| 00:05.08 | rfreire | navaismo: thanks for stopping by, man! I'm doing a pretty simple & straightforward local SIP service for home. see logger.conf, cdr.conf and cdr_custom.conf: http://pastebin.com/jECH0t4E |
| 00:10.01 | [TK]D-Fender | rfreire: And then you look in modules.,conf to see if it's even loading.... |
| 00:10.24 | rfreire | [TK]D-Fender: ooo, /me looks |
| 00:11.50 | Penguin | You can also check in asterisk with the "module show" command. |
| 00:12.14 | rfreire | no mentions to log on modules.conf. checking cli module show |
| 00:12.57 | Penguin | Most of the time, if you have autoload set to yes and you have not explicitly prevented the module from loading, if there is a valid conf for the module, it will load. |
| 00:13.13 | rfreire | Penguin: [TK]D-Fender hnm really no module. http://pastebin.com/bJ0rSenK |
| 00:13.24 | [TK]D-Fender | yeah, that's what it's feel like |
| 00:13.24 | rfreire | autoload=no |
| 00:13.26 | rfreire | changes |
| 00:13.36 | [TK]D-Fender | rfreire: restart * and check "logger<tab> |
| 00:13.48 | [TK]D-Fender | rfreire: change to autoload=es |
| 00:13.50 | [TK]D-Fender | yes* |
| 00:14.11 | Penguin | With autoload set to no, you have to explicitly load every single module that you want to use. |
| 00:14.15 | rfreire | oooooooo |
| 00:14.17 | rfreire | here it comes |
| 00:14.20 | Penguin | With it set to yes, all you have to do is supply a valid conf. |
| 00:14.31 | rfreire | [TK]D-Fender++ Penguin++ A-W-E-S-O-M-E |
| 00:14.41 | rfreire | Thanks a LOT guys ;-D |
| 00:15.16 | rfreire | I was just frustrated; wrote crazy dialplans, everything working flawlessly... but couldn't manage to make a stupid log work! ;-D |
| 00:15.35 | rfreire | Thank you very mucho again! \o |
| 00:18.29 | navaismo | see you all |
| 00:22.55 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 00:24.41 | rfreire | [TK]D-Fender Penguin add this one: While /var/log/asterisk had proper access, /etc/asterisk/<some config files> file mode was 600 or 660 and user:Group were root:root, thus preventing the asterisk user to parse the file! /o\ me-- |
| 00:25.21 | Penguin | It isn't the first time that has happened. |
| 00:27.50 | rfreire | Penguin: yep. Upon initial Asterisk setup, I moved the original config fikes to /etc/asterisk/orig folder and then was crafting individually the needed/relevant files. However, when copying /etc/asterisk/orig/<Blah>.conf, it also inherited the permission, and thus, preventing asterisk:asterisk to read it. |
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| 02:03.10 | ChannelZ | I know fail2ban uses gamin (if you have it) but does it notice when another process rotates the log? I had a bunch of bad registrations that it seemed to ignore yet testing the regex against the log, it sees. |
| 02:07.05 | Penguin | I've got one AsteriskNOW box that the fail2ban was giving me trouble with, and I don't understand it either. It apparently watches the log file correctly, because it adds the offending IP addresses to the correct chain in iptables. But I continue to see the same hosts trying to make calls after they should be blocked. And if I monitor iptables, the byte count on those lines do not increase. |
| 02:07.31 | Penguin | I realize it isn't exactly the same as what you said, but what you said made me think of this problem I encountered. |
| 02:09.52 | ChannelZ | hmm |
| 02:12.13 | ChannelZ | my box at work just ignored the same IP.. what the hell |
| 02:13.06 | pensmit | Are WaitExten and Background the only apps that accept DTMF? |
| 02:13.36 | pensmit | They seem to pause to long for me. I'd like them to jump to the extension dialed as soon as one digit is pressed |
| 02:13.42 | pensmit | Hope this makes sense |
| 02:13.46 | ChannelZ | ReadExten |
| 02:13.59 | ChannelZ | That has more to do with your extensions |
| 02:14.27 | ChannelZ | It's waiting for more digits because they might match another extension. |
| 02:15.33 | pensmit | ahh |
| 02:15.39 | pensmit | How do I stop that |
| 02:15.46 | pensmit | say they only need to press one |
| 02:15.54 | ChannelZ | one what? one digit? |
| 02:15.58 | pensmit | yes |
| 02:16.04 | pensmit | 1-9 |
| 02:16.15 | pensmit | so as soon as that digit is pressed |
| 02:16.19 | pensmit | it moves on |
| 02:16.50 | ChannelZ | Your WaitExten or whatever would need to be in a context that only contained those extensions |
| 02:16.54 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 02:18.18 | pensmit | ahh |
| 02:18.21 | pensmit | makes perfect sense |
| 02:18.22 | pensmit | thank you |
| 02:19.40 | ChannelZ | yah |
| 02:38.02 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 02:42.17 | *** join/#asterisk popers (~toxie@189.191.52.77) |
| 02:42.24 | popers | good night |
| 02:42.40 | popers | does anybody know why console command doesnt work in asterisk 11.5 |
| 02:42.52 | ChannelZ | which one |
| 02:42.59 | popers | i put asterisk -rx 'console dial' |
| 02:43.05 | popers | but it dont work |
| 02:43.56 | popers | im trying to dial from the console |
| 02:44.05 | popers | does anybody know how can i do this |
| 02:44.14 | ChannelZ | do you have chan_console loaded? I think that's where that's provided |
| 02:44.38 | ChannelZ | if you're trying to just originate a call through your dialplan, look at 'channel originate' |
| 02:44.50 | popers | ok |
| 02:46.21 | popers | how can i restart modules in asterisk |
| 02:47.35 | ChannelZ | depends. some have reload commands of their own. Otherwise you can 'module unload' and 'module load' them (or 'module reload |
| 02:50.43 | *** join/#asterisk popers (~toxie@189.191.52.77) |
| 02:51.32 | popers | no such command 'console dial' |
| 02:51.41 | popers | i've restarted my system |
| 02:51.46 | Penguin | Stop trying to use console dial, then. |
| 02:51.47 | popers | i've archlinux |
| 02:51.53 | Penguin | Or load the channel driver. |
| 02:52.16 | popers | then how i dial from console |
| 02:52.23 | Penguin | Typing the command over and over again won't make it magically appear. |
| 02:52.31 | Penguin | What do you mean by "dial from console"? |
| 02:52.47 | Penguin | Do you mean initiate a phone call between two end points? |
| 02:53.11 | popers | yes |
| 02:53.21 | Penguin | channel originate |
| 02:53.25 | popers | im following this tutorial |
| 02:53.39 | popers | http://raspimods.blogspot.mx/2012/09/portero-ip-con-asterisk-y-tarjeta-de.html |
| 02:54.05 | popers | it is in spanish but i suppose that the commands are clear |
| 02:54.25 | popers | asterisk -rx 'console dial' |
| 02:54.37 | popers | in a bash script in the raspberry pi |
| 02:55.34 | Penguin | If you are trying to call to one phone and then connect it to another phone, console dial is the wrong command. |
| 02:55.45 | Penguin | channel originate <------ |
| 02:57.32 | popers | its kind of fuzzy for me |
| 02:58.07 | popers | can you say me where can i find a good tuto about asterisk or a book or something |
| 02:58.09 | ChannelZ | the console channel is a very specific thing, a local ALSA channel (your sound card) |
| 02:58.20 | ChannelZ | What is it you're actually trying to do? |
| 02:58.35 | popers | a call between my computer and a rpi |
| 02:58.39 | popers | raspberry pi |
| 02:58.42 | Penguin | If you're trying to use a PHONE, then the console dial command isn't the right command. |
| 02:59.09 | popers | ok correct me if im wrong |
| 02:59.31 | popers | my raspberry pi can handle calls from asterisk whith s extension |
| 02:59.51 | Penguin | Extension s has zero to do with that. |
| 03:07.11 | popers | a book about new versions of asterisk |
| 03:07.26 | popers | because im reading the future of telephony |
| 03:07.58 | popers | and it is a little old |
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| 04:33.02 | pensmit | What's the :3 here? exten => _011.,1,Dial(SIP/${EXTEN:3}@flowroute) |
