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| 01:32.54 | tompaw | ok, something just exploded my brain |
| 01:33.22 | tompaw | So I have this little cluster of asterisks sharing the same postgres for realtime configuration of pjsip endpoints |
| 01:34.15 | tompaw | The plan is to have a few agents assigned to each box (via conference bridges) and have boxes pull extra agents if needed using 2 contexts: agent-local, agent-otherbox |
| 01:34.42 | tompaw | It's very simple, because the CRM that orchestrates the whole thing knows on which box the conference is held and the newly pulled agent is registered. |
| 01:35.00 | tompaw | So if it's the same box, it would pull via @agent-local, otherwhise via @agent-otherbox. |
| 01:35.57 | tompaw | *NOW* I accidentaly called an agent registered on server B from server A using PJSIP/123, so as a Local extension (from A's point of view). And the call went through! |
| 01:36.40 | tompaw | So obviously server A knew where to knock, because it shares AOR DB with server B, but why would zoiper regiestered to server B allow a call from server A to go through? |
| 01:41.46 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 01:42.56 | tompaw | I wonder if it's a freak of nature, or something I can rely on in production. |
| 01:44.24 | tompaw | I'm gonna draw it out, because I doubt anyone understands what I'm talking about :> |
| 01:48.47 | haryv | pretty quiet tompaw |
| 01:49.51 | haryv | talk about zoiper, I do not hear audio on my install on android. I think this is a first for a internal issue. normally, its linking up with a sip provider that it can be a issue. |
| 01:56.32 | *** join/#asterisk BBone_ (~Thunderbi@s75-157-233-55.bc.hsia.telus.net) |
| 01:56.46 | tompaw | ok, here it is: http://tompaw.net/astbridge.png |
| 01:56.57 | tompaw | The line market with "X" is what I'm asking about. |
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| 01:59.03 | tompaw | I mean, it's great it works like this - but I'm not sure it's supposed to. I thought a SIP client should only ever speak SIP to his registrar. |
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| 02:13.43 | WIMPy | No, SIP phones often doens't care at all where a call comes from. |
| 02:14.14 | WIMPy | Some do have a secutity setting for it. |
| 02:14.48 | WIMPy | The thing is that ITSPs may send calls from multiple IPs as well. |
| 02:15.04 | tompaw | WIMPy: did you check out my awesome chart? |
| 02:15.14 | WIMPy | Yes. |
| 02:15.31 | WIMPy | But it was clear without the picture. |
| 02:15.41 | tompaw | And this behavior is normal and can be used in production? |
| 02:16.19 | WIMPy | It obviousely depends on the client used and its configuration, but usually yes. |
| 02:16.37 | tompaw | I mean, this jus makes the whole thing so much better, because now my bridge "Conf." will only use resources of one box, leaving AST-B idling. |
| 02:18.21 | WIMPy | The stateless nature of SIP can have its advantages. |
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| 02:19.12 | tompaw | Everything is awesome! |
| 02:19.25 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 02:20.46 | tompaw | WIMPy: do you reckon it's ok to share aor/contacts via single DB like in my picture? |
| 02:21.41 | WIMPy | I'd expect so, but for the ultimate answer, wait for file. |
| 02:21.55 | tompaw | Cheers. |
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| 04:13.19 | Jason_weileilei | exten => _+.,n,set(LIMIT_PLAYAUDIO_CALLER=yes) is right ? |
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| 08:49.02 | EmleyMoor | I had an unexpected failure of my "fake number detection" regular expression yesterday, when it seemingly let a number beginning with 000 through... have had to add another dialplan line for now but am wondering why it failed... |
| 08:53.32 | overyander | EmleyMoor, some more information about the full number that passed through and your full dialplan would be extremely helpful. or i can just take a guess and say that it failed because of jesus. |
| 08:54.04 | EmleyMoor | overyander: Will provide, shortly |
| 09:01.51 | EmleyMoor | Relevant section, with note indicating relevant line: http://paste.debian.net/289653/ |
| 09:02.10 | EmleyMoor | Number that was not detected: 0008000931608 |
| 09:04.53 | EmleyMoor | (ARG1 carries the extension that the call was directed to) |
| 09:05.02 | *** join/#asterisk danielbw (~danielbw@70.167.122.27) |
