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| 00:50.29 | jeffspeff | i'm installing * in vmware esxi virtual machine of centos. after i run ./configure i get the following. any ideas why it doesn't see what OS i'm running? http://pastebin.com/KyjjB7B5 |
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| 01:18.52 | j4jackj | luldick |
| 01:19.02 | j4jackj | I bore of life |
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| 02:59.16 | Mango45 | When on a call, I receive an INVITE from my service provider at exactly 16 minutes. Then the call drops. |
| 02:59.48 | WIMPy | Sounds like a session-timer. |
| 02:59.48 | Mango45 | SIP Debug says: "Peer audio RTP is at port xx.xx.xx.xx:48388". However, that is the IP address of the SIP switch and not the media gateway. |
| 03:00.03 | Mango45 | Is that something I have control over or is it strictly on the provider's end? |
| 03:00.41 | WIMPy | Sounds like there's more to it. |
| 03:00.55 | Mango45 | Googles session-timer |
| 03:01.00 | WIMPy | But you can try to play with the session-timer config non the less. |
| 03:01.14 | WIMPy | sip.conf |
| 03:02.07 | Mango45 | Thanks WIMPy. Found my exact problem here: http://pbxinaflash.com/community/index.php?threads/time-limit-dropped-calls.10200/ |
| 03:05.25 | Mango45 | makes the change, makes a test call, and settles in to listen to Hayley Westenra for 15 minutes. |
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| 03:24.18 | Mango45 | Hrm. Apparently I send an invite at 14:55. That seems to work. Then I receive one at 15:30, then things go to crap. |
| 03:24.32 | Mango45 | I still sent the invite even though Session Expires is set to 10800 secs. |
| 03:24.33 | j4jackj | xX |
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| 05:04.44 | FuriousGeorge | hey all |
| 05:05.20 | Mango45 | FuriousGeorge, are you Canadian by any chance? |
| 05:06.00 | FuriousGeorge | reading up on cisco 5XXg provisioning, and it says "Finally, really important for multiple phones on the same local network, they must transmit out through different ports, start with 5060 and work your way up. i.e. 5060, 5061 etc for each phone you setup." Is that right? I've never had to change that setting before |
| 05:06.30 | FuriousGeorge | Mango45: no, sorry. are you a 45 year old fruit bearing tree by any chance? ;) |
| 05:07.01 | Penguin | No, that is not right. Every phone can use 5060. |
| 05:07.02 | Mango45 | Some routers will do this for you automatically. If all your phones are set to 5060 you'll see one registers to an external server on 5060 and the rest have arbitrary port numbers. |
| 05:07.39 | Mango45 | And if the Asterisk server is on the same subnet, then they will all appear as 5060 and work perfectly. |
| 05:07.48 | Mango45 | You have a popular nickname. And maybe I am. :) |
| 05:09.49 | FuriousGeorge | Thanks for the info. I'll never eat a mango again |
| 05:09.52 | FuriousGeorge | :P |
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| 05:13.00 | snadge | what does status unmonitored mean? |
| 05:13.30 | snadge | im pretty sure I know what OK and UNREACHABLE means |
| 05:15.37 | Penguin | unmonitored means it is not monitored using qualify. No checks are being done. |
| 05:17.04 | ChannelZ | It means you can run around naked and pee in the back yard |
| 05:17.46 | FuriousGeorge | ChannelZ: and so do OK and UNREACHABLE |
| 05:18.26 | ChannelZ | Unreachable is when you're running away, giggling, from the cops |
| 05:19.50 | snadge | im doing level 1 tickets at the moment.. and it's at this point I realise my lack of knowledge/experience is a limiting factor ;) |
| 05:20.14 | FuriousGeorge | he knows not and knows he knows not is a student |
| 05:22.01 | snadge | im starting to realise though.. 99% of level 1 support, is to just confuse the customer and close the ticket |