| 04:33.19 | pensmit | Is this for international calls? |
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| 04:43.38 | j4jackj | pensmit: it's to strip the first 3 off. |
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| 04:59.13 | pensmit | thanks |
| 05:01.10 | Penguin | It's the offset. |
| 05:01.36 | Penguin | And the correct syntax would be Dial(SIP/flowroute/${EXTEN:3}) |
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| 05:27.49 | j4jackj | Hi [TK]D-Fender |
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| 07:08.45 | afidegnum | hello all, anyone online ? |
| 07:10.44 | bulkorok | hi |
| 07:18.28 | ChannelZ | nope |
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| 07:28.37 | riantsoa | Hi guys, i tried everything but i cannot use reliabily DTMF inside asterisk |
| 07:28.44 | riantsoa | here is my config: |
| 07:29.47 | riantsoa | GSMPHONE <-> huawei 3g Dongle <-> chan_dongle <-> Asterisk <-> Diaplan |
| 07:30.41 | riantsoa | The thing is that Freeswitch can handle flawlessly DTMF |
| 07:31.42 | riantsoa | i played with all asterisk parameter but i cannot make it work |
| 07:31.46 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
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| 07:42.53 | riantsoa | i see on debug " DTMF begin '3' received on Dongle/dongle0-0100000007 |
| 07:42.54 | riantsoa | [Sep 15 10:38:45] DTMF[1532] channel.c: DTMF begin ignored '3' on Dongle/dongle0-0100000007" |
| 07:43.14 | riantsoa | why it ignored the digit??? |
| 07:43.32 | riantsoa | he see the begening but it ignore it after |
| 07:43.52 | riantsoa | how can i solve that |
| 07:43.53 | riantsoa | ? |
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| 08:00.05 | riantsoa | huhu |
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| 08:04.51 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 08:06.52 | riantsoa | no body can help? |
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| 08:16.56 | afidegnum | hi all, pls i am in a situtaion |
| 08:17.26 | afidegnum | I am looking for a way to get virtual number for SMS verification... anyone know anything good that can help? |
| 08:17.58 | *** join/#asterisk jhlavacek (~jirka@dhcp192-73.enst.fr) |
| 08:19.10 | ChannelZ | uhmmm.. what? |
| 08:21.28 | afidegnum | any answer ? |
| 08:22.28 | ChannelZ | doesn't understand the question |
| 08:23.46 | wdoekes | afidegnum wants a throwaway number on which he can receive sms |
| 08:23.52 | wdoekes | afaict |
| 08:24.20 | ChannelZ | burner phone... google voice... |
| 08:24.39 | afidegnum | yes |
| 08:24.54 | afidegnum | google voice is not working in my end... I am located in Ghana |
| 08:25.56 | wdoekes | may we ask what verification you're attempting to bypass? |
| 08:26.13 | ChannelZ | Credit card fraud! Are you a prince? |
| 08:26.50 | wdoekes | I'm not sure all africans are 419sters |
| 08:26.59 | afidegnum | nonono |
| 08:27.12 | *** join/#asterisk CeBe (~CeBe@port-92-206-114-40.dynamic.qsc.de) |
| 08:27.19 | afidegnum | I am not into that... the truf is, I am creating various facebook pages to promote my site, |
| 08:27.44 | afidegnum | I am creating users who will also help promote my site |
| 08:28.00 | afidegnum | but I am the one doing all that... |
| 08:29.31 | wdoekes | bloating the facebook database.. that is a noble cause in itself |
| 08:29.51 | afidegnum | I wanted to buy sim cards but we are monitored... against SIM box phone termination... I don't want any unecessary detention over how I don't know how I can explain to them I am creating facebook users for that. |
| 08:30.57 | afidegnum | I hope you understand my situation? |
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| 08:35.03 | TriJetScud | naw still prince |
| 08:37.37 | ChannelZ | So you want to create a bunch of fraudulent accounts so you can verify other fradulent accounts to promote a website that does I-can-only-imagine-what. |
| 08:38.28 | TriJetScud | ^ |
| 08:38.38 | TriJetScud | why don't we just say "we can't help you" |
| 08:39.04 | TriJetScud | because that's just what we can only do |
| 08:39.06 | TriJetScud | we don't do fraud |
| 08:40.20 | ChannelZ | well that and Ghana is a problem. There's a few voip providers in the US that can do SMS but I'm assuming if he can't get ahold of a couple of cell phones that he's not going to get up on something with a credit card |
| 08:40.55 | afidegnum | see... I repeat once again, I am not into fraud... Ghana being blacklisted for fraud doesn't mean all Ghanaians are fraudsters |
| 08:41.29 | TriJetScud | still afidegnum, what you're doing constitutes fraud, creating sock puppets to promote a site |
| 08:41.42 | afidegnum | let me remind you, the Real fraudsters are Nigerians they migrate into other countries to dirty the image of that country |
| 08:42.10 | afidegnum | TriJetScud: how do you promote your site? |
| 08:42.13 | ChannelZ | Not condeming all of you (my original comment was a joke) but you are basically wanting to make a bunch of fake FB accounts |
| 08:42.33 | TriJetScud | afidegnum, I don't. I make good content and people come |
| 08:43.05 | ChannelZ | I presume FB limits you from using the same phone number so you also want a bunch of fake ("virtual") numbers to verify your fake accounts. So you're not on _that_ much of the straight-and-narrow. |
| 08:43.21 | afidegnum | I have a good content... how do you want users to come to your site when you don't promote your site? |
| 08:43.39 | afidegnum | unless you don't know the rudiment of website promotion. |
| 08:43.39 | TriJetScud | this place is #asterisk, not ##seo |
| 08:44.07 | TriJetScud | you have better luck on asking seo related questions at seo places |
| 08:47.35 | ChannelZ | I got a good spam today from "Mr. John Kerry, the secretary of the state" (lower-case theirs) that my "CONTRACT/INHERITANCE Payment/Lotto winners price" was ready to be collected.. I guess he didn't know which. |
| 08:49.42 | ChannelZ | It seemed totally legit to me, that the Head of ATM CARD Department of CITI BANK OF NIGERIA has a gmail address. He's a Reverend too! |
| 08:51.12 | TriJetScud | heh |
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| 08:59.00 | riantsoa | i see on debug " DTMF begin '3' received on Dongle/dongle0-0100000007 |
| 08:59.00 | riantsoa | <riantsoa> [Sep 15 10:38:45] DTMF[1532] channel.c: DTMF begin ignored '3' on Dongle/dongle0-0100000007" |
| 08:59.00 | riantsoa | <riantsoa> why it ignored the digit??? |
| 08:59.00 | riantsoa | <riantsoa> he see the begening but it ignore it after |
| 08:59.00 | riantsoa | <riantsoa> how can i solve that |
| 08:59.01 | riantsoa | <riantsoa> ? |
| 08:59.24 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 09:02.54 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 09:03.14 | nextime | hello all |
| 09:03.27 | nextime | i'm experiencing this very same issue: https://issues.asterisk.org/jira/browse/ASTERISK-17410 |
| 09:03.46 | nextime | using asterisk 1.8 from debian sid |
| 09:04.03 | nextime | i can assume is solved in more recent asterisk versions? |
| 09:04.39 | *** join/#asterisk eject_ck (~Eugene@95.67.72.22) |
| 09:14.10 | kaldemar | nextime: Resolution: Unresolved |
| 09:14.22 | kaldemar | nextime: don't make any assumptions on that being fixed. |
| 09:20.03 | nextime | kaldemar : well, thanks. sad that is a huge issue for me. |
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| 10:32.36 | Galaxor | Let's say that asterisk negotiates two sip clients to talk to each other. Does asterisk step out of the way and let traffic flow directly between the clients? Or does all the traffic flow through the asterisk box? |
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| 10:50.46 | wdoekes | Galaxor: directmedia=yes makes the RTP flow between the clients |
