XandriX | BeachBall: people on tor will gladly help you with that :P |
| 14:52.12 | BeachBall | i might of done a stupid thing :{ |
| 14:52.20 | BeachBall | upgrade my asterisk while it was still running |
| 14:53.11 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:53.11 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:53.16 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:53.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:54.06 | *** join/#asterisk smirker (Smirker@CPE-58-169-177-185.lns5.cha.bigpond.net.au) |
| 14:56.01 | *** join/#asterisk yottanami (~yottanami@5.237.113.75) |
| 14:56.18 | BeachBall | is a nose trimmer suppose to tickle? |
| 14:56.28 | [TK]D-Fender | Depends where you put it... |
| 14:58.28 | XandriX | just got a weird scene in his said in regards to the nose trimmer tickleing and it depends where you put it |
| 15:02.19 | Synx|hm_ | How does out of band DTMF traverse a network, specifically a firewall? If it occures on a seperate channel (RTP) as RFC2833 seems to imply how does a firewall know to forward that back to the correct device if there was not previously a state established? |
| 15:02.53 | [TK]D-Fender | Synx|hm_: There is a call state... it's negotiated in SDP |
| 15:05.09 | Synx|hm_ | [TK]D-Fender: ok, thanks, im trying to troubleshoot why i am not getting return DTMF from an answering endpoint outside my LAN |
| 15:08.58 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 15:10.15 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:10.15 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:22.47 | XandriX | needs to read up on IVR the possibilities look endless |
| 15:23.28 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:24.11 | *** part/#asterisk yottanami (~yottanami@5.237.113.75) |
| 15:25.17 | *** join/#asterisk stefmtl (~stef@69-165-193-197.cable.teksavvy.com) |
| 15:29.48 | stefmtl | Hi, I enabled automixmonitor in my features.conf, and using the app Dial option Xx in my dialplan, but the recording is in WAV format. How do I change to another format (gsm for example) ? Do I have to set the monitor format in features.conf ? Also how do I change the default output folder ? I read all the sample files but I can't figure it out, thanks for your support |
| 15:34.54 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 15:36.37 | *** join/#asterisk adeel (~adeeln@199.48.103.26) |
| 15:43.13 | *** part/#asterisk mjordan (~matt@50-192-47-77-static.hfc.comcastbusiness.net) |
| 15:44.10 | ipengineer | Hello everyone! Is func_realtime working in v12? I am getting no application 'REALTIME': exten=>lookupdid,n,REALTIME(outbound_cid,id,${PEERNAME}) |
| 15:45.13 | [TK]D-Fender | ipengineer: because .. it is a FUNCTION |
| 15:48.07 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.70) |
| 15:48.39 | ipengineer | [TK]D-Fender: So do I need to use that within a SET and store it as a variable? |
| 15:49.15 | [TK]D-Fender | ipengineer: or do "whatever" with it |
| 15:49.41 | stefmtl | nobody knows how to use the Dial Xx options and default monitor format/directory please ? |
| 15:53.20 | ipengineer | [TK]D-Fender: Ok thanks.. makes sense now. |
| 15:54.31 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.70) |
| 15:56.48 | *** join/#asterisk pietro (~pietro@88-149-210-128.v4.ngi.it) |
| 15:56.56 | pietro | hello |
| 15:57.09 | stefmtl | hello |
| 15:57.17 | stefmtl | I enabled automixmonitor in my features.conf, and using the app Dial option Xx in my dialplan, but the recording is in WAV format. How do I change to another format (gsm for example) |
| 15:57.23 | stefmtl | <PROTECTED> |
| 15:57.53 | pietro | I just opened this issue (with patch attaced) https://issues.asterisk.org/jira/browse/ASTERISK-22939 can someone assign it or add to the roadmap ? |
| 15:58.36 | ipengineer | How can we get pjsip channel info. Similar to SIPCHANINFO |
| 16:01.59 | file | there is no function to do that at this time |
| 16:02.05 | file | but mjordan has two reviews up for it |
| 16:02.13 | ipengineer | file: will CHANNEL() not work? |
| 16:02.25 | file | https://reviewboard.asterisk.org/r/3038/ and https://reviewboard.asterisk.org/r/3035/ |
| 16:02.35 | file | no because it wasn't written, that is what mjordan has up for review |
| 16:02.37 | ipengineer | I am trying to get the peer name so I don't know the best way to do that currently is |
| 16:02.51 | file | there is no way currently without the above |
| 16:03.13 | ipengineer | file: Ok. Thanks I will keep an eye on it |
| 16:03.50 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:05.02 | *** join/#asterisk Greenlight (wluke@cpc1-dund9-0-0-cust142.16-4.cable.virginm.net) |
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| 16:13.44 | zamba | everytime i reboot my ubuntu server i have issues with the DAHDI module needing rebuilding |
| 16:13.49 | zamba | what can i do to fix this? |
| 16:14.08 | zamba | i thought dkms was supposed to fix this? |
| 16:16.37 | *** join/#asterisk lnb (~lnb@CPE000347b24a71-CM602ad06bec2f.cpe.net.cable.rogers.com) |
| 16:17.32 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
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| 16:22.23 | lnb | hi, have a script that when users press *2000 it acts as a punch clock. What is supposed to happen when punch in a recording should play 'you-have-successfully-punched-in.wav' but it doesn't. It only plays the standard goodbye recording. Watching it in cli I see the following: -- Executing [*2000@from-internal:17] NoOp("SIP/270-00002708", "Unmatched option for stat: ..0..") in new stack |
| 16:25.56 | *** join/#asterisk nickw1234 (~nick@66-188-11-134.static.bycy.mi.charter.com) |
| 16:27.46 | lnb | in extensions_custom.conf the line is there ' exten => *2000,n,NoOp(Unmatched option for stat: ..${stat}..)' but it doesnt see to run the next 3 lines but goes direct to goodbye... exten => *2000,n(punched-in),Playback(custom/you-have-successfully-punched-in) |
| 16:27.46 | lnb | <PROTECTED> |
| 16:27.46 | lnb | <PROTECTED> |
| 16:27.46 | lnb | <PROTECTED> |
| 16:29.31 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:33.35 | *** join/#asterisk navaismo (~navaismo@189.241.125.73) |
| 16:35.36 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 16:35.47 | pietro | I just opened this issue (with patch attaced) https://issues.asterisk.org/jira/browse/ASTERISK-22939 can someone assign it or add to the roadmap ? |