| 05:22.15 | snadge | unless its easy.. then you just do what they are asking for because it is simple |
| 05:22.41 | snadge | i guess thats why they're getting me to look at level 1 stuff |
| 05:33.44 | snadge | also in show peers.. where it says port.. is that the source port or the destination port? |
| 05:34.25 | snadge | i can see one phone registered on 5061, another on 5062.. and the unmonitored one, which i think is the one they're having problems with.. is on port 1055 |
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| 06:21.11 | DruidZ | Hmmm. I have two that are on port 1024 but they work just fine. |
| 06:21.36 | snadge | would you believe that one of our level 1 techs had the answer |
| 06:21.46 | snadge | you can't beat experience sometimes |
| 06:22.21 | snadge | a couple of the managed pbxes we have.. dont set a few options by default.. like they're supposed to.. one of them is NAT (should be enabled), and the other is Qualify.. im not even sure what that does |
| 06:22.26 | snadge | but setting it to 3000 seems to be a good thing ;) |
| 06:22.33 | DruidZ | So what's the answer? |
| 06:22.58 | snadge | basically i've enabled the NAT option.. told the customer to remove the stun settings.. and it should be right now |
| 06:23.33 | snadge | the customer had to periodically log into the phone.. and click "confirm" on the account settings (without changing anything) to fix a "bad auth" problem.. that we weren't seeing in our server logs |
| 06:25.02 | DruidZ | What did you set NAT to? |
| 06:29.36 | DruidZ | nat=force_rport,comedia is what I have. |
| 06:36.01 | snadge | i set nat to yes ;) |
| 06:36.10 | snadge | which is force_rport |
| 06:36.43 | snadge | ok so why would you set qualify to yes, no, or a number? |
| 06:36.52 | snadge | i've learned that qualify seems to be a good thing |
| 06:47.27 | kaldemar | qualify is the "monitor" you see in sip show peers. no means unmonitored, yes means that asterisk sends qualify packets to the peer, and a number like 3000 enables it with a value in milliseconds, the default being 2000. |
| 06:48.09 | kaldemar | the value is the time within which the peer should answer the qualify message before asterisk thinks it is unreachable. |
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| 06:53.16 | snadge | excellent |
| 06:53.30 | snadge | so what does qualify = yes mean? |
| 06:53.52 | snadge | err i mean.. what is the default unreachable time? |
| 06:54.35 | snadge | 2000... my bad.. thanks kaldemar |
| 06:58.04 | snadge | and apparently by default a qualify (options) packet is sent every 60 seconds |
| 06:58.15 | snadge | which is good for NAT, because it keeps the connection open |
| 07:00.04 | kaldemar | that is controlled by qualifyfreq |
| 07:01.16 | *** join/#asterisk morenoh159 (~morenoh14@173-228-123-147.dsl.dynamic.sonic.net) |
| 07:01.41 | morenoh159 | should I use asterisk if I don't actually have to interface with a telephone number? |
| 07:02.04 | morenoh159 | I don't want to use it if I don't have to and I'm not 100% clear on what asterisk is |
| 07:03.02 | morsing | morenoh159: What exactly are you trying to do? |
| 07:04.04 | morenoh159 | I want to make a device I can connect to my laptop and have it listen to a webservice that broadcasts short messages |
| 07:04.28 | morenoh159 | I think I can get away with a rails or node server that broadcasts the messages though |
| 07:06.04 | morenoh159 | but I also want the listener to be able to establish a 'phonecall' with the messager if desired, hence the interest in asterisk |
| 07:08.37 | morenoh159 | morsing: - |
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| 07:14.06 | arysh1111 | Hi Everyone |
| 07:14.23 | morenoh159 | hey |
| 07:14.31 | arysh1111 | hows you doing? |
| 07:14.58 | morenoh159 | ok learning about asterisk |