| 10:50.58 | wdoekes | but the SIP always passes through the asterisk |
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| 11:29.29 | asghar144 | hi, asterisk-1.8.23 send debug to cli with debug set off |
| 11:30.00 | asghar144 | [Sep 18 12:22:05] DEBUG[21529]: channel.c:6390 ast_set_owners_and_peers: setting peeraccount to 9682092255 for SIP/ |
| 11:30.45 | asghar144 | how i can stop this |
| 11:31.24 | kaldemar | core show help core set debug |
| 11:32.16 | *** join/#asterisk andrewya_ (~andrewyag@8-104-141-114.static-dsl.realworld.net.au) |
| 11:34.25 | asghar144 | it is already set off |
| 11:34.53 | asghar144 | core set debug off |
| 11:36.41 | kaldemar | what does "core show settings" say about debug level? |
| 11:38.32 | jmetro | Katty: I've been onsite for the past...week. And am leaving in an hour for another. XD |
| 11:38.44 | jmetro | Katty: let me know how the cross-register went |
| 11:38.48 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 11:38.51 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 11:40.10 | asghar144 | <PROTECTED> |
| 11:40.10 | asghar144 | <PROTECTED> |
| 11:43.07 | hebber | exit |
| 11:49.25 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.221) |
| 11:59.52 | *** join/#asterisk davlefouAMD (~david@197.15.33.146) |
| 12:01.42 | asghar144 | ast_log(LOG_DEBUG, "setting peeraccount to %s for %s from data on channel %s\n", |
| 12:02.45 | asghar144 | this is line number 6390 in channel.c |
| 12:05.50 | *** join/#asterisk dongola7 (~dongola7@unaffiliated/blair/x-0911782) |
| 12:07.20 | asghar144 | i can change to ast_debug(1, "setting peeraccount to %s for %s from data on channel %s\n", |
| 12:07.45 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:07.59 | asghar144 | but i am not sure it is corract way |
| 12:09.38 | *** join/#asterisk davlefouAMD (~david@197.15.33.146) |
| 12:16.56 | kaldemar | asghar144: that's how it is in newer versions (ast_debug(1, ...)). |
| 12:17.26 | karl-s | asghar144, check out ./includes/asterisk/logger.h |
| 12:18.43 | karl-s | I think you are OK with your change (but do you really want that to debug??? i'm sure you have a good reason) |
| 12:24.05 | asghar144 | i am setting accountcode in dialplan so it maybe sometime usefull to debug. |
| 12:27.57 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:28.49 | asghar144 | ok changed, it should not be change also in upstream version_ |
| 12:28.52 | asghar144 | ? |
| 12:32.57 | karl-s | i checked 10.x and there is no AST_log that references peeraccount in channel.c |
| 12:33.28 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:33.52 | kaldemar | karl-s: it has been changed to ast_debug with level 1. |
| 12:34.33 | karl-s | thats what I figured except it looks like they changed the wording a little too. Which would be why i didnt see it from a cursory search |
| 12:37.25 | *** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
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| 12:42.08 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.234) |
| 12:44.53 | asghar144 | in asterisk-11.5 they changed to ast_debug(1, "setting peeraccount to %s for %s from data on channel %s\n", |
| 12:44.56 | *** join/#asterisk tuxx- (tuxx@2a02:2308::216:3eff:feac:73b6) |
| 12:45.12 | tuxx- | hiya. whats the best option to keep cdr records for realtime queues? |
| 12:46.24 | *** join/#asterisk davlefouAMD (~david@197.15.33.146) |
| 12:47.29 | *** join/#asterisk Turl (~Turl@cyanogenmod/maintainer/Turl) |
| 12:48.33 | Turl | hi |
| 12:49.02 | Turl | I am currently having an issue with calls when using tls |
| 12:49.11 | *** join/#asterisk davidbowlby (~textual@99-67-52-166.lightspeed.clmboh.sbcglobal.net) |
| 12:49.39 | Turl | my asterisk is behind a nat, and so is the remote phone |
| 12:50.25 | Turl | if I switch to using udp, it all works fine, but if I use tls it seems the remote phone never acks the 200 OK asterisk sends |
| 12:51.44 | *** join/#asterisk danjenkins_ (~danjenkin@213.106.234.250) |
| 12:53.04 | wdoekes | Turl: if you're both behind NAT, you probably have an ALG that makes things work |
| 12:53.11 | wdoekes | the ALG cannot read your TLS session |
| 12:54.07 | Turl | wdoekes: I'm pretty sure there's no ALG on asterisk's side and ports should be forwarded correctly |
| 12:54.25 | Turl | wdoekes: I have no control over the other nat (3G network) |
| 12:54.48 | raidghost | When i trunk doesnt show up with sip show registry, but does show up with sip show peers, What am i missing then? |
| 12:55.08 | *** join/#asterisk Vann (~manny@c-76-97-51-58.hsd1.ga.comcast.net) |
| 12:55.44 | *** join/#asterisk serafie (~erin@nat/digium/x-ixpyjmsznrgsyuod) |
| 13:01.17 | Turl | wdoekes: the other interesting thing is that if I call the other way (invert caller and called party) it works fine |
| 13:02.51 | Turl | so when caller is outside and calls inside it doesn't work (caller never acks), and if caller is inside and called party outside it works |
| 13:03.53 | Katty | jmetro: it went just peachy! |
| 13:04.04 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
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| 13:07.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:07.54 | niplo | Am trying to display greek characters on grandstream phone. Does anybody know what charset should i use? Am trying to setup CID |
| 13:08.09 | niplo | character set* |
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| 13:11.41 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:13.47 | wdoekes | Turl: and the Contact:-header in the 200 holds how you are reachable over the internet? |
| 13:16.01 | Turl | wdoekes: looks like so |
| 13:16.14 | Turl | wdoekes: hmm, except the port is wrong |
| 13:16.24 | Turl | somehow it put a 5061 on there |
| 13:18.00 | Turl | on the UDP case it uses the correct port |
| 13:21.30 | Turl | wdoekes: adding a "externtlsport" option seems to do the trick, thanks for the pointers |
| 13:22.35 | Turl | it's weird that it uses 5061 when it's configured to listen on something else with tlsbindaddr=ip:port though |
| 13:25.21 | Katty | hello my asterisk does not work at all how to fix plz is urgent thx |
| 13:25.59 | Qwell | Katty: have you tried turning it off and then on again? |
| 13:26.15 | Katty | how to turn off plz???? |
| 13:26.43 | Qwell | Katty: rm -f /proc/power |
| 13:27.15 | karl-s | huh, that actually ran... |
| 13:27.48 | karl-s | root@ast-dev-03-:~# rm -f /proc/power |
| 13:27.56 | Qwell | yes |
| 13:28.11 | karl-s | was hoping for more smoke though |
| 13:28.19 | Qwell | it'll shut down in 3 minutes |
| 13:28.22 | bacobart | /proc/power doesn't exist |
| 13:28.26 | bacobart | and rm doesn't complain about it |
| 13:28.28 | Qwell | bacobart: not anymore |
| 13:28.29 | bacobart | no harm done |
| 13:29.25 | karl-s | root@ast-dev-03-:~# rm -rf /proc/acpi/ ?? |
| 13:29.36 | karl-s | rm: cannot remove `/proc/acpi/battery': Operation not permitted |
| 13:30.00 | karl-s | I probably should get back to coding instead of finding creative ways to break my system |
| 13:30.02 | Katty | it does not work it says 'rm' is not recognize as an internal or external command, operable program or batch file????? |
| 13:30.56 | bacobart | superior operating systems do not support rm |
| 13:31.22 | bacobart | they delete files when it senses you do not need it anymore |
| 13:31.28 | bacobart | or at random |
| 13:31.32 | bacobart | whatever it feels like |
| 13:31.47 | [TK]D-Fender | New system feature : "Feelings" |
| 13:33.38 | asghar144 | my system feel pain when i write rm -f |
| 13:35.03 | wdoekes | Turl: file a bug report |
| 13:35.26 | raidghost | [TK]D-Fender: Trying to setup asterisk without freepbx, added trunk. But for some weird reason it doesnt show up with sip show registry. (i have added the registry string) so it should not be the issue. What could be wrong? |