| 16:44.08 | [TK]D-Fender | pietro: Affects Version/s: 1.8.15.0 <- this is not current. retest against current and update if still applicable |
| 16:44.19 | [TK]D-Fender | We're at 1.8.24 |
| 16:45.13 | file | pietro, it will get triaged as soon as possible |
| 17:00.47 | pietro | [TK]D-Fender: thanks, 1.8.24 is affected too. I updated the JIRA. (I mentioned 1.8.15 because is the last certified version). |
| 17:00.52 | pietro | thanks file |
| 17:01.04 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 17:05.54 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
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| 17:32.30 | XandriX | lnb: is $(stat) declared in that file ? |
| 17:33.00 | lnb | XandriX: which file? |
| 17:33.23 | XandriX | extensions_custom |
| 17:33.40 | lnb | in extensions_custom.conf ? |
| 17:33.48 | XandriX | yes |
| 17:33.49 | lnb | i think so, one sec |
| 17:36.23 | lnb | what i am pasting is from where the user would enter their userid/pin |
| 17:36.25 | lnb | <PROTECTED> |
| 17:36.25 | lnb | <PROTECTED> |
| 17:36.25 | lnb | <PROTECTED> |
| 17:36.25 | lnb | <PROTECTED> |
| 17:36.25 | lnb | <PROTECTED> |
| 17:36.26 | lnb | <PROTECTED> |
| 17:37.04 | WIMPy | ~pb |
| 17:37.04 | 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. |
| 17:37.20 | navaismo | slap lnb |
| 17:37.23 | XandriX | other quick question um (people of the channel correct me if im wrong) but technicaly after your first exten => your other entries should be same => if i am not mistaken |
| 17:37.44 | navaismo | not necessary |
| 17:37.55 | XandriX | oh haha |
| 17:38.04 | XandriX | thx for the clearup navaismo :) |
| 17:38.24 | lnb | after (done) should play either logged in success or failure. But what it does is play the standard 'goodbye' |
| 17:38.26 | navaismo | same is like a shorcut to avoid to rewirte again exten => blablabla |
| 17:38.52 | lnb | navaismo: yes, one gets to save one letter |
| 17:39.17 | lnb | actually no, same/exten same number of characters |
| 17:39.27 | XandriX | no |
| 17:39.40 | XandriX | same = 4 chars exten = 5 |
| 17:39.40 | navaismo | wondering why this is not in the #freepbx? |
| 17:39.42 | lnb | exten has 1 more my bad |
| 17:40.08 | XandriX | navaismo: some of us dont use freepbx ? :P |
| 17:40.13 | [TK]D-Fender | XandriX: you also don't have to repeat the pattern.. so it's not a 1-char savings |
| 17:40.31 | XandriX | [TK]D-Fender: o.O |
| 17:40.54 | [TK]D-Fender | exten => 1NXXNXXXXXX,1,NoOp(LONG) |
| 17:40.57 | XandriX | good to know |
| 17:41.07 | [TK]D-Fender | same => n,NoOp(SHORT) |
| 17:41.16 | [TK]D-Fender | You missed the "big print" somehow... |
| 17:41.17 | XandriX | sweet |
| 17:41.32 | navaismo | XandriX, i mean lnb use freepbx |
| 17:41.44 | XandriX | navaismo: aah :) |
| 17:42.03 | lnb | navaismo: freepbx is on top of asterisk |
| 17:42.05 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.89) |
| 17:42.12 | lnb | and what is wrong with asking about it here |
| 17:42.17 | lnb | its not a freepbx issue |
| 17:42.24 | navaismo | yeah but it uses different macros and stuff |
| 17:42.37 | lnb | what macro? |
| 17:42.47 | lnb | there are php scripts for this |
| 17:42.54 | XandriX | lnb: freepbx does make certain syntax / macro changes from what i heard a while ago and like navaismo just mentioned |
| 17:42.57 | lnb | which are not part of freepbx |
| 17:43.04 | lnb | ok |
| 17:43.06 | lnb | thats cool |
| 17:43.14 | navaismo | shrugs |
| 17:43.18 | lnb | and perhaps navaismo is right |
| 17:43.19 | [TK]D-Fender | not "certain", more like "ALL" |
| 17:43.33 | lnb | i have asked in freepbx before but it was never solved |
| 17:43.45 | lnb | functionally, it does update the timeclock db |
| 17:43.56 | lnb | for both punch-in and punch-out |
| 17:44.07 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 17:44.07 | XandriX | im testing out the asterisk package for pfsense it works quite well but the bastards forgot to implement other tts engines than festival and they give you the module for festival but the package does not include festival and any festvox voices |
| 17:44.25 | [TK]D-Fender | With FreePBX , it is configuring your system, not you. You can't just change anything you want in there... the config files get blown away when you apply changes unless you do it in specifically allowed places and that only allows you a certain level of customizability anyway |
| 17:44.32 | [TK]D-Fender | You need to know what you're doing. |
| 17:44.37 | lnb | but the person doing the actions never knows if it is really working because all they hear is, 'enter your userid, enter your pin -> goodbye |
| 17:45.33 | lnb | well the question remains, why the caller hear 'enter your user/pin' but not hear you've successfully punched in? |
| 17:45.36 | XandriX | lnb: the console does not complain about not finding your custom sound file for you-are-punched-in |
| 17:45.42 | lnb | no |
| 17:45.47 | lnb | i wish it did |
| 17:45.54 | lnb | and the format it correct too |
| 17:46.06 | navaismo | lnb, iddindt see the cli output or complete dialplan if you can point to that i can take a look |
| 17:46.21 | navaismo | s/iddindt/I didn't/ |
| 17:46.36 | lnb | ok i will past that in pb |
| 17:46.42 | lnb | paste that is |
| 17:47.28 | XandriX | the b0t can really do that ? |
| 17:47.34 | XandriX | s/b0t/Bot/ |
| 17:47.50 | XandriX | sweet |
| 17:48.10 | lnb | http://pastebin.ca/2492168 |
| 17:51.32 | lnb | you want to see complete dialplan.. one sec i paste it to pb |
| 17:52.02 | navaismo | yep |
| 17:52.34 | navaismo | at line 90 the values of the last two variables are ok? auth & stat |
| 17:52.39 | lnb | http://pastebin.ca/2492169 |
| 17:53.44 | navaismo | In your AGI i think you need to decalre auth as ${auth} & stat as ${stat} |
| 17:54.12 | lnb | ok |
| 17:55.01 | lnb | navaismo: you talking about lines like: |
| 17:55.01 | navaismo | in your first pastebin since stat = 0 it goes to unmatched option then the goodbye you only evaluate stat against 1 or 2 |
| 17:55.13 | lnb | elseif ($status==1) { |
| 17:55.27 | lnb | no you wrote 'stat' |
| 17:55.36 | navaismo | <PROTECTED> |
| 17:55.55 | lnb | looking at wrong file.. |
| 17:56.18 | lnb | $agi->set_variable($user_stat,$stat); |
| 17:57.21 | navaismo | in this pastebin http://pastebin.ca/2492169 at line 13 the last two words called auth & stats are variables? IF so put it with the ${} |
| 17:57.44 | navaismo | like: exten => *2000,n,Agi(timeclockinit.php,${userid},${userpin},${auth},${stat}) |