| 07:15.09 | arysh1111 | Great, me too. |
| 07:15.19 | morenoh159 | what are you using it for? |
| 07:15.30 | morsing | morenoh159: I don't thnk you'd need Asterisk for that |
| 07:15.46 | arysh1111 | for PBX and few custom call scripts |
| 07:16.19 | morsing | I'm new to Asterisk as well - got it to register with my provider, but it keeps rejecting my handsets |
| 07:16.29 | morsing | Still trying to configure it up... |
| 07:16.44 | arysh1111 | I am stuck with getting an asterisk variable for incoming dahdi channel, could anyone please help me? |
| 07:16.58 | morenoh159 | morsing: an interesting thought though, with voip/asterisk you can establish a communication channel directly between the users correct? How would I do that with a nodejs server? I think if I don't use asterisk the server would be relay the info, no? |
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| 07:17.38 | morsing | morenoh159: But Asterisk would just do what your client would do anyway? |
| 07:18.02 | morsing | Maybe I don't quite understand your question... |
| 07:19.33 | morenoh159 | I guess I'm not sure what asterisk is. with webrtc you establish a 'direct' connection between peers. same as voip I would assume. Is there a way to use nodejs so that it isn't repeating the bits? |
| 07:20.05 | morenoh159 | or does asterisk repeat the bits in this sense as well? |
| 07:21.48 | morenoh159 | or more succintly can asterisk help make a p2p connection? |
| 07:21.53 | morenoh159 | I'll just google that |
| 07:28.35 | arysh1111 | <PROTECTED> |
| 07:36.20 | ChannelZ | Can you elaborate? I have no idea what you mean. |
| 07:39.44 | morsing | morenoh159: Asterisk is a PBX - it just allows you to have many extensions for a phone number plus a lot of other fancy stuff. Really don't think it will help with what you're trying to do |
| 07:45.12 | arysh1111 | ChannelZ: yes sure. |
| 07:46.23 | arysh1111 | [Mar 18 12:29:47] DEBUG[24736] dsp.c: dsp busy pattern set to 0,0 |
| 07:46.24 | arysh1111 | [Mar 18 12:29:47] VERBOSE[24736] logger.c: -- Accepting call from 'XXXXXXXXXX' to 's' on channel 0/2, span 1 |
| 07:46.24 | arysh1111 | [Mar 18 12:29:47] VERBOSE[13697] logger.c: -- Executing [s@from-pstn:1] Set("DAHDI/2-1", "__FROM_DID=s") in new stack |
| 07:46.39 | arysh1111 | here is the Asterisk CLI verbose when we get an incoming dahdi call |
| 07:46.52 | arysh1111 | See this line: Accepting call from 'XXXXXXXXXX' to 's' on channel 0/2, span 1 |
| 07:47.05 | arysh1111 | in verbose it shows span 1 which is span ID |
| 07:47.32 | arysh1111 | I would like to know if there is any variable in asterisk which i can use to get span ID of channel? |
| 07:48.08 | ChannelZ | ${CHANNEL(dahdi_span)} perhaps |
| 07:48.51 | ChannelZ | see 'core show function CHANNEL' |
| 07:49.05 | arysh1111 | okay sure, let me check |
| 07:50.27 | arysh1111 | Great, I hope that should work. System is not accessible right now but i will try it out. |
| 07:50.39 | arysh1111 | ChannelZ: Thank you very much for the hint :) |
| 07:50.50 | ChannelZ | Sure, good luck. I'm off to bed. |
| 07:51.10 | arysh1111 | Sure. Thanks for your time. Have some good sleep sir. Good night :) |
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| 10:55.09 | vader_ | hi can anybode help me out. we are trying to connect to asterisk via websockets and we get an 488 error: data: "SIP/2.0 488 Not acceptable here |
| 10:55.09 | vader_ | ↵Via: SIP/2.0/WS 3g39kjd1m133.invalid;branch=z9hG4bK9684259; |
| 10:56.57 | vader_ | asterisk 11.8.1 |
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| 11:04.01 | alami | \j cisco |
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| 11:25.21 | zapphir | Hi! Is it possible for agents to login and then hangup their phone and then recieve calls from a queue? (so that Asterisk calls the agents physical phone when a call gets assigned to the agent). |