| 13:36.09 | Katty | a severe lack of cookies. |
| 13:36.21 | Turl | wdoekes: I will, thanks |
| 13:36.37 | [TK]D-Fender | raidghost: maybe you did it wrong. |
| 13:37.22 | [TK]D-Fender | raidghost: Starting with the term "Added trunk". Making a peer entry is just that. A register statement is also somewhat independant of that technically. |
| 13:37.43 | karl-s | raidghost, unless you added a register statement, it wont appear with sip show registry |
| 13:37.47 | karl-s | try sip show peers |
| 13:38.47 | [TK]D-Fender | that won't show it |
| 13:39.07 | raidghost | sip show peers |
| 13:39.09 | raidghost | shows the trunk |
| 13:39.13 | [TK]D-Fender | If you don't see your entry in "sip show registry" ... then you did it wrong. |
| 13:39.57 | raidghost | registry statement as registry string? |
| 13:40.06 | karl-s | not too many commercial sip trunks require registration these days. its probably a 80% shot that when you say 'add a trunk' you just mean adding a peer |
| 13:40.21 | karl-s | raidghost, yes exactly |
| 13:40.26 | raidghost | allready added that |
| 13:40.35 | raidghost | But it doesnt make a diff |
| 13:40.37 | karl-s | ah well in that case |
| 13:40.45 | karl-s | no no, it may make the difference |
| 13:41.02 | karl-s | if you really did add a register string, it should appear in sip show registry |
| 13:41.16 | [TK]D-Fender | raidghost: that would be the "did it wrong" part.... |
| 13:41.22 | karl-s | like [TK]D-Fender said, it was probaly enter wrong |
| 13:41.37 | karl-s | err yea, what he said... |
| 13:42.11 | raidghost | register => 47mynumber:password:47mynumber@sip.provider.com/XXXXXXXX |
| 13:42.43 | karl-s | be sure its under the [general] heading |
| 13:43.08 | raidghost | So thats a must? it cant be in the [My trunk] ? |
| 13:43.17 | [TK]D-Fender | raidghost: NO |
| 13:43.23 | [TK]D-Fender | raidghost: Location counts. |
| 13:43.26 | Katty | your mom's trunk. |
| 13:43.36 | [TK]D-Fender | raidghost: at the end of [general], and before ny other heading |
| 13:43.39 | karl-s | on command I use sometime to sift out those pesky comments is: grep -v ^\; /etc/asterisk/sip.conf | grep -v ^$ |
| 13:44.01 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.243) |
| 13:44.01 | [TK]D-Fender | any* |
| 13:44.02 | karl-s | that will verify that you didnt accidentally put the register in the wrong place |
| 13:44.13 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 13:44.51 | Katty | updates Qwell's emergency contact info |
| 13:47.04 | raidghost | well. ive added the info under [general] in the end, and no diff |
| 13:47.36 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 13:47.44 | karl-s | you did reload right? (have to ask) |
| 13:48.16 | Katty | i'll reload your sip in a minute |
| 13:48.32 | raidghost | yes. |
| 13:48.50 | raidghost | i did reload, karl-s |
| 13:49.00 | Katty | did you ...sip reload? |
| 13:49.20 | raidghost | sip reload or /etc/init.d/asterisk reload |
| 13:49.31 | raidghost | its the same |
| 13:50.05 | karl-s | gothca. Probably need a pastebin of sip.conf (unless you noticed wierd warnings during the sip reload) |
| 13:50.19 | karl-s | theres not too much to the sip registry string... |
| 13:50.43 | Katty | i vote we blame leifmadsen |
| 13:50.49 | Katty | has not blamed leifmadsen in awhile. |
| 13:50.56 | karl-s | i kinda agree with that |
| 13:51.15 | karl-s | we should force him to rewrite the book again |
| 13:51.21 | raidghost | Been trying to find examples at google to use as a example how to setup the sip.conf |
| 13:51.31 | raidghost | But of course NONE of the examples ive found made a diff |
| 13:51.36 | Katty | karl-s: leifmadsen did a wonderful job writing the book |
| 13:51.46 | leifmadsen | Katty: sounds fishy |
| 13:52.08 | karl-s | :) |
| 13:52.08 | Katty | speaking of fish. |
| 13:52.15 | Katty | leifmadsen: have any new recipes for me? |
| 13:52.19 | Katty | file: or you. |
| 13:52.20 | leifmadsen | I do not! |
| 13:52.24 | file | moo. |
| 13:52.24 | Katty | :< |
| 13:52.43 | Katty | hugs file |
| 13:52.59 | karl-s | raidghost, which ast version? |
| 13:52.59 | file | hugs Katty |
| 13:53.10 | raidghost | 1.8 |
| 13:53.11 | Katty | file: how's life n stuff n things? |
| 13:53.30 | file | BUSY |
| 13:53.49 | Katty | file: that's good, yes? |
| 13:53.53 | file | yes |
| 13:54.23 | Katty | yay |
| 13:56.25 | karl-s | yea sorry raidghost cant think of anything it should look something like this guys sip.conf: http://agix.com.au/blog/?p=2656 |
| 13:57.47 | afidegnum | hello, can someone pls tell me what are teh steps of creating a virtual number right from my end? |
| 13:58.09 | afidegnum | this number will be able to receive SMS messages |
| 13:58.41 | Katty | step 1. locate a crayon and post it note |
| 13:58.48 | Katty | step 2. write number on post it note |
| 13:58.55 | Katty | step 3. post to monitor. |
| 13:59.19 | raidghost | http://pastebin.com/Gqi0MZBF |
| 13:59.44 | *** join/#asterisk BK (ca7d90e3@gateway/web/freenode/ip.202.125.144.227) |
| 14:00.01 | Katty | BK: have it your way. |
| 14:00.11 | afidegnum | Katty: I don't get u |
| 14:00.20 | Katty | you're right. you don't. |
| 14:00.56 | Katty | this conversation needs more drmessano |
| 14:01.54 | *** join/#asterisk Stll0 (~tux@nethservice.nethesis.it) |
| 14:01.56 | afidegnum | ok, can u please break it down? I don't understand |
| 14:02.51 | Katty | i'm not nearly caffeinated enough for that yet. |
| 14:03.40 | Katty | did anyone start gta 5 last night? |
| 14:05.26 | Guest47772 | Hi. I am using chan_dongle to take calls from dongle device |
| 14:05.39 | Guest47772 | in asterisk and dialout from asterisk |
| 14:06.05 | karl-s | raidghost, you got some errors |
| 14:06.07 | afidegnum | can anyone assist ? |
| 14:06.27 | karl-s | well actually only 1 error raidghost. Move the useragent line under the [general] context |
| 14:06.35 | Guest47772 | now my requirement to send a sms from console to recharge sim. |
| 14:06.47 | karl-s | because of that lingering line, sip.conf was refusing to load |
| 14:06.58 | *** join/#asterisk asghar144 (~asghar144@host118-24-dynamic.8-87-r.retail.telecomitalia.it) |
| 14:07.04 | Guest47772 | need assistance regarding to it |
| 14:08.26 | raidghost | karl-s: i now learned that the useragent has nowhere to be other than below the [useragent] |
| 14:08.50 | karl-s | ??? i didnt catch that one... sorry |
| 14:09.24 | karl-s | did you mean [general] as opposed to [useragent] ? |
| 14:10.15 | raidghost | karl-s: as putting the useragent = thingy below the [general] |
| 14:10.26 | karl-s | yes! thats exactly it |
| 14:10.33 | karl-s | does sip show registry work ok now? |
| 14:10.39 | raidghost | has to be in the [general] , ive learning. |
| 14:11.04 | raidghost | karl-s: gotta change my firewall settings, cause im guessing that it needs forwarded some ports. |
| 14:11.27 | raidghost | My freepbx server does use the 5060 port at the moment, so i need to change it to see if it makes a diff on the asterisk only server |
| 14:11.37 | [TK]D-Fender | raidghost: It does, but that won't stop it from showing from CLI. the worst that could do is cause it to fail |
| 14:12.08 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:12.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:12.18 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:13.07 | raidghost | so that only shows that my configs are not propper. |
| 14:13.56 | raidghost | cause it should say failed (as long as the 5060 port are in use by my freepbx computer. |
| 14:14.05 | [TK]D-Fender | raidghost: Fix your other settings for port definitions, etc, and show us the new configs & dumps after |
| 14:14.44 | raidghost | port destinations, like the firewall? |
| 14:14.47 | [TK]D-Fender | raidghost: You'll also have to set up a different RTP range <---------- |