| 17:58.20 | navaismo | anyway in the cli output your stat=0 at line 99 of the PB-->-- Executing [*2000@from-internal:11] NoOp("SIP/270-00002708", "Stat=..0..") in new stack |
| 17:59.03 | navaismo | then jumps to 'good' label and didnt match any gotoif so it jumps to 'done' label |
| 17:59.14 | navaismo | and finist playing the goodbye sound |
| 17:59.23 | navaismo | s/finist/finish/ |
| 18:05.14 | *** join/#asterisk HumpyDumpy (~eXcAliBuR@206.162.174.6) |
| 18:05.19 | HumpyDumpy | i'm back |
| 18:07.01 | BeachBall | did anyone miss me? |
| 18:07.04 | *** join/#asterisk fullstop (~fullstop@64-121-16-14.c3-0.tlg-ubr1.atw-tlg.pa.cable.rcn.com) |
| 18:07.28 | navaismo | nein |
| 18:07.42 | BeachBall | i'm gonna take that as a yes :} |
| 18:08.02 | *** join/#asterisk mjordan (~matt@50-192-47-77-static.hfc.comcastbusiness.net) |
| 18:08.02 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:08.15 | [TK]D-Fender | BeachBall: With every bullet so far... |
| 18:09.00 | navaismo | we miss the old nickname about the mighty sword |
| 18:09.17 | *** join/#asterisk Ibrahim22 (~Ibrahim21@ip51ced6a4.adsl-surfen.hetnet.nl) |
| 18:10.20 | Ibrahim22 | Hi, is it possible to install asterisk 12 beta, just to get a feel of the new system or is it a bad idea? And also, where do I find on the issue tracker, open issues for asterisk 12? |
| 18:10.35 | *** join/#asterisk boratynskikamil (5311a3f3@gateway/web/freenode/ip.83.17.163.243) |
| 18:10.54 | navaismo | yes its possible, and in the JIRA page of asterisk you can find the opened issues |
| 18:11.02 | [TK]D-Fender | Ibrahim22: On the tracker... it IS a list of the issues, and it isn't in RC let alone "released" expect plenty of instability |
| 18:12.10 | boratynskikamil | Good evening. My today's question is. I registered one [Asterisk1] as SIP client to another[Asterisk2] via register => line, and when I phone to Asterisk2 I got an error message: Failed to authenticated device [phonenumber] etc. Where should I look for the reason? |
| 18:12.10 | Ibrahim22 | Yeah, I installed it today, and found it difficult to get it running, but was wondering if it was a bad idea :) |
| 18:12.25 | [TK]D-Fender | boratynskikamil: SIP DEBUG.... |
| 18:12.51 | [TK]D-Fender | Ibrahim22: "difficult to get it running" could be the result of anything. |
| 18:13.47 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 18:14.46 | lnb | navaismo: so you |
| 18:15.02 | lnb | navaismo: you are saying make line 13 to be: exten => *2000,n,Agi(timeclockinit.php,${userid},${userpin},${auth},${stat}) |
| 18:15.54 | Ibrahim22 | [TK]D-Fender: the difficulty was my endpoints sometimes registering, sometimes not, channels dropping like flies for no reasons and a bunch of other issues. That's why I wanted to see where the open issues for asterisk 12 was. I tried looking in JIRA, but it was overwhelmingly complicated :) |
| 18:16.48 | [TK]D-Fender | Ibrahim22: Three are multiple SIP channel driver in that version, and you've got no debug in hand to show... so we really can't say... |
| 18:17.01 | [TK]D-Fender | Ibrahim22: If you just want to start learning * I recommend the 11 branch |
| 18:17.10 | *** join/#asterisk apb1963 (~quassel@174.134.232.228) |
| 18:17.27 | lnb | navaismo: did not help. same thing after entering correct user/pin, only hear 'goodbye' |
| 18:18.33 | navaismo | check your cli output if your stat value is different from 1 or 2 it always playback the goodbye sound |
| 18:18.36 | boratynskikamil | Damn. [TK]D-Fender is it possible to hide SIP read log using SIP DEBUG mode? |
| 18:18.58 | Ibrahim22 | [TK]D-Fender: I've been with asterisk since 1.6, I think I'm out of the beginner's group. I've already got my software running on 11. I wanted to get my feet wet with 12 |
| 18:19.02 | [TK]D-Fender | boratynskikamil: No. |
| 18:19.20 | [TK]D-Fender | Ibrahim22: Blood is "wet" too.... |
| 18:19.50 | Ibrahim22 | [TK]D-Fender: haha, are you running on 12? |
| 18:20.06 | [TK]D-Fender | Ibrahim22: Nope... I like having a working PBX... |
| 18:20.38 | Ibrahim22 | [TK]D-Fender: hahaha, i'm crazy like that, I like broken stuff, gives me stuff to fix |
| 18:21.06 | lnb | navaismo: http://pastebin.ca/2492174 |
| 18:21.07 | WIMPy | You should try Windows. |
| 18:21.42 | lnb | navaismo: is the ast_carefulwrite error from the script? |
| 18:22.39 | lnb | says success, then goto 15 then 16 but it doesnt say anything |
| 18:22.50 | dar123 | hey guyz, i am stuck since a month. Can't get incoming calls to work, everything seems normal from configuration prespective. Outgoing works, incoming call never hits asterisk |
| 18:24.15 | dar123 | opened case with digium as well, engineer asking to check with service provider :( |
| 18:24.24 | navaismo | lnb, in the last pb the value of STAT is equal to 3 |
| 18:24.37 | navaismo | lnb, again you dont have an option for value 3 |
| 18:24.42 | [TK]D-Fender | dar123: If you register to get incoming calls, look at that. For everything else... debug the channel, dump firewalls, etc |
| 18:24.46 | lnb | i never typed a '3' |
| 18:25.01 | navaismo | lnb, you only play a sound file if stat is 1 or 2 |
| 18:25.01 | lnb | there is no 3 in the file |
| 18:25.08 | lnb | right |
| 18:25.16 | lnb | where is it getting 3 from ? |
| 18:25.35 | navaismo | i guess from your agi |
| 18:26.01 | lnb | i see in the extensions_custom.conf 3rd line: exten => *2000,n,Set(stat=3) |
| 18:26.24 | lnb | should that be like that? |
| 18:26.39 | lnb | should line 3 have stat=3 ? |
| 18:27.16 | dar123 | inbound calls are from PSTN via FXO, will have nothing to do with the firewall i guess |
| 18:27.30 | navaismo | lnb, have you checked in your agi id stat take another value? |
| 18:27.49 | WIMPy | dar123: Not if they come in via a PCI card. |
| 18:27.58 | lnb | i will pb it navaismo |
| 18:28.04 | [TK]D-Fender | dar123: Do you see DAHDI register a change in the line at CLI? |
| 18:28.42 | dar123 | yups, its in service |
| 18:28.56 | lnb | navaismo: http://pastebin.ca/2492176 |
| 18:29.04 | [TK]D-Fender | dar123: No.. when a call is supposed to be ringing against it.. do you SEE any activity in CLI for it? |
| 18:29.49 | *** part/#asterisk Ibrahim22 (~Ibrahim21@ip51ced6a4.adsl-surfen.hetnet.nl) |
| 18:31.46 | *** join/#asterisk fprior (b33c60c8@gateway/web/freenode/ip.179.60.96.200) |
| 18:32.34 | dar123 | nope, no activity |
| 18:32.49 | [TK]D-Fender | darmax-out your core debug. Still nothign? |
| 18:34.20 | navaismo | lnb, yeah you are rewriting the value of stat from the agi file |