| 09:05.54 | EmleyMoor | Ah, two lines were pasted in twice... |
| 09:06.20 | EmleyMoor | (24 and 25 should be disregarded) |
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| 09:12.00 | overyander | correct me if i'm wrong, just going off of memory, but doesn't line 35 only catch it if the number is exactly "000"? |
| 09:14.55 | overyander | i know you can use something like "0001":3 to cut the first 3 characters off (which would result in a variable with a value of "1"), if the CID is always the same length you can cut the extra off then pass that value to 35 and see if it matches to "000" |
| 09:15.19 | EmleyMoor | It's the : operator... matches regular expression at beginning |
| 09:15.48 | EmleyMoor | If it was =, you'd be right. |
| 09:16.33 | EmleyMoor | Either that or I've misunderstood how Asterisk's =~ and : operators work |
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| 09:18.15 | EmleyMoor | Besides, lines 39 and 40 do exactly what I want them to (pass 01... numbers of the right length only), so it seems I have understood it correctly |
| 09:18.43 | overyander | yes, sorry, i misread it |
| 09:18.51 | overyander | i've always hated regex's |
| 09:19.24 | overyander | have you tried testing it with a regex tester? there are several sites that i always use to make sure i'm not missing anything |
| 09:19.35 | EmleyMoor | Yes, I've tried that - it passes |
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| 09:31.13 | *** join/#asterisk Dovid (~dovid@ool-4356e96f.dyn.optonline.net) |
| 09:39.26 | Rico | I'm looking at rfc3665. When asterisk acts as an UAC (fax receiving for example), I guess it should set a "Route :" header to its BYE when terminating the communication |
| 09:39.28 | Rico | isn't it ? |
| 09:40.29 | Rico | it does not |
| 09:41.04 | Rico | ooops |
| 09:41.05 | Rico | it does |
| 09:41.07 | Rico | :) |
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| 11:20.05 | overyander | Rico, were you just talking to yourself? |
| 11:30.33 | Rico | kind of... :) |
| 11:31.28 | Rico | I'm trying to understand why BYEs sent by my asterisk to my SBC are not correctly understood |
| 11:46.12 | Ice_Strike | Can you give me suggestion to write a dialplan something like AgentLogin() without agent.conf requrement? |
| 11:46.44 | Ice_Strike | Basically login to queue and wait for the calls? |
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| 12:24.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:25.32 | [TK]D-Fender | Ice_Strike: Yesterday you were given the command for this.... |
| 12:30.07 | *** join/#asterisk rvvliet (~rvv@2001:610:600:86a5:1:ffff:fffe:e329) |
| 12:39.33 | ruelp | is asterisk very ideal for just a b2bua? or is it overkill? |
| 12:41.51 | mub | ruelp: Naw, asterisk isn't overkill |
| 12:42.07 | mub | I have it on a small arm box just for me at home |
| 12:42.11 | [TK]D-Fender | ruelp: That's all * is.... |
| 12:42.35 | [TK]D-Fender | Many people run it on shitty Linksys routers... |
| 12:49.29 | *** join/#asterisk creativx (~ntttt@226.62-97-205.bkkb.no) |
| 12:49.38 | Ice_Strike | [TK]D-Fender No, that is different. |
| 12:50.55 | [TK]D-Fender | Ice_Strike: Caller enters the queue a sits there waiting for a caller to be distributed to them. They hear a beep and answer. All while sitting in a call |
| 12:51.21 | Ice_Strike | Indeed |
| 12:52.57 | [TK]D-Fender | https://reviewboard.asterisk.org/r/2657/ |
| 12:54.05 | Ice_Strike | The new app_agent_pool - AgentLogin() still requre agents.conf ? |
| 12:54.43 | [TK]D-Fender | You were told "no" yesterday |
| 12:55.27 | Ice_Strike | Hmmm |
| 12:55.32 | Ice_Strike | Let me scroll up |
| 12:57.52 | Ice_Strike | [TK]D-Fender I have your conversation yesterday, no where you said "no" yesterday |
| 12:57.54 | *** join/#asterisk bkruse (~Adium@user-24-96-51-167.knology.net) |
| 13:01.43 | [TK]D-Fender | http://picpaste.com/ast-DBIdBDbK.png |
| 13:04.13 | Ice_Strike | I am gonna play around with it. |
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| 13:25.37 | mub | Leave an unsecured public-facing asterisk box running overnight. Wake up to a kick-ass media server |
| 13:25.48 | mub | Make sure to leave the default passwords |
| 13:26.57 | [TK]D-Fender | #no_KY |
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| 15:23.42 | aRdoR | Cheap PSTN provider to terminate phone calls? any idea? |