| 14:15.01 | [TK]D-Fender | raidghost: what port you BIND for SIP |
| 14:15.14 | raidghost | gonna replace the freepbx computer with the asterisk none freepbx |
| 14:15.38 | raidghost | so to set up a different RTP range is not needed. |
| 14:16.00 | raidghost | But now its dinner. Be back after dinner. |
| 14:34.28 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
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| 14:50.22 | raidghost | the bindaddress thingy in [general] is that the external voip ip to be entered or my asterisk server ip? |
| 14:53.27 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 14:54.06 | mjordan | raidghost: your IP. |
| 14:55.22 | *** part/#asterisk monsterco (~monsterco@64.231.101.21) |
| 14:55.25 | bacobart | or just leave 0.0.0.0 to bind to all interfaces on your system |
| 14:55.28 | [TK]D-Fender | raidghost: just leave as 0.0.0.0 |
| 14:56.01 | [TK]D-Fender | raidghost: and let * bind to all IP's on your server. Yor WAN stuff is in the externaddr |
| 14:56.12 | [TK]D-Fender | raidghost: Along with several otrher required settings for working from behind NAT. |
| 15:00.11 | *** join/#asterisk danjenkins_ (~danjenkin@213.106.234.250) |
| 15:01.15 | *** join/#asterisk aberrios (~aberrios@195.130.201.200) |
| 15:04.04 | raidghost | http://pastebin.com/CEcJQfNu |
| 15:04.12 | raidghost | IS how my sip.conf file looks like right now. |
| 15:04.47 | raidghost | and still it doesnt show up with sip show registry |
| 15:05.58 | [TK]D-Fender | raidghost: register =>47xxxxxxxx:mypassword:47xxxxxxxx@sip.provider.com/71XXXXXX <- has to be at the END of [general] |
| 15:06.10 | Qwell | [TK]D-Fender: it shouldn't matter |
| 15:06.25 | [TK]D-Fender | Qwell: register cuts off all the other [general] setting below it |
| 15:06.34 | [TK]D-Fender | Qwell: Dawn-of-time issue |
| 15:06.47 | *** part/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:06.51 | [TK]D-Fender | :( |
| 15:06.53 | raidghost | Okey. but then i move it to the end of general. 2 sec |
| 15:06.54 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:06.54 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:06.57 | [TK]D-Fender | :) |
| 15:07.09 | [TK]D-Fender | Qwell: @boing |
| 15:07.56 | Katty | jigs through the channel |
| 15:07.59 | Katty | randomly jazzhands |
| 15:08.02 | Katty | jigs out |
| 15:08.05 | raidghost | didnt make any difference |
| 15:08.07 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 15:08.17 | raidghost | So i try to think where the issue could be. |
| 15:09.04 | Qwell | raidghost: show us what it says when you do sip reload |
| 15:09.17 | Katty | hi mister Qwell |
| 15:09.18 | [TK]D-Fender | raidghost: Along with the other status dumps we've asked for |
| 15:09.23 | Qwell | Katty: ohai |
| 15:09.35 | [TK]D-Fender | [11:08]raidghostdidnt make any difference <_ SHOW, don't "say" |
| 15:10.27 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:10.31 | raidghost | http://pastebin.com/yXngSZDt |
| 15:11.26 | [TK]D-Fender | raidghost: show the new configs AND the CLI dumps for both. And while you're at it -> "ls -la /etc/asterisk" |
| 15:13.38 | raidghost | New configs? the one i just pasted 17:04 was the latest sip.conf , but now the ls -la is coming up in pastebin |
| 15:15.45 | [TK]D-Fender | raidghost: after which we told you to make changes |
| 15:15.53 | [TK]D-Fender | raidghost: so it should not be current |
| 15:17.45 | raidghost | http://pastebin.com/hDJvBeS3 (the ls -la |
| 15:22.17 | raidghost | i did make changes. been google some more, but still not show up as registered |
| 15:23.46 | boom^time | I put a limit on my SIP channel for simultaneous outbound, I'm getting this Notice Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) |
| 15:23.48 | boom^time | Which is great |
| 15:23.53 | boom^time | But the disposition is showing up as BUSY |
| 15:24.24 | boom^time | in my CDRs. Is there anyway to discern between the end point being busy and the local trunk being congested? |
| 15:25.44 | boom^time | I have congestions=yes in my cdr.conf but it still logs as BUSY |
| 15:27.18 | [TK]D-Fender | raidghost: I've asked for a complete set of updated configs, CLI dumps, and that folder dump. What you have pasted in the past does not account for the file size I see there and I do not have those updated outputs as requested. If you are not going to be thorough and consistent I will cease assisting you on this. |
| 15:28.14 | raidghost | When you say updated confings. i guessing you talk aboute other files than only the sip.conf file |
| 15:28.24 | asghar144 | raidghost: are you sure asterisk reading from the sip.conf wich you changing and not from some other sip.conf? what are the content of asterisk.conf? |
| 15:29.26 | raidghost | asghar144: asterisk.conf doesnt have the same things as my sip.conf file |
| 15:30.11 | raidghost | asterisk got some [compat] stuff, and thats all. |
| 15:30.55 | asghar144 | raidghost: asterisk.conf have config file locations |
| 15:31.15 | [TK]D-Fender | raidghost: I am talking about your updated sip.conf. It should have changed. We don't see that. The file size does not look like it matches what you have shown us in the past at all. We don't see the nuew dumps from CLI |
| 15:32.05 | raidghost | [TK]D-Fender: Whats the point of showing new dumps of CLI when there is nothing changed. its the same text |
| 15:32.37 | raidghost | I have tried to pull together a sip.conf file. But you talking aboute change stuff. i dont know what more to change |
| 15:33.12 | asghar144 | mine is astetcdir => /etc/asterisk in asterisk.conf |
| 15:33.29 | raidghost | its the same in my asterisk.conf file |
| 15:33.41 | raidghost | i didnt see on the top, where the file dir stuff was |
| 15:35.56 | *** join/#asterisk danjenkins_ (~danjenkin@213.106.234.250) |
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| 15:45.15 | [TK]D-Fender | moves on to other matters |
| 16:02.18 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:10.56 | *** part/#asterisk Stll0 (~tux@nethservice.nethesis.it) |
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| 16:18.59 | *** join/#asterisk pcAngel (yoink@S0106602ad072bdf0.vc.shawcable.net) |
| 16:19.38 | pcAngel | Hi guys, I am implementing queue member priorities today and we have some users with two phones (SIP/295 and SIP/296 - one being a desk phone, the other a soft phone) |
| 16:20.07 | pcAngel | Is there a way for us to tie the two phones together so that if either are on the phone, it considers this person to be unavailable (ie pause both or consider both to be inuse)? |
| 16:20.40 | pcAngel | I was thinking it may have something to do with adding them as a Local extension, with a single hint |
| 16:21.00 | [TK]D-Fender | just create the hint and specify it as the state_device |
| 16:22.09 | pcAngel | do you know of any documentation or example that can help me with that? |
| 16:22.31 | [TK]D-Fender | state_device is where you add the memeber |
| 16:22.41 | [TK]D-Fender | and composite hints are just & <- |
| 16:24.05 | pcAngel | so maybe have one phone register to 295A, the other 295B, and then create [context] exten => 295,hint,SIP/295A&SIP/295B \ exten => 295,1,Dial(SIP/295A&SIP/295B) ? |
| 16:24.38 | [TK]D-Fender | I didn't say anything about a dial |
| 16:24.42 | [TK]D-Fender | your memeber is your member... |
| 16:24.47 | [TK]D-Fender | the hint is composite though. |
| 16:25.00 | pcAngel | is the hint supposed to be in the queue member config? |
| 16:26.05 | [TK]D-Fender | yes |
| 16:29.59 | pcAngel | Thanks I have enough to research now & am getting somewhere =) |