| 18:34.28 | saliak | is there a way to test my no-audio issue for a remote SIP extension (presumably an RTP issue) w/out having to call it each time (and having someone sit there, pick it up, say hello, etc.)? |
| 18:34.38 | navaismo | lnb, How do you set it and based on what? I dont know |
| 18:35.32 | navaismo | ok im wrong you are setting --->$user_stat |
| 18:36.54 | dar123 | the only activity i see is when i use "dahdi_monitor 3 -vv" |
| 18:37.04 | navaismo | lnb hmmm maybe thats why you was passing the auth & stat as string--> exten => *2000,n,Agi(timeclockinit.php,${userid},${userpin},auth,stat) |
| 18:37.46 | navaismo | lnb rollback the line exten => *2000,n,Agi(timeclockinit.php,${userid},${userpin},auth,stat) |
| 18:38.14 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 18:38.23 | navaismo | and finally stat variable will be modified by your AGI |
| 18:40.14 | lnb | ok did that |
| 18:40.27 | lnb | now does same... enter user/pin .. -> goodbye |
| 18:40.45 | navaismo | lnb you need to contact the person who writes your dialplan and the AGI. |
| 18:41.08 | navaismo | lnb, the value of stat is set in this line-->$stat=punch_clock($user_id); |
| 18:41.42 | navaismo | then you return the value to asterisk in this line--->$agi->set_variable($user_stat,$stat); |
| 18:43.39 | navaismo | lnb, so this is not an asterisk or diaplan issue, your dialplan is doing what you are telling to do. the issue here is that your agi never set your variable to 1 or 2 |
| 18:44.30 | lnb | agreed |
| 18:44.41 | lnb | i see the 3 in cli |
| 18:44.48 | lnb | and it s/b 1 or 2 |
| 18:44.56 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.59) |
| 18:45.22 | navaismo | maybe you need to start to debug your agi and see what does the function punch_clock |
| 18:47.24 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 18:51.35 | fprior | Hi all, I've a serious fax issue with this scenario PRI <--> Gigium Gateway G100 <--> Adtran. I'm loosing a customer. Here this topic is OT but if someone can redirect me to another resource I'll be grateful |
| 18:54.56 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.99) |
| 18:59.17 | navaismo | Hi fprior, what is the issue? and what module are you suing to receive the fax? |
| 19:06.39 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:11.48 | dar123 | wots the command to debug channel |
| 19:12.14 | [TK]D-Fender | dar123: All you have for your case is core debug |
| 19:13.00 | dar123 | i tried core set debug 10 |
| 19:13.13 | dar123 | and channel debug too nothing shows up |
| 19:19.27 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:20.31 | [TK]D-Fender | dar123: you've got a DAHDI issue then. make sure it's even loaded, test outbound. If that all passes... well.. where are you located again? |
| 19:21.02 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 19:21.10 | BeachBall | i thinks i'm missing something for the subscribe part |
| 19:21.25 | BeachBall | what file is that stuff in? |
| 19:21.40 | [TK]D-Fender | BeachBall: Too vague, try again. |
| 19:22.06 | BeachBall | ok when a person is on the phone, the light beside there name on the other phones dont light up to show status of that person |
| 19:22.24 | BeachBall | i don't have my book with me |
| 19:22.25 | BeachBall | :{ |
| 19:22.40 | [TK]D-Fender | extensions.conf for hints, sip.conf for call-counter, susbscribecontext, etc |
| 19:22.49 | [TK]D-Fender | it's ONLINE. There goes that excuse... |
| 19:22.50 | BeachBall | hints |
| 19:22.53 | BeachBall | thats what it's called |
| 19:24.29 | dar123 | i am located in KSA (Saudi Arabia), i tested outgoing it works |
| 19:25.10 | [TK]D-Fender | dar123: Good odds DAHDI isn't set to detect the signalling your telco providers to indicate an incoming call. |
| 19:25.22 | [TK]D-Fender | dar123: You bought your card new, right? |
| 19:25.31 | dar123 | yups, it's a new card |
| 19:25.45 | [TK]D-Fender | dar123: Go call up the manufacturer for support with it then... |
| 19:26.02 | dar123 | i opened a case with digium they are asking me to check with my telco |
| 19:26.14 | dar123 | i wont get any info from the telco, know that for sure |
| 19:27.03 | lvlinux | Hey guys what's the best way to do 3 way calling?---i.e. caller calls in, agent answers, another agent joins in. |
| 19:27.31 | lvlinux | btw didn't mean to interrupt... |
| 19:27.51 | navaismo | lvlinux, you mean like chanspy? |
| 19:28.34 | BeachBall | i have a confession to make :< |
| 19:28.37 | BeachBall | I don't know what i'm doing |
| 19:28.48 | lvlinux | navaismo: I'm not familiar with chanspy, but I need both agents to be able to communicate with the caller |
| 19:28.53 | XandriX | BeachBall: thats ok i have no idea what im doing either :P |
| 19:29.32 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 19:29.32 | XandriX | but the people in this channel are kind enough to laugh there assess off only behind the screen and help you out :P |
| 19:30.09 | XandriX | im sure Penguin and [TK]D-Fender probably told them selves wow this XandriX is le retarded :P |
| 19:30.25 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 19:30.41 | navaismo | lvlinux, the asy way i the agent1 create a conference from his phone LOL |
| 19:32.02 | lvlinux | you mean using the phone's built-in conference---some of the phones don't have that---what's the "right" way to do it? :-) |
| 19:32.44 | [TK]D-Fender | lvlinux: And what kind of phones are those? |
| 19:33.05 | lvlinux | Cisco 7940 |
| 19:33.23 | [TK]D-Fender | those have 3-way conferencing... |
| 19:33.26 | lvlinux | He also has 8961s - they have conference feature but need it to work regardless of the phone |
| 19:33.42 | lvlinux | oh they do? I didn't know that hmmmm... |
| 19:34.07 | [TK]D-Fender | I have never seen any SIP hard-phone without 3-way calling capabilities |
| 19:34.07 | XandriX | did not know that aswell and i am using a 7940 at home |
| 19:34.11 | XandriX | writes that down |
| 19:34.35 | lvlinux | will that work even with the caller coming in on PSTN? |
| 19:34.46 | [TK]D-Fender | lvlinux: Irrelevant. |
| 19:34.51 | [TK]D-Fender | the phone is SIP |
| 19:34.57 | [TK]D-Fender | it knows nothing about "PSTN" |
| 19:35.01 | [TK]D-Fender | ASTERISK is calling the phone. |
| 19:35.11 | [TK]D-Fender | and that is just a call |
| 19:35.33 | lvlinux | yes sorry that's right lol---so the conf features on the phones just make two RTP sessions and bridge them within the phone? |