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| 12:14.56 | brittel | Does anyone know any services which can interface with asterisk where i can offer my clients top up on many different networks? |
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| 12:15.44 | brittel | like sending them an sms with a voucher code which they can redeem on there local network to get free topic |
| 12:15.47 | brittel | topup* |
| 12:21.09 | [TK]D-Fender | huh? |
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| 12:38.38 | mirela666 | Hi, on asterisk 1.6.2, I have issue of cli command not executing |
| 12:39.20 | WIMPy | ~upgrade asterisk |
| 12:39.21 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 12:39.52 | mirela666 | it is working after the restart for 1-2 times (dialplan reload for example) then third time it displays nothing and after few seconds it frees |
| 12:40.54 | WIMPy | Don't waste your time on 1.6.2.whatever. |
| 12:41.10 | mirela666 | can't for now ( |
| 12:41.24 | mirela666 | but I do agree that upgrade is needed |
| 12:42.42 | Chainsaw | mirela666: Other then our sympathies... no assistance will be forthcoming. It is too old, and even if it had bugs it will not be fixed. |
| 12:43.10 | mirela666 | true |
| 12:43.15 | Chainsaw | mirela666: You need to be on either 1.8 or 11, and I would recommend 11 so that we don't have to hassle you again soon. |
| 12:43.31 | Chainsaw | mirela666: Because 1.8 doesn't have long. |
| 12:43.44 | mirela666 | ) |
| 12:44.16 | mirela666 | wierd thing is that I have 20-30 boxes on 1.6.2 and this one is the only one with problems |
| 12:44.28 | mirela666 | looks like I'll have to roll out new one :D |
| 12:45.55 | Chainsaw | mirela666: If you want a guess, I will make one. |
| 12:46.14 | Chainsaw | mirela666: You have DNS delays. |
| 12:46.35 | Chainsaw | mirela666: Asterisk will not cope with that at all. And no, DNS delay on one peer will hold up *all* peers. |
| 12:47.05 | Chainsaw | mirela666: Consider hosting a local BIND instance, and remove configuration for any peer that is currently unresponsive. |
| 12:48.06 | mirela666 | aha |
| 12:48.30 | mirela666 | thx, i'll try that |
| 12:48.48 | Chainsaw | mirela666: You're welcome. But seriously, upgrade to 11 as soon as you can. There is no way 1.6.2 is still secure. |
| 12:52.08 | mirela666 | Chainsaw: thanks for the tips \o/ |
| 12:56.37 | Chainsaw | tips hat and disappears into Kings Cross station |
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| 13:10.09 | toresbe | Hey folks! I have a semi-OT question about telephony hardware. I have a channel bank that I need to attach to an E1 card. The E1 card has the standard 8-pin modular jack - that's RJ48, right? |
| 13:11.01 | WIMPy | Probably. |
| 13:11.07 | eirirs | toresbe: o/ |
| 13:11.23 | toresbe | eirirs: wow, hi. LTNS :) |
| 13:11.27 | WIMPy | I've seen different and mostly vague descriptions of "RJ-48". |
| 13:11.56 | WIMPy | But the cards have a standard PRI S2 pin-out. |
| 13:11.59 | toresbe | The channel bank has two connectors I've never seen before - one is DA15, the other is ...some miniaturized coaxial connector, I think I've seen E1 drops wired with those before. |
| 13:13.45 | toresbe | Is the DA15 one a standard? Because I've Googled some pinouts of a "standard" for hooking E1 to DA15, but not sure it's standard enough that I could reasonably expect it to work |
| 13:14.13 | WIMPy | No. Standard is the 8P8C modular plug. |
| 13:15.22 | toresbe | Hrm. I'm excited to see whether this will work. |
| 13:16.29 | toresbe | eirirs: Just got a DAHDI FXS in the mail, BTW. Finally I can use Oslo rotary-dial telephones with Asterisk. It's just what I've been missing in life ;P |
| 13:16.37 | *** join/#asterisk alami (~ialami@unaffiliated/alami) |
| 13:17.46 | eirirs | toresbe: \o/, TV calling games next? Hugo? :D |