| 16:35.23 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 16:49.36 | pcAngel | D-Fender: do you have any suggestions on how I can get two queue members to share the same state_interface? I've tried specifying the state_interface as SIP/FirstAccount&SIP/SecondAccount, it didn't report in use or ringing, even though an extension hint set up that way reported both as InUse/Ringing from CLI core show hints |
| 16:51.10 | file | you can't join devices like that for a state interface, it allows only one |
| 16:51.53 | pcAngel | file: I'm trying to make it so that if a user with two phones has either phone in use, they aren't rang on the other phone, from a queue that has ringinuse=no |
| 16:52.17 | pcAngel | file: do you have any suggestions? I also tried adding the member through a Local/xxx@context with the hint set up and functioning |
| 16:52.25 | file | not off the top of my head |
| 16:55.12 | [TK]D-Fender | pcAngel: You should be able to point to a singular hint directly |
| 16:55.17 | *** join/#asterisk deweydb (~deweydb@athedsl-4465989.home.otenet.gr) |
| 16:56.29 | deweydb | hello. I'm running Asterisk 1.8 and FreePBX 2.10, and for some reason it won't auto load app_stack.so on boot. I've tried adding: "load => app_stack.so" into the /etc/asterisk/modules file but when i reboot, i still have no access to modules in that file, such as gosub. |
| 16:56.30 | deweydb | if i do: "core show application gosub " it says: Command 'core show application gosub' failed. but i can do: "module load app_stack.so" and it loads fine. |
| 16:56.30 | deweydb | how do i get it to always load app_stack.so automatically? |
| 16:59.05 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:59.05 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 17:01.04 | paulc | pcAngel: How about making a Local channel agent for the two devices, and using a hint for that end point pointing to the both devices as previously discussed? Not 100% sure it'd work, but might be worth a try |
| 17:01.38 | file | hints indicate extension state which is an aggregation of device state |
| 17:01.48 | file | a state interface within app_queue is for device state |
| 17:02.13 | paulc | ah.. ignore my ramblings then :) |
| 17:02.15 | pcAngel | yeah I tried that and it didn't work, depending on how I specified it, it showed as Invalid or Not in use |
| 17:02.43 | file | I don't believe there is a "device" state which is actually a view of an extension state, although that could be interesting |
| 17:03.00 | file | putting it into a loop would be hilarious |
| 17:03.03 | pcAngel | I think I need two extensions to share a device state-no idea how to do that. |
| 17:03.06 | pcAngel | I have a dirty idea |
| 17:03.43 | pcAngel | SIP/299A, SIP/299B, SIP/299 -- register the phones to SIP/299A and SIP/299B, have asterisk register to 299@localhost |
| 17:04.26 | pcAngel | then to call the extensions from the queue, call SIP/299A@299 & SIP/299B@299 from a Local/ |
| 17:04.31 | pcAngel | with device state set to SIP/299 |
| 17:05.38 | pcAngel | Unfortunately I actually have five extensions to link for myself, four for another person, and a few other people with at least 2 or 3, so even if it works it'd be a configuration management nightmare.. |
| 17:08.18 | pcAngel | oh. I can check the device states on a local channel and reject the call if any of the set of devices are already InUse/Ringing |
| 17:12.02 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:14.05 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 17:14.16 | Katty | i need a new series to watch. |
| 17:14.20 | *** join/#asterisk j4jackj (jack@99.199.11.127) |
| 17:18.01 | file | I wonder what else such a device state provider could be used for... |
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| 17:40.47 | paulc | Katty: Have you seen Derek? (from Ricky Gervais) |
| 17:41.03 | Katty | doesn't ring a bell |
| 17:41.06 | Katty | is that a series? |
| 17:41.30 | paulc | Katty: Yeah.. http://www.youtube.com/watch?v=Hd5WdxGRNG8 |
| 17:42.21 | paulc | Of course.. the British humour may not cross the pond successfully.. can be a bit hit and miss sometimes ;-) |
| 17:53.34 | *** join/#asterisk ibercom (5121944c@gateway/web/freenode/ip.81.33.148.76) |
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| 18:12.39 | _Corey_ | Anyone know of a provider offering DIDs in Rwanda? |
| 18:13.14 | *** join/#asterisk mic__ (~mic@0305ds4-vby.0.fullrate.dk) |
| 18:16.39 | *** join/#asterisk roderickm (~roderickm@67.63.143.254) |
| 18:21.25 | mic__ | channel.c: Didn't receive a media frame from SIP/provider-00000000 within 500 ms of answering. Continuing anyway |
| 18:21.36 | mic__ | I checked networking etc. |
| 18:21.47 | mic__ | and I am out of ideas. |
| 18:22.16 | *** join/#asterisk viasanctus (~viasanctu@unaffiliated/viasanctus) |
| 18:22.29 | *** join/#asterisk monsterco (~monsterco@64.231.101.21) |
| 18:22.56 | monsterco | where can I find official documentation on parameters like "insecure=very", "insecure=port,invite" for various versions of Asterisk? |
| 18:25.17 | [TK]D-Fender | ver = 1.2 and lower |
| 18:25.31 | [TK]D-Fender | And was documented in the "upgrade.txt" |
| 18:25.39 | [TK]D-Fender | very* |
| 18:28.52 | monsterco | does Digium provide any official reading material on this online? |
| 18:29.11 | [TK]D-Fender | It's in the tarball |
| 18:29.18 | [TK]D-Fender | That is where the changes are documented |
| 18:30.54 | Katty | headdesks repeatedly |
| 18:32.07 | [TK]D-Fender | Katty: Target-lock acquired! |
| 18:32.16 | [TK]D-Fender | wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM!wham!WHAM! |
| 18:32.20 | Katty | *hee* |
| 18:32.36 | monsterco | [TD]D-Fender - changes are there but how about documentation on those features? Is there any way I can pull that info from CLI? I mean what parameters I am allowed to use and what other options are available like mediartp or something like that I have seen before... |
| 18:33.40 | *** join/#asterisk roderickm (~roderickm@67.63.143.254) |
| 18:34.00 | [TK]D-Fender | monsterco: No, there is no "complete documentation" built into CLI. That is what the changelogs descibe as far as what it was changed FROM is concerned. For synax, that is what the SAMPLE CONFIGS help show you |
| 18:48.56 | [TK]D-Fender | monsterco: There is also the official WIKi for more background. |
| 18:51.14 | mjordan | and, as an aside, while we at Digium does a lot of work and all on this stuff, documentation is a community affair. It's the overall Asterisk community that provides documentation, not just us. |
| 18:51.20 | mjordan | s/does/do |
| 18:52.13 | mjordan | [TK]D-Fender: as a random aside, and only tangentially related, we did add CLI documentation for a *very* limited subset of configuration files in Asterisk 12 (but I'm pretty sure that isn't what he's running) |
| 18:54.18 | [TK]D-Fender | mjordan: Given the general stability of dev branches it'd be "skipping" at best ;) |
| 18:54.42 | [TK]D-Fender | "Look how much further it bounced!" |
| 18:55.12 | mjordan | Well, there are some rather large changes |
| 18:55.14 | mjordan | :-) |
| 18:55.40 | mjordan | But, that's what standard releases are for: to get the architectural changes in so that we don't do that during an LTS |
| 19:01.28 | monsterco | [TK]D-Fender - so core show help or anything like that can't help with me with configs like "insecure" then? |
| 19:02.25 | *** join/#asterisk modesto916 (~modesto@189-90-192-72.isimples.com.br) |
| 19:02.57 | [TK]D-Fender | monsterco: No, that is in the sample config |
| 19:03.03 | [TK]D-Fender | monsterco: and an ANCIENT one at that |
| 19:05.07 | monsterco | I don't see anything on Wiki pretaining to insecure or canreinvite |
| 19:06.05 | [TK]D-Fender | [15:02][TK]D-Fendermonsterco: No, that is in the sample config |
| 19:06.52 | [TK]D-Fender | Applications & APIs are documented on the WIKI, straight-up configs typically within the sample config files |