| 19:35.52 | lvlinux | Or do I need to setup asterisk to handle the conference someway? |
| 19:36.38 | [TK]D-Fender | all on the phone |
| 19:37.07 | *** join/#asterisk mmikeym (~mikeym@184.70.65.118) |
| 19:38.07 | lvlinux | ok well that simplifies things for me. thanks! |
| 19:42.02 | Penguin | But you CAN set up a conference on asterisk, and then blind transfer several calls from your phone to the conf. |
| 19:43.01 | *** join/#asterisk u0m3_ (~u0m3@92.80.74.24) |
| 19:44.19 | Penguin | ConfBridge comes to mind. |
| 19:44.33 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.90) |
| 19:48.10 | lvlinux | Penguin: is ConfBridge the same thing as meetme? |
| 19:49.48 | [TK]D-Fender | lvlinux: same kind of thing.... superior in * 11 |
| 19:50.38 | Penguin | Similar concept, different implementation. |
| 19:50.50 | lvlinux | which one is superior? |
| 19:51.11 | lvlinux | Penguin: so they are different functions? |
| 19:51.20 | Penguin | In 1.8, ConfBridge is very basic. If you use 11, it's quite robust. |
| 19:51.33 | Penguin | They both serve the purpose of conferencing. |
| 19:56.00 | *** join/#asterisk f0ner00t (~jvandyke@69-170-21-20.static-ip.telepacific.net) |
| 19:56.35 | f0ner00t | Hello is anybody getting this error when compiling asterisk 11.6.0 .pjlib-i686-pc-linux-gnu.depend:1021: *** missing separator. Stop. |
| 20:03.49 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:03.50 | XandriX | Penguin: got a scenario question for ya im thinking it is doable but can i setup asterisk so that when i call my sip number from any phone i can have a prompt where ill force my sip phone to make a conference call between it me and another number ? |
| 20:04.29 | XandriX | i did not phrase that properly i think |
| 20:04.53 | XandriX | i call my sip number from any phone i get a prompt i press say option 3 it asks me for the phone number i type in the phone number and it makes a conference call |
| 20:05.24 | Penguin | Of course. See dialplan. |
| 20:05.33 | Penguin | That's all dialplan. |
| 20:05.43 | XandriX | cools |
| 20:06.24 | Penguin | It sounds like you want an auto attendant. |
| 20:06.30 | XandriX | i still haveto figure out how to get festival running properly on my setup one day i will have an extra rpi and ill use that as my asterisk box with 11.6 |
| 20:06.45 | f0ner00t | Hello Xandrxi and Penguin. |
| 20:06.46 | XandriX | Penguin: maybe |
| 20:06.46 | Penguin | "For plumbing, press 1. For electrical, press 2. For ..." |
| 20:06.57 | XandriX | well yeah |
| 20:07.10 | Penguin | "If you know the extension of whatever, dial it now." |
| 20:07.16 | Penguin | ~aa |
| 20:07.16 | infobot | AA is Auto Attendant, or a digital receptionist. The Auto Attendant accepts input during BackGround() or WaitExten(), and executes extensions. Do not confuse this with an IVR. |
| 20:07.43 | XandriX | googles |
| 20:08.05 | XandriX | theres that that i want to do fix festival for my current scenario and make it so when someone calls both extensions ring and whoever picks up picks up |
| 20:08.22 | Penguin | Extensions don't ring. |
| 20:08.24 | Penguin | Phones ring. |
| 20:08.26 | f0ner00t | Can you set up an IVR and set an extention to sip/outbound/ and have it dial out depending on what option you pick? |
| 20:08.30 | navaismo | XandriX, many users for home projects use the Google tts API |
| 20:08.31 | XandriX | *err phone |
| 20:08.45 | *** join/#asterisk mirela666 (~mirko.bra@95.180.56.37) |
| 20:09.14 | Penguin | To make several phones ring at the same time, you call them from the same Dial(). E.g., Dial(SIP/jack&SIP/jill,32) |
| 20:09.15 | XandriX | navaismo: ill look into that cuz the build they made for pfsense (only spare machine i currently have to run asterisk on at home) has very little modules and is missing a fiew features |
| 20:09.30 | navaismo | http://zaf.github.io/asterisk-googletts/ |
| 20:09.34 | f0ner00t | Hi Navaismo |
| 20:09.41 | navaismo | hi there f0ner00t |
| 20:10.00 | XandriX | navaismo: thx |
| 20:10.16 | f0ner00t | Long time no talk navaismo. |
| 20:10.18 | XandriX | funny thing is i almost got festival working to i just need to find 1 conf file and change its dictionary to CMU |
| 20:12.23 | Penguin | find / -name festival.conf |
| 20:13.17 | navaismo | yup f0ner00t how are you? |
| 20:13.46 | f0ner00t | I'm good just working hard.. Playing with my asterisk box every day to learn new stuff how about you? |
| 20:16.18 | navaismo | broken, prepraing myself to back to the slavery, 2 months with flu but hey i have Internet! |
| 20:18.10 | XandriX | Penguin: haha thats what i did last time and now i just realized i had a bug with my console that day cuz it would only return me festival.conf and not its containing folder lol |
| 20:18.16 | XandriX | now it does display it |
| 20:20.48 | XandriX | SIOD ERROR: could not open file /usr/local/share/festival/lib/dicts/oald/oaldlex.scm |
| 20:20.52 | XandriX | is what i need to fix hehe |
| 20:23.50 | XandriX | and for some reason all find / -name festival.conf only outputs the festival.conf for asterisk for some reason |
| 20:24.10 | Penguin | "for asterisk"? |
| 20:24.44 | Penguin | Try it again, this time in plain English. |
| 20:31.44 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 20:33.11 | XandriX | [2.1-RELEASE][admin@pf2.x-networks.info]/(19): find / -name festival.conf |
| 20:33.13 | XandriX | /cf/conf/asterisk/festival.conf |
| 20:33.39 | XandriX | in other words not the right one it has nothing in it in regards to dictionaries etc its just is festival running on localhost etc etc |
| 20:33.46 | f0ner00t | navaismo: That flu is awful ... My kids all have it! |
| 20:36.01 | XandriX | keeps searching |
| 20:39.11 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 20:43.26 | *** join/#asterisk staykov (~wiggin@pdpc/supporter/active/staykov) |
| 20:44.07 | staykov | hey, wondering whats the best way to setup two servers to be able to dial extensions on some contexts of each other |
| 20:44.36 | staykov | i tried setting it up with iax2 but had issues, anyone know of a good tutorial or should i just do it with SIP? |
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| 20:44.50 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 20:45.00 | staykov | my issues were authenticating but i probably messed it up |
| 20:45.12 | XandriX | victory festival now starts and doesnt give that error |
| 20:46.26 | *** join/#asterisk CeBe (~CeBe@port-92-206-106-63.dynamic.qsc.de) |