| 13:18.09 | eirirs | bets toresbe got über rotary-dialling skills. |
| 13:18.17 | toresbe | eirirs: Oh, for Frikanalen? |
| 13:18.34 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 13:18.42 | toresbe | that's not a bad idea, we've been throwing it around. But the first stupid anachronism we're going for is Teletext ;) |
| 13:19.03 | toresbe | we're going to do so much stupid shit with that when we get it up. |
| 13:19.25 | eirirs | You got SIP trunks to play with it? |
| 13:19.50 | toresbe | I have my personal one, but I'd have to get some more if we were making it a public service. |
| 13:20.29 | alami | i want to make some SIP conference, and i'm confused what i should use? debian with asterisk or elastix or freepbx, can any one help |
| 13:20.49 | toresbe | eirirs: the main problem is that the Oslo dial has reverse digits. In most other places except Sweden and NZ, the ascending order of pulses dials 1234567890, but in Oslo it's 9876543210. So if you get one of the rare ATAs that do support loop-disconnect dialling, you don't get one that supports the Oslo dial. :P |
| 13:21.16 | eirirs | toresbe: or get outside Oslo :P |
| 13:21.18 | toresbe | eirirs: but with the DAHDI card I think I'm going to be able to accomplish it by modding asterisk |
| 13:21.47 | toresbe | eirirs: true, but I have a very cool-looking phone which I'm pretty sure was exclusive to Oslo Telefonanlegg |
| 13:21.53 | toresbe | actually it says "Telegrafverket" ;) |
| 13:22.52 | toresbe | eirirs: btw, I dunno if you still hang out at Ping, but if you do happen to see any telephony kit going out because UiO is going to VoIP, please do let me know! |
| 13:23.42 | eirirs | toresbe: I found a such phone once - http://roynormann.files.wordpress.com/2013/03/tele2.jpg |
| 13:23.48 | eirirs | mint condition! |
| 13:24.04 | eirirs | and, no, I'm living in Sandnes atm |
| 13:24.20 | toresbe | Oh wow, that's even older than mine. Those are cool. |
| 13:24.21 | napnap | WIMPy, you say standart is PRI S2 pinout, I do connect BRI lines to my B400P card, this card expect PRI S2 pinout ? |
| 13:24.38 | eirirs | toresbe: I didnt save it, think they gave it to a museum. |
| 13:24.46 | toresbe | eirirs: ah, good |
| 13:24.57 | toresbe | eirirs: this is all setting up for The Gathering, btw. |
| 13:25.11 | eirirs | <-- the only Gathering I've been at is this one in Stavanger in '94. |
| 13:25.16 | eirirs | For one-day visit only. |
| 13:25.32 | eirirs | I'll rather go skiing these days :P |
| 13:25.35 | toresbe | ah. Well, we're doing a really fun thing where we're integrating our mobile radios with the VoIP network. |
| 13:26.09 | toresbe | Turns out CUCM can let you use Cisco hardphones as walkie-talkies, creating a push-to-talk button on the terminal. |
| 13:26.20 | eirirs | :D |
| 13:26.26 | eirirs | conference-mode? |
| 13:27.02 | toresbe | It's a special mode afaik |
| 13:29.32 | WIMPy | napnap: No. That's S0 and different. See http://voice.yeti.dk/Asterisk_vs_ISDN/7 |
| 13:30.46 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 13:31.41 | napnap | WIMPy, oh cool, I search that since a long time..but I see I've a good wires order...so it is not the cable. |
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| 13:33.41 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 13:36.21 | hrnt | toresbe: uhm, isn't that a pretty standard feature in voip phones? (push-to-talk) |
| 13:36.27 | hrnt | (or did i miss something obvious) :) |
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| 13:47.48 | napnap | WIMPy, on the website I see "Unless you only want to connect a PBX this setup usually involves operating the NT interface in ptmp mode." So, me I want to connect 2 BRI lines from my national provider..I do set NT mode right ? |