| 19:08.13 | *** join/#asterisk roderickm (~roderickm@67.63.143.254) |
| 19:10.39 | monsterco | right - so I don't see canreinvite in Asterisk 1.8.2.3 sip.conf.sample - does that mean it's replaced by directmedia? |
| 19:10.50 | monsterco | [TK]D-Fender^^ |
| 19:11.26 | [TK]D-Fender | yes, in 1.6 |
| 19:11.33 | [TK]D-Fender | monsterco: You are looking at very old stuff |
| 19:21.49 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 19:22.38 | boom^time | Hey guys, what's a good way to limit the amount of simultaneous calls with AMI? With call files I can simply limit the amount of files in the dir. I can't figure out how to do it with an AMI connection. |
| 19:22.58 | [TK]D-Fender | boom^time: there is none |
| 19:23.05 | boom^time | I keep going back and forth between AMI and call files. I can't make up my mind I like them both. |
| 19:23.19 | boom^time | [TK]D-Fender, That's disappointing. |
| 19:26.14 | boom^time | Darn, and the originate response action doesn't tell me when the call is finished so I can't manage it that way. |
| 19:27.05 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.87) |
| 19:32.18 | boom^time | [TK]D-Fender, I would like to defer to your expertise if you don't mind. What do you think is more practical for call origination? I like AMI because of TLS and connecting to multiple asterisk servers for load balancing, future scalability in both the number of servers and commands I can send. But call files are simpler to implement and I can manage the amount of simultaneous calls. |
| 19:34.23 | [TK]D-Fender | tons of ways to manage the count with AMI. and AMI doens't have to have FS access |
| 19:35.40 | boom^time | How can I manage the count, (I'm assuming you mean amount of concurrent calls)? |
| 19:37.31 | [TK]D-Fender | dial local channel, use group count. |
| 19:37.45 | [TK]D-Fender | Set a channel variable. Poll active channels for it |
| 19:37.45 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 19:37.52 | [TK]D-Fender | res_psychic |
| 19:37.55 | [TK]D-Fender | Tons of ways |
| 19:38.00 | [TK]D-Fender | use your imagination. |
| 19:38.53 | mic__ | can someone throw a pointer - or a clue - when calling in locally -> busy tone ok. Calling from outside via mobile -> silence |
| 19:39.10 | mic__ | (nah, not busy tone = waiting tone, sorry) |
| 19:39.20 | boom^time | [TK]D-Fender, thanks |
| 19:40.41 | mic__ | when calling locally it gets immediately the RTP |
| 19:41.09 | mic__ | when calling from outside asterisk throws a message "didn't receive a media frame from xxx within 500 ms". continuing anyway |
| 19:48.22 | monsterco | directmedia is new - isn't it? I just solved a one-way audio issue with putting directmedia=no - this took few days - darn it - why change things around like that |
| 19:50.46 | [TK]D-Fender | [15:11][TK]D-Fenderyes, in 1.6 <- several years ago |
| 19:51.05 | roderickm | f/k/a canreinvite |
| 19:51.07 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
| 19:53.19 | boom^time | [TK]D-Fender, earlier you mentioned polling active channels for a channel variable. What's the best way to do that? I was trying something like core show channel sip/501-00000-00000001 and then searching for the variables section but I was hoping for a better way. |
| 19:54.10 | [TK]D-Fender | boom^time: AGI Get Variable, raw dump, whatever, take your pick |
| 19:58.18 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:59.46 | mic__ | monsterco: well, it's not one way audio |
| 20:00.21 | mic__ | I am more concerned about asterisk complainig, that it did not get any frame within 500 ms after I run ANSWER via AGI |
| 20:05.35 | *** part/#asterisk monsterco (~monsterco@64.231.101.21) |
| 20:05.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.87) |
| 20:13.11 | mic__ | smells both asterisk and the sip provide are waiting for RTP |
| 20:13.19 | mic__ | and since nobody is sending, it's silence. |
| 20:15.05 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 20:26.52 | boom^time | Anyone know how to tie an AMI originate to the Newchannel event associated with it? |
| 20:28.35 | *** join/#asterisk MrUC (~MrU@cpe-70-95-157-210.hawaii.res.rr.com) |
| 20:29.02 | boom^time | This is what happens: http://pastebin.com/WDEd35QS |
| 20:29.34 | boom^time | But that Newchannel could be for any originate that is to SIP/501 |
| 20:30.43 | nextime | boom^time : se a channel variable in the originate call |
| 20:30.48 | nextime | and trace this variable |
| 20:30.51 | nextime | s/se/set |
| 20:31.34 | [TK]D-Fender | checkout time, BBIAB |
| 20:32.14 | boom^time | nextime, I set blah=hi just now and I don't see it anywhere after the originate |
| 20:32.49 | boom^time | http://pastebin.com/NaArEm0T |
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| 20:34.24 | nextime | boom^time : make the originate call an extension, in the extension use agi or fastagi to trace the call |
| 20:34.52 | mic__ | ok - I fixed it |
| 20:35.29 | mic__ | I just push some data on the channel, so that my side starts "talking" - and then it works ;) |
| 20:36.15 | boom^time | nextime, I'm sorry I'm a little confused on how that would work. Would you mind elaborating further? How does the AGI trace the call? |
| 20:37.21 | nextime | boom^time : the agi or fastagi can detect the variable you set in the originate and then recognize the call. It can also get the source channel, and then pass it to the ami daemon to continue to trace it from ami |
| 20:37.47 | nextime | so, you need to make the agi/fastagi side and the ami side communicate in some way |
| 20:38.18 | nextime | personally i have a daemon that implement both fastagi and ami to do that, using twisted python and starpy |
| 20:38.30 | WIMPy | Is there something I've missed? Why use an AGI? |
| 20:38.32 | nextime | but of course there are other ways to do the same concept |
| 20:38.51 | nextime | WIMPy : did you have a better way to do that? |
| 20:39.05 | boom^time | Well if I could just follow the uniqueid of the channel that would be great except that isn't set at origination |
| 20:39.23 | WIMPy | If you're using AMI you will see the variable getting set. |
| 20:39.24 | nextime | boom^time : exactly, in the originate you don't know the unique id yet |
| 20:40.06 | nextime | WIMPy : when you call an originate you have the return from the originate, but you don't know the channel that the originate will create as it doesn't exist yet |
| 20:40.30 | nextime | and in the other events in the channel you don't see the var you set in the originate |
| 20:40.33 | boom^time | WIMPy, here is an example, I don't see the variable blah=hi get set http://pastebin.com/Wi2EHyA4 |
| 20:41.44 | WIMPy | Hmm. Maybe not if you AMI originate? But you could always GetVar it. |
| 20:42.25 | boom^time | WIMPy, how would that work exactly? |
| 20:42.35 | WIMPy | Seems strange. Usually you get events for variables being set, even on channel creation. |
| 20:43.06 | WIMPy | When you see a channel being created, you could try to red your variable from that channel. |
| 20:43.25 | WIMPy | If it exists, you know it's a channel you're interested in. |
| 20:43.30 | boom^time | I see, doesn't seem very efficient. |
| 20:43.57 | boom^time | I wish the ActionID would follow |
| 20:44.04 | boom^time | seems like that's pretty much what it was designed for too. |
| 20:44.25 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 20:44.30 | WIMPy | Might be easiest to use a local channel. |
| 20:44.57 | boom^time | I'm having a hard time envisioning how that would work, D-Fender suggested it to me earlier. |
| 20:45.16 | nextime | WIMPy : no, you can't just use getvar in all events as i remember |
| 20:45.42 | nextime | if i remember right, there is no way to get it in the newchannel event |
| 20:46.13 | nextime | ( maybe i'm wrong, but i remember something like this for which i've implemented the way with a fastagi ) |