| 15:23.59 | aRdoR | I mean, I need a sip provider for my softphone. |
| 15:24.07 | aRdoR | but need to call normal US numbers. |
| 15:24.43 | [TK]D-Fender | ~itsplist-us |
| 15:24.43 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
| 15:24.51 | [TK]D-Fender | ^ |
| 15:24.59 | aRdoR | thanks I will look at them |
| 15:25.22 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 15:27.25 | aRdoR | thanks TKD-Fender these are already way better than what I was finding |
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| 16:04.44 | marceloamorim | guys, what kind of hash asterisk work with? I used to put this on linux to set my hash "echo "word" |md5sum |cut -c1-10" |
| 16:04.58 | marceloamorim | I'm trying the app Authenticate |
| 16:14.21 | marceloamorim | I found one, rfc1321 |
| 16:25.22 | *** join/#asterisk spicyramen_ (~Adium@173.227.7.2) |
| 16:41.03 | *** join/#asterisk TechBob (~John@203.215.117.43) |
| 16:42.47 | TechBob | anyone one? |
| 16:42.54 | TechBob | on* |
| 16:43.05 | [TK]D-Fender | possibly |
| 16:43.07 | pjensen00 | no |
| 16:43.10 | TechBob | damn |
| 16:43.13 | [TK]D-Fender | Guess not |
| 16:43.17 | pjensen00 | sorry, try #* |
| 16:43.32 | TechBob | lol |
| 16:43.37 | ChannelZ | There are currently 212 people not here. |
| 16:43.41 | ChannelZ | Well, 211 minus you. |
| 16:43.48 | TechBob | :D |
| 16:43.53 | [TK]D-Fender | and minus a few bots |
| 16:44.00 | pjensen00 | I've never been so happy to not exist. |
| 16:44.20 | TechBob | i need help with playing background music on a meetme conference room. can't seem to get it working. |
| 16:45.04 | [TK]D-Fender | Show us how you're trying to do it and what happens |
| 16:45.06 | TechBob | hopefully AI has matured enough to answer me a not-so-trivial question. |
| 16:47.20 | TechBob | okay so I set up a dialplan |
| 16:47.35 | TechBob | [local] |
| 16:47.35 | TechBob | include => sounds |
| 16:47.35 | TechBob | [sounds] |
| 16:47.35 | TechBob | exten => 1,1,meetme(8600051) |
| 16:47.35 | TechBob | exten => 1,n,Playback(start-of-meeting) |
| 16:47.37 | TechBob | exten => 1,n,NoOp(test) |
| 16:47.40 | pjensen00 | post it to pastebin next time |
| 16:47.48 | TechBob | oh okay, hold on. |
| 16:47.51 | pjensen00 | thanks |
| 16:48.41 | TechBob | http://pastebin.com/HFsQeDYe |
| 16:50.07 | [TK]D-Fender | that just isn't going to work no matter what |
| 16:50.14 | [TK]D-Fender | you have misunderstood your dialplan mbasics |
| 16:50.24 | [TK]D-Fender | you SIT in that meetme... dialplan does not conitue while you sit in there |
| 16:50.31 | [TK]D-Fender | continue* |
| 16:51.46 | TechBob | that's what's happening it seems. |
| 16:51.54 | [TK]D-Fender | it'll only execute when you leave by a means other than hanging up the call |
| 16:52.15 | [TK]D-Fender | dialplan processing is linear |
| 16:52.48 | TechBob | So, if that's the case, I transfer the call to a neww conference, then transfer it back again once done with playback? |
| 16:53.45 | TechBob | like create a new conference room, move the current one to that new one, play the audio file, switch back to the old conference room? |
| 16:53.56 | TechBob | or something like that? |
| 16:54.10 | [TK]D-Fender | none of this happen in a linear process |
| 16:54.36 | [TK]D-Fender | if you want to play that audio into a conference then you're going to have a new channel join it. |
| 16:54.59 | [TK]D-Fender | This would involve doing an Originate/call-file with a local channel to join and then execute the playback, etc |
| 16:56.14 | TechBob | okay, I'll see if I can cook up something that might work... |
| 16:58.12 | TechBob | Thanks for the help. Might get back to the room on that one. might take me a while though. I've been researching and it's been telling me to create call files, which I haven't done before. and no working test cases to modify. |
| 16:58.23 | TechBob | and then run. |
| 17:18.56 | robmal | http://pastebin.com/TG9Nkx0u |
| 17:19.19 | robmal | Here's my go at this problem. |
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| 17:49.15 | Ice_Strike | When I use AgentRequest(1001) to connect with an agent then I can hear a beep sound. After beep sound - We can't hear our conversations, why? |
| 17:50.03 | *** join/#asterisk CeBe1 (~CeBe@x4d099997.dyn.telefonica.de) |
| 17:52.05 | [TK]D-Fender | did you perr a button to ACCEPT the call? |
| 17:52.10 | [TK]D-Fender | Or configure it not to need one? |