| 20:46.56 | Penguin | staykov: The tech between the two is not important. |
| 20:47.21 | Penguin | Contexts and extensions are just as available over IAX2 as they are over SIP. |
| 20:48.33 | [TK]D-Fender | Sort of... |
| 20:48.55 | [TK]D-Fender | IAX2 can technically target remote contexts in their dial. |
| 20:49.13 | [TK]D-Fender | Something SIP can't... but I'd never let a peer have free riegn like that anyway |
| 20:49.54 | Penguin | The inverse of what I said could be considered inaccurate. |
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| 21:29.41 | pigpen | hi all, I am trying to process an all page to 35 phones. Each phone call is 17 char (SIP/ACME_001_1234&). |
| 21:29.54 | pigpen | It is stripping off the command at 120 char. |
| 21:30.00 | pigpen | all else is ignored. |
| 21:30.11 | pigpen | ideas? Pulling list from realtime. |
| 21:30.35 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 21:33.39 | pigpen | maybe limit on realtime values? |
| 21:36.27 | pabelanger | pigpen: *CLI> core show function dialgroup |
| 21:37.16 | pabelanger | actually, not sure if that works with paging |
| 21:37.21 | pabelanger | but give it a try |
| 21:39.18 | pigpen | yeah, it might "fake" out the command length. |
| 21:40.04 | pigpen | not a bad idea. I have done this before by making several variables, then calling all the variables at the same time. |
| 21:40.14 | pigpen | 1 variable by itself didn't work. |
| 21:40.32 | pigpen | in the past we had to modify the C code before compiling. |
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| 21:41.41 | pigpen | But now, if "LOW_MEMORY" is not being defined at compile time, it should default it to "8192", otherwise, it is at 256 |
| 21:41.58 | jonno11 | Hey guys. Getting a segmentation fault after starting asterisk on precise x64... Built from source. http://pastebin.com/GeK79she |
| 21:42.02 | pigpen | but, when it got to about 3000 char, asterisk would puke. |
| 21:43.04 | Chainsaw | jonno11: "precise x64"? What's wrong with regular AMD64? |
| 21:43.29 | Penguin | I've never even heard of "precise x64." Sounds made up. |
| 21:44.01 | Chainsaw | nods |
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| 21:49.20 | pabelanger | ~backtrace |
| 21:49.20 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 21:49.32 | pabelanger | jonno11: ^ do that |
| 21:53.07 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 21:56.54 | jonno11 | pabelanger: "/usr/sbin/core: No such file or directory." Where would the core file be? |
| 21:57.08 | jonno11 | wait |
| 21:57.10 | jonno11 | ignore me |
| 21:57.33 | eirirs | ** ignored jonno11 ** |
| 21:57.36 | eirirs | wait |
| 21:58.19 | jonno11 | Nope, still wrong. How do I get a core file? |
| 21:58.50 | pabelanger | jonno11: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-PreparingAsteriskToProduceCoreFilesOnCrash |
| 21:58.57 | pabelanger | It is all in the wiki page |
| 21:59.08 | jonno11 | pabelanger: ah ok. So the core file is a crash file? |
| 22:00.02 | pabelanger | yes |
| 22:00.11 | pabelanger | if asterisk crashed, and you have -g it will create the core file |
| 22:00.20 | pabelanger | you need to check where your OS defaults it to |
| 22:00.22 | jonno11 | pabelanger: Hmm I just tried that. |
| 22:00.29 | pabelanger | I think /var/crash in ubuntu |
| 22:00.32 | jonno11 | Ah so it's specific to OS? |
| 22:01.02 | WIMPy | You can een configure the name. |
| 22:01.06 | WIMPy | even |
| 22:08.17 | pabelanger | sysctl controls the coredump location |
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| 22:09.43 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 22:10.37 | jonno11 | pabelanger, WIMPy: http://pastebin.com/j3Whg28a |
| 22:11.00 | jonno11 | that's the entire file... |
| 22:12.44 | pabelanger | jonno11: we need a full bt |
| 22:13.39 | jonno11 | pabelanger: How can I get that? |
| 22:14.22 | dan_j | Hi. I have a server with multiple calls running constantly. I need to pull off a sip debug for one leg of a call to diagnose a CLI issue. The CLI is set by asterisk and sent to the provider, but the calls are coming up as from UNKNOWN. |
| 22:14.45 | dan_j | Is it possible to get a sip debug of one channel, without ending up dumping all sip debug data? |
| 22:15.21 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:16.27 | jonno11 | pabelanger: I've followed the steps above - and that's the entire of the backtrace.txt file |
| 22:18.27 | navaismo | dan_j, try with sip set debug ip <ip> but IMO is better to pull the complete debug. You can also use ngrep/sngrep and wireshark |
| 22:19.38 | dan_j | navaismo: thanks. I forgot about debug ip. But is there any way to single out one call? For example, we could have multiple calls to the same ip at any time. |
| 22:21.44 | navaismo | all the pbx in the world have many calls at the same time. You have the full log to get traces and as I said you have tools like ngrep/sngrep or wireshark to retrieve specific parts of the debug. |
| 22:22.13 | navaismo | specially wireshark is very useful with the voip traces |
| 22:22.32 | dan_j | Ok. I didnt realise i could you wireshark to strip out specific calls. thanks. i'll give it a try. |
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| 22:27.34 | navaismo | dan_j, http://wiki.wireshark.org/VoIP_calls |
| 22:28.41 | dan_j | Amazing. Thank you. |
| 22:31.22 | nextime | ok, webrtc on asterisk isn't yet enough functional, sadly. |
| 22:31.52 | navaismo | still... |
| 22:33.03 | nextime | navaismo : except for ICE issues, there other limitations: for example no video support as no vp8 codec, chrome/chromium works, but firefox doesn't apparently.... and so on |
| 22:33.50 | navaismo | you can patch your asterisk to support vp8, firefox need certificates |
| 22:33.52 | navaismo | etc tec |
| 22:34.04 | navaismo | all this is implementation stuff |
| 22:34.26 | nextime | navaismo : the patch for asterisk in vp8 doesn't do translation as i know |
| 22:34.33 | navaismo | and keep in mind that webrtc still in development |
| 22:34.51 | nextime | so, no way to video call from vp8 (Chrome) to h263+ (grandstream hw phone) for example |
| 22:35.35 | [TK]D-Fender | There is no video transcoding of any kind |
| 22:35.42 | [TK]D-Fender | This isn't a vp8 thing |
| 22:35.43 | nextime | exactly |
| 22:35.56 | nextime | and this sadly is a limitation when you go on webrtc |
| 22:36.07 | [TK]D-Fender | is it? |
| 22:36.15 | [TK]D-Fender | Who else out there is transcoding in realtime? |