| 20:46.27 | WIMPy | You obviousely don't use it IN an event. |
| 20:46.59 | WIMPy | boom^time: What exactely are you doing? |
| 20:47.56 | nextime | WIMPy : right, but what if i need to get the "newchannel" event and know that this newchannel is relative to my originate, or many if the originate fail and no channel is created? |
| 20:48.04 | boom^time | WIMPy, would you mind explaining the local channel method for me? |
| 20:48.16 | nextime | basically, if the originate fail your way using getvar won't work |
| 20:49.16 | WIMPy | What could make it fail without creating a channel? Other than originating something invalid? |
| 20:49.28 | nextime | WIMPy : a busy line? |
| 20:49.43 | nextime | a non registered sip user? |
| 20:50.01 | WIMPy | Yu can't find out somethign is busy without trying and that needs a channel. |
| 20:50.18 | WIMPy | Ok, that might "work". |
| 20:51.01 | WIMPy | boom^time: With a local channel you go to your dialplan, where you can do whatever you want, like using UserEevent. |
| 20:52.39 | boom^time | How do you originate to a local extension? Action: Originate Channel: a8005551212 |
| 20:52.52 | boom^time | and then look for anyhing _a[NXXNXXXXXX] |
| 20:53.09 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 20:53.18 | boom^time | do an odbc function ot upload the channels unique id to a database |
| 20:53.25 | WIMPy | local/extension@context |
| 20:53.30 | boom^time | Thanks |
| 20:53.40 | boom^time | Does the rest of my guesswork sound right? |
| 20:53.52 | WIMPy | channel: local/extension@context that is. |
| 20:54.08 | boom^time | I ugess i don't need the a prefix |
| 20:54.13 | boom^time | just a custom context |
| 20:54.20 | WIMPy | You can name your extensions anythiong you like. e.g. the peer you want to call or a number. |
| 20:54.28 | WIMPy | yes |
| 20:55.06 | navaismo | dont forget the /n |
| 20:55.23 | WIMPy | Why? |
| 20:55.31 | boom^time | /n? |
| 20:55.41 | navaismo | WIMPy: just in case |
| 20:55.54 | WIMPy | In case of what? |
| 20:56.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:57.14 | navaismo | issue with vars if i recall |
| 20:57.54 | boom^time | What if you just always assumed you wouldn't call the same channel during the same time. So you just followed the newchannels and cut off anything following - |
| 20:58.48 | navaismo | WIMPy: in the past not using /n on the local channel works weird for me, let me digg where i found the tip about using the /n |
| 20:58.50 | WIMPy | Don't use channel names. That will only work with luck. |
| 20:58.55 | boom^time | ie if I call SIP/501 I follow Channel: SIP/501-00000003 |
| 20:59.08 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 20:59.34 | boom^time | Only follow the uniqueid then? |
| 20:59.36 | WIMPy | That seems to be true for chan_sip, but it's nether a documented behaviour nor true for other channeltypes. |
| 20:59.43 | boom^time | Gotcha, thanks for the warning |
| 21:00.11 | WIMPy | You can use the channel name, once you have identified the cahnnel by other means. |
| 21:00.16 | boom^time | Okay |
| 21:00.36 | navaismo | ahahaha "Esoteric"--->http://www.voip-info.org/wiki/view/Asterisk+local+channels |
| 21:00.42 | boom^time | Thanks for all of the help guys I need to run. Will let you know how I do. |
| 21:01.59 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 21:10.13 | *** join/#asterisk jpoz (~jpoz@184.169.152.1) |
| 21:11.07 | jpoz | maybe a newbie question but is there anyway via AMI or CLI to get a rrmemory queues order? |
| 21:11.46 | jpoz | to know who's going to be the next agent a call will be connected to |
| 21:17.00 | cusco | er |
| 21:17.13 | cusco | I don't think so.. |
| 21:24.13 | MrUC | followed the asterisk quick start guide... dialing 2600 from sip phone calls "IAX2/guest@pbx.digium.com/s@default"; problem is the call is not connecting to the IVR |
| 21:24.39 | [TK]D-Fender | IIRC Digium took down their IAX2 test server |
| 21:25.13 | MrUC | i was about to ask that; how could we verify if its online or offline? |
| 21:25.55 | navaismo | iax2 show peers |
| 21:26.35 | MrUC | status: unmonitored |
| 21:26.53 | MrUC | demo/asterisk 216.207.245.47 (S) 255.255.255.255 4569 Unmonitored |
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| 21:28.00 | navaismo | qualify it and check again hehe |
| 21:28.31 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:29.26 | MrUC | how do you qualify it? |
| 21:29.52 | MrUC | (i'm new to * and i'm starting in the quick start guide) |
| 21:30.57 | [TK]D-Fender | what "quick start guide" exactly? |
| 21:31.11 | navaismo | MrUC: below the iax2 peer config add qualify=yes |
| 21:32.05 | MrUC | http://www.asterisk.org/sites/asterisk/files/mce_files/documents/asterisk_quick_start_guide.pdf |
| 21:32.50 | jpoz | anyone who rrmemory queues are ordered? |
| 21:34.50 | MrUC | linked from http://www.asterisk.org/get-started |
| 21:36.32 | [TK]D-Fender | mrcCould be a defunct carry-over... |
| 21:36.45 | [TK]D-Fender | MrUC: I would move on with other testing |
| 21:41.19 | MrUC | fender - sure. so i'm new to asterisk could you recommend other reads/guides available on the web? i ordered the * oreilly book; waiting it for to arrive |
| 21:45.22 | *** join/#asterisk s14ck (uid6427@gateway/web/irccloud.com/x-wxaixdhhyrukmhql) |
| 21:45.36 | s14ck | hey yo |
| 21:47.10 | [TK]D-Fender | MrUC: you can view it online in the meantime |
| 21:48.01 | mic__ | that was a productive day |
| 21:48.19 | mic__ | only 16h at work and enden up with a sky-high headache :D |
| 21:48.32 | MrUC | only? |
| 21:48.36 | MrUC | ouch |
| 21:49.15 | mic__ | running your own business is fun |
| 21:49.22 | mic__ | but sometimes it's also like that... ;) |
| 21:49.40 | mic__ | and sometimes it's just bloody tears and PITA |
| 21:49.55 | mic__ | :D |
| 21:50.05 | MrUC | cool - how long have ya been in business? |
| 21:50.26 | mic__ | 2.5 years |
| 21:51.13 | MrUC | very cool |
| 21:51.21 | MrUC | i'm guessing IT/voip related? |
| 21:52.49 | mic__ | open source, voip is a new thing |
| 21:53.06 | mic__ | kind of started doing that "by mistake" |
| 21:53.13 | mic__ | I went to a guy to fix his database |
| 21:53.14 | mjordan | MrUC: I've removed that guide you found. It's quite out of date. Asterisk: the Definitive Guide is the best book on Asterisk. You can read it online at asteriskdocs.org |
| 21:53.40 | mic__ | MrUC: and then I ended up having VOIP on the menu some months later ;) |
| 21:53.44 | mjordan | MrUC: for supplemental stuff, you may want to check out the Asterisk wiki, which has command reference for the various versions, as well as some getting started/configuration sections (wiki.asterisk.org) |
| 21:54.02 | *** join/#asterisk luke_ftw (~luke_ftw@unaffiliated/luke-ftw/x-9245329) |
| 21:54.26 | MrUC | mjordan - roger and thanks |
| 21:55.25 | MrUC | mic - databases... heh |
| 21:59.00 | mic__ | MrUC: I recommend the asterisk book |
| 21:59.43 | mic__ | MrUC: I was also going through Asterisk 1.6 from packt (not much use currently unless you have to deal with legacy stuff) |
| 22:00.37 | MrUC | thanks mic - good 2 know |
| 22:00.38 | mic__ | MrUC: VoIP hacks (O'Reilly) was also interesting - inspired me to do a few nice things. And a book all other programmers here have to go through is "Packet guide to Voice over IP" |
| 22:02.31 | mic__ | good night from this timezone ;) |
| 22:02.33 | mic__ | & |
| 22:02.48 | MrUC | good night |
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| 23:26.26 | MrUC | broken link - http://devcon.digium.com/calendar/AsteriskDevCall |
| 23:27.30 | MrUC | from https://wiki.asterisk.org/wiki/display/AST/Asterisk+Developer+Conference+Call |
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| 23:52.00 | *** mode/#asterisk [+o newtonr] by ChanServ |