| 17:52.58 | Ice_Strike | There is no button to accept call. |
| 17:53.08 | Ice_Strike | Where do I configure that I dont need one |
| 17:53.21 | Ice_Strike | http://pastebin.com/si9GASq8 |
| 17:54.04 | Ice_Strike | Dialed 200 -> I hear Agents Login. On other phone I dailed 300 and agent can hear the beep. |
| 17:54.23 | Ice_Strike | After that is silent |
| 17:55.27 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/13/configs/samples/agents.conf.sample?revision=420494&view=markup |
| 17:59.52 | Ice_Strike | ackcall=no |
| 17:59.59 | Ice_Strike | That fixed it :D |
| 18:00.00 | Ice_Strike | Thanks |
| 18:00.31 | [TK]D-Fender | Read the docs |
| 18:00.34 | [TK]D-Fender | read the sample configs |
| 18:00.36 | [TK]D-Fender | seriously |
| 18:00.46 | *** join/#asterisk ghoti (~paul@hq.experiencepoint.com) |
| 18:00.46 | [TK]D-Fender | you aren't doing the basic work for this |
| 18:00.48 | Ice_Strike | Sure boss |
| 18:04.28 | Ice_Strike | Catch you later, im going to gym! |
| 18:07.36 | Ice_Strike | Doc for new app_agent_pool is really lacking |
| 18:08.12 | Ice_Strike | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_AgentLogin |
| 18:08.21 | Ice_Strike | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_AgentRequest |
| 18:08.28 | Ice_Strike | I have been reading them two |
| 18:08.50 | Ice_Strike | and configs |
| 18:09.05 | [TK]D-Fender | ackcall has always been a part of agentlogin |
| 18:09.12 | [TK]D-Fender | and the config is right there |
| 18:09.24 | [TK]D-Fender | samples = documentation |
| 18:12.38 | *** join/#asterisk drathir (~kamiljk8@unaffiliated/drathir) |
| 18:13.25 | *** join/#asterisk airjump (~Thunderbi@p5B0A374D.dip0.t-ipconnect.de) |
| 18:13.26 | rmudgett | CLI: "config show help app_agent_pool" |
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| 18:29.46 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 18:59.00 | cj | http://paste2.org/tE8N6JIg <- why doesn't 4012 ring when it hits line 8 ? |
| 19:00.33 | [TK]D-Fender | "sip set debug on" <- you'll have to actuall look at the call out to it |
| 19:00.59 | cj | oh, right. thanks |
| 19:10.45 | cj | I'm getting a 404 from the SIP phone itself. I guess it doesn't recognize itself as sip:4012@98.125.225.13 |
| 19:11.27 | cj | but the SIP INVITE makes it all the way to the end device... the device doesn't recognize itself by the name I've given it, though. |
| 19:29.01 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 19:36.00 | tompaw | Hello |
| 19:36.36 | tompaw | So, I have a setup where I use one shared postgresql for aors/contacts storage used by multiple asterisk boxes, as shown here on this awesome chart: |
| 19:36.38 | tompaw | http://tompaw.net/astbridge.png |
| 19:37.24 | tompaw | The "active" accounts used by the agents (via various ast-boxes) work perfectly fine, however we have a couple of regular phones that idle most of the time. |
| 19:37.57 | tompaw | And I noticed that some asterisks (cannot yet tell which ones, if their "home" boxes or another ones) deregister them by doing a DELETE FROM ps_contacts... |
| 19:38.19 | tompaw | Which parameter would be responsible for this? |
| 19:40.46 | tompaw | ...also - reducing the "Subscribe Period" on the phones should help, shouldn't it? |
| 19:42.36 | robmal | Try enabling keepalive on the phones. |
| 19:44.53 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:45.08 | WIMPy | Do you let them register on ANY server? |
| 19:45.08 | tompaw | robmal: that's what I was thinking - maybe they were sent too rarely. |
| 19:45.11 | tompaw | WIMPy: yes |
| 19:46.20 | WIMPy | That must be the one that received the previous register that has expired. |
| 19:46.37 | tompaw | Reducing the period to 60 seconds from 1800, will see if that helps. |
| 19:46.52 | robmal | Anything below 360 should. |
| 19:47.12 | tompaw | It's just that - as long as that one box was a standalone peer (i.e. had exclusive access to the db), this didn't happen. |
| 19:48.25 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 19:48.54 | WIMPy | It's not the number of Asterisks that access the DB, but the number of Asterisks the phone can pick. |
| 19:49.24 | WIMPy | Registrations are only valid for a given time. |
| 19:49.33 | tompaw | Got it. |
| 19:49.37 | tompaw | OK, well let's see. |
| 19:49.44 | WIMPy | If that's over it will be removed. |
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