| 22:36.15 | nextime | yes, it is, at least for me |
| 22:36.17 | *** join/#asterisk igustin (~igustin@alpha.linux.hr) |
| 22:36.21 | navaismo | well webrtc isnt developed to use with a PBX |
| 22:36.28 | navaismo | is most to use peer 2 peer |
| 22:36.38 | nextime | [TK]D-Fender webrtc2sip seems to do that |
| 22:36.58 | navaismo | media gateway != PBX |
| 22:37.33 | igustin | trying to originate call via AMI, but getting "Response: Error, Message: Originate failed", any hints? |
| 22:37.40 | nextime | well, i need to put my video-citophone ( a sip based one, registered on * ) to work, this is my goal |
| 22:37.54 | nextime | so, i really need video translation |
| 22:38.24 | navaismo | not with asterisk, or you use the same codecs or put a media gateway in front |
| 22:38.26 | [TK]D-Fender | nextime: Certainly not happening any time soon... |
| 22:39.06 | [TK]D-Fender | igustin: You should see an Originate message concerning retry failure, etc... |
| 22:39.06 | nextime | [TK]D-Fender : i understand that and i'm not asking asterisk to do things i need but apparently majority don't need |
| 22:39.23 | [TK]D-Fender | igustin: it would help to actually see what you're doing & getting... |
| 22:39.25 | nextime | i'm just saying "isn't eough featured for my own needs" |
| 22:39.47 | nextime | so, i need to find something to put in front of it for the webrtc part |
| 22:40.11 | igustin | [TK]D-Fender: thx, trying via PHPAGI script and via telnet, same result |
| 22:40.29 | navaismo | i dont want to be an ass but that is very different than: "ok, webrtc on asterisk isn't yet enough functional, sadly" |
| 22:41.36 | navaismo | but you have the media gateway: webrtc2sip and you can get hekp from their discussion board |
| 22:41.48 | nextime | navaismo : well, it is anyway even ice doesn't fully work apparently |
| 22:42.26 | navaismo | seems like many people is using it not only with asterisk, I saw threads with freeswitch, 3cx etc |
| 22:42.26 | nextime | navaismo : wevrtc2sip is bad coded and needs a very specific version of ffmpeg, i don't really like it |
| 22:42.32 | igustin | [TK]D-Fender: I have 2 softphones, originate via CLI works, but I can't get working originate via telent norIGI |
| 22:42.54 | [TK]D-Fender | igustin: we need to see the actual requests, and AMI output... |
| 22:43.02 | navaismo | nextime, hmmm then start to coding something that fits your needs |
| 22:43.39 | nextime | navaismo : this is what i was trying to avoid as my todolist is already very long, but sadly it seems that this will be the only way to go |
| 22:44.01 | igustin | GGI |
| 22:44.14 | navaismo | yep, when we cant find a perfect solution to our problems in the opensource world the next step is create it |
| 22:44.31 | navaismo | or patching |
| 22:44.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.70) |
| 22:44.51 | nextime | navaismo : i agree with you in that, but before to start to code i was investigating the other options :) |
| 22:45.17 | igustin | [TK]D-Fender: originate with the same parameters works via CLI, but not via AMI telnet and PHPAGI |
| 22:46.09 | igustin | [TK]D-Fender: via telnet I first login, then type Action: Originate, Channel: SIP/400, Exten: 200, Priority: 1, Context: some |
| 22:46.11 | *** join/#asterisk lnb (~lnb@CPE000347b24a71-CM602ad06bec2f.cpe.net.cable.rogers.com) |
| 22:46.37 | [TK]D-Fender | igustin: We need actual debug and details. |
| 22:46.54 | igustin | [TK]D-Fender: and get Response: Error Message: Originate failed |
| 22:47.08 | [TK]D-Fender | igustin: Not just that one statement. |
| 22:47.49 | navaismo | nextime what about the telepresence system? |
| 22:48.29 | igustin | [TK]D-Fender: I set 'manager set debug on' but all I get is: "Manager 'ami' logged on from 127.0.0.1" and logoff |
| 22:49.22 | [TK]D-Fender | igustin: tcpdump the port and provide real backup |
| 22:49.24 | igustin | [TK]D-Fender: on telnet screen I get: http://dpaste.com/1493103/ |
| 22:49.35 | nextime | navaismo : isn't just a component of the doubango/webrtc2sip packages? |
| 22:50.41 | navaismo | this is doubango/telepresence most for conference like hangouts but hmm yes it use the same old version of ffmpeg |
| 22:51.38 | igustin | [TK]D-Fender: this is my PHPAGI code: |
| 22:51.42 | nextime | ok, i really need to choose if to start to code from scratch or to patch doubango code |
| 22:51.44 | igustin | [TK]D-Fender: this is my PHPAGI code: http://pastebin.com/pNWK2FaJ |
| 22:52.18 | igustin | [TK]D-Fender: I'll take tcpdump output for a moment |
| 22:52.28 | [TK]D-Fender | igustin: "sop show peer 400" <- |
| 22:52.34 | [TK]D-Fender | igustin: "sip show peer 400" <- |
| 22:52.36 | igustin | [TK]D-Fender: thx for trying to help |
| 22:53.04 | igustin | both sip softphone are registered, can make call between them, can orignate call via CLI |
| 22:54.11 | igustin | [TK]D-Fender: 400 is extension number, should I use registered name? :S |
| 22:54.16 | *** join/#asterisk bkruse (~Adium@64.89.97.127) |
| 22:54.22 | [TK]D-Fender | igustin: I see I am never going to get the pieces I ask for but instead some poor substitute every time. Perhaps someone else will feel like carrying this forward, |
| 22:54.56 | *** join/#asterisk barbosa2 (~juliano.b@177.159.131.227) |
| 22:55.04 | navaismo | nextime, http://www.slideshare.net/MoisesSilva6/implementation-lessons-using-webrtc |
| 22:55.29 | lnb | if an office is going to have 10 sip phones, do the sip ports have to be changed i.e. 5060 -> 5060-5070 ? |
| 22:55.53 | igustin | [TK]D-Fender: sip show peer -> http://pastebin.com/EwxZxDQW |
| 22:56.05 | lnb | also assuming the pbx is remote |
| 22:56.09 | [TK]D-Fender | 'Channel' => 'SIP/400' <--- this is very clearly a SIP DEVICE being dialed |
| 22:56.24 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:56.29 | [TK]D-Fender | igustin: does SIP/400 look like SIP/LinphoneWin to you? |
| 22:56.40 | [TK]D-Fender | igustin: You weren't looking at what you are DIALING |
| 22:56.42 | navaismo | lnb, nope |
| 22:56.45 | igustin | uhm :/ :blush: |
| 22:57.30 | lnb | navaismo: then when do the ports on the sip phone have to be changed from 5060? |
| 22:57.49 | [TK]D-Fender | igustin: Substituting for what now appears to be the proper device, is it starting to work? |
| 22:58.11 | igustin | [TK]D-Fender: Geee, that was the problem! Shame on me. :-( THANK YOU!!! |
| 22:58.35 | lnb | navaismo: for example here in my small office, my gateway nat is set to same-ports. So if I set a sip phone up to use 5065, on the remote pbx sip show peers -> ... 5065 |
| 22:58.35 | [TK]D-Fender | igustin: Glad it's resolved. |
| 22:59.20 | lnb | whereas if same-ports is removed from pf nat config, then sip show peers will show something like 55348 |
| 23:00.16 | igustin | [TK]D-Fender: works via telnet, but still don't work via PHPAGI script, I'll take look at manager.conf privileges |
| 23:02.35 | navaismo | lnb i dont follow/understand the question |
| 23:03.37 | lnb | navaismo: I think you mentioned before you have large client(s). Just wondering in a place where there is 10, 20 or more sip phones, how you would manage the ports |
| 23:03.53 | lnb | you can change the ports for ext. in gui, and in sip phone |
| 23:03.57 | lnb | there must be a reason |
| 23:03.58 | navaismo | i dont manage the ports |
| 23:04.01 | lnb | really |
| 23:04.02 | lnb | hmm |
| 23:04.24 | navaismo | asterisk listen on 5060 i sent the rerquest to 5060 thats all |
| 23:05.38 | *** join/#asterisk theron (~theron@mpk-nat-1.thefacebook.com) |
| 23:06.12 | XandriX | aaah finaly back home |
| 23:06.36 | lnb | navaismo: are you talking about local PBX or remote PBX ? |
| 23:06.52 | navaismo | lnb, I only change the port in the SIP client when im using two or more accounts like in the yealink phones account1 use 5060 account 2 use 5061 for obvious reasons, or qhen my softphone is in the same machine as Asterisk |
| 23:07.51 | navaismo | lnb, in both situations I leave asterisk listening on port 5060 |
| 23:08.20 | igustin | [TK]D-Fender: works even via PHPAGI now (timeout was too short), thanks for helping |
| 23:08.31 | lnb | what is the obvious reason for 5061? |
| 23:08.50 | [TK]D-Fender | igustin: glad to hear... |
| 23:09.21 | navaismo | lnb, the phone already is using the port 5060 |
| 23:10.31 | navaismo | If I recall the pap2t have the same behavior for the accounts |
| 23:10.33 | navaismo | brb |
| 23:11.29 | lnb | navaismo: here is some connections from office to remote PBX: D N A 5064 OK (62 ms), A 28264 OK (46 ms), A 5061 OK (39 ms) |
| 23:11.43 | lnb | there are about 40 connections |
| 23:12.02 | lnb | some are 50xx and others are 28263 and higher |
| 23:12.33 | XandriX | Penguin: festival even with cmu does indeed sound horrible :P |
| 23:12.59 | navaismo | yep i guess the non 50XX ports are for remote phones |
| 23:13.00 | lnb | navaismo: in your scenario is the PBX local to the phones or is the PBX remote somewhere? |
| 23:13.05 | navaismo | but anyway BRB |
| 23:13.21 | lnb | navaismo: all the phones are remote to the PBX |
| 23:13.26 | lnb | in my case |
| 23:14.32 | lnb | i ask becasue they have some 3-4 year old linksys phones. they complain to me (i did not provide them) that some of the linksys phones after dialing, the lines (4 line phones) will turn red |
| 23:14.37 | lnb | and they can't use them |
| 23:15.00 | lnb | pull power, wait a few minutes, plug back in -> green. untill the next outbound call |
| 23:15.26 | XandriX | erm can festival not read a sentence that has a commma in it past said comma ? |
| 23:15.32 | lnb | so i am wondering if this has anything to do with the ports (i wouldn't think so) but i ask anyway |
| 23:15.50 | wdoekes | lnb: linksys phones behind nat generally work fine with: via-send-rport, nat-mapping=yes, nat-ping=yes |
| 23:16.14 | lnb | wdoekes: not sure if they have the via-send-rport |
| 23:16.28 | lnb | but i can check after i finish something here |
| 23:16.58 | lnb | i brought them about 15 yealink phones and none do this behavior |
| 23:17.09 | lnb | but i do have rport enabled |
| 23:17.12 | lnb | on the yealinks |
| 23:18.28 | wdoekes | did you reset the linksyses to factory default? or do they have other strange settings perhaps? |
| 23:19.08 | lnb | never reset them |
| 23:19.18 | lnb | its a busy office |
| 23:19.28 | lnb | stupid answer |
| 23:19.54 | lnb | so you think best to reset the phone and configure it? |
| 23:20.53 | wdoekes | if you don't, who knows what settings might be messed up.. reboot on dial is not something I have an explanation for |
| 23:21.56 | wdoekes | alternatively, you could setup a syslog server and look at that |
| 23:22.23 | wdoekes | but that might not yield anything useful |
| 23:23.56 | lnb | wdoekes: doesn reboot |
| 23:24.08 | lnb | lines just turn reddish amber |
| 23:24.26 | lnb | and last week, two of the phones lost their configuration completely |
| 23:26.28 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 23:26.28 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 23:31.20 | navaismo | back |
| 23:31.35 | navaismo | lnb also check if they have the nat-keep-alive option |
| 23:31.56 | lnb | they do |
| 23:32.23 | lnb | i changed all the phones (each voip account) to use IP address instead of fqdn |
| 23:32.55 | lnb | now at least they don't drop off line (no talking about the red/amber) |
| 23:37.27 | *** join/#asterisk jonno11 (~jonno11@cpc1-walt12-2-0-cust582.13-2.cable.virginm.net) |
| 23:44.20 | XandriX | wonder if i can slap mbrola on to my festival install it apparently sounds way better than any festvox package |
| 23:44.32 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.87) |
| 23:45.46 | navaismo | festival, swift and some cepstarl voices will sound like robots, the best TTS are expensive and the one free is google api but if you use it you are breaking the TOS |
| 23:49.25 | *** join/#asterisk toresbe (~toresbe@158.36.190.254) |
| 23:49.46 | toresbe | Hey guys, I think I may have found behaviour that looks an awful lot like a bug |
| 23:50.00 | toresbe | [2013-12-05 00:48:36] WARNING[15493]: frame.c:1159 ast_codec_get_len: Unable to calculate sample length for format slin44 |
| 23:50.33 | XandriX | navaismo: yeah the default voice i currently have sounds like a robot indeed ive been reading post that say that mbrola us is way better |
| 23:50.34 | toresbe | If I set up a moh class with format slin16, I will get a very weird and wonderful sound which is hard to describe |
| 23:50.53 | XandriX | plus it speaks to fast to be honest |
| 23:51.05 | *** join/#asterisk jonno11 (~jonno11@cpc1-walt12-2-0-cust582.13-2.cable.virginm.net) |
| 23:51.08 | XandriX | doesnt read past a comma xD |
| 23:51.26 | toresbe | Are developers active in here, btw? |
| 23:58.18 | *** join/#asterisk benklop (~quassel@208.91.2.2) |
| 23:58.24 | benklop | hola |
| 23:58.55 | benklop | i'm having way too much trouble getting my asterisk box to connect to my sip provider |