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| 13:50.15 | napnap | I understood the opposite, so I do change jumper of my card :-O |
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| 13:53.01 | toresbe | hrnt: I'm not aware of that, but it might be; the big deal is that when PTT is received, CUCM and the channel unit will actually key a radio transmitter, so you can carry a conversation on a simplex medium. |
| 14:06.57 | napnap | oh no...I did not understand this sentence, I do keep te mode to bri from my national provider..back to the begining |
| 14:06.59 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 14:09.55 | *** join/#asterisk fireman_biff (~biff@208.0.98.13) |
| 14:11.19 | napnap | and no...according to http://www.asteriskguru.com/tutorials/bri.html NT mode : interface between an ISDN uuser and ISDN provider, so I need NT mode ! |
| 14:17.20 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 14:19.22 | qakhan | Sip provide gave me 10 DIDs 2847801 - 11 and i should send same DID number with SIP:2847801@ip to make outbound call |
| 14:19.38 | qakhan | i also setup 801 ext for my phone |
| 14:19.55 | WIMPy | napnap: No, that was only referring to needing ptmp in the NT-mode case. To connect a line, you want TE-mode. |
| 14:20.56 | qakhan | is there any way i use on 801 - 811 ext but when i make outbound call 2847 attache on SIP:801@ip |
| 14:21.14 | WIMPy | N = Network, T = Terminal. You want to be a terminal. |
| 14:22.29 | WIMPy | qakhan: I'm sure you already know the answer yourself. 1x application Set and 2x function CALLERID. |
| 14:25.54 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 14:28.28 | *** join/#asterisk aness (~aness@cm-84.215.76.18.getinternet.no) |
| 14:28.54 | qakhan | WIMPy it is attaching my ext 801 in SIP:801@IP i want to add full DID number 2847801 in SIP:801@ip |
| 14:29.33 | WIMPy | That hint wasn't good enough? |
| 14:29.37 | qakhan | my sip provide will not accept call and thats way i was getting messgae 484 "Address Incomplete" back from ip |
| 14:30.02 | WIMPy | CALLERID(num)=2847${CALLERID(num)} |
| 14:30.36 | qakhan | yesterday i was talking to [TK]D-Fender and he was right my sip provide is not liking what i am sending to them |
| 14:31.47 | [TK]D-Fender | [10:19]qakhanSip provide gave me 10 DIDs 2847801 - 11 and i should send same DID number with SIP:2847801@ip to make outbound call <- nowhere in this do I see the number you are dialing. |
| 14:32.49 | qakhan | here is my teacher :P |
| 14:32.58 | qakhan | let me send you |
| 14:33.28 | [TK]D-Fender | qakhan: You shouldn't wait before sending backup. |
| 14:35.47 | qakhan | http://pastebin.com/bNj6B8Bf |
| 14:37.00 | *** join/#asterisk ghost75 (~quassel@dslb-088-064-055-121.pools.arcor-ip.net) |
| 14:37.15 | [TK]D-Fender | qakhan: And....? |
| 14:37.21 | napnap | WIMPy, sorry of my understanding, but I have 2 BRI which came from my national ISP , I don't need ptmp ?? You say I've need to use TE mode without ptmp it's ok ? |
| 14:37.40 | [TK]D-Fender | qakhan: Looking for 0 in from-internal (domain 192.168.1.250) SIP/2.0 484 Address Incomplete |
| 14:37.48 | [TK]D-Fender | qakhan: They don't like what you dialed.... |
| 14:38.11 | [TK]D-Fender | qakhan: And just by looking at it I'm not surprised |
| 14:38.43 | fireman_biff | Hi all, I need a PHP script that can show me all received calls and which extension was involved (whether it passed through a ring group, queue, got transfered, etc). I have an old script I wrote which looks at CDR records, but its inadequate and I was thinking to rewrite it using CEL instead. Is that the current best practice? Is there something else I can use? (I'm using Asterisk 1.8) |
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| 14:38.58 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:39.01 | WIMPy | napnap: To connect to a line (the PSTN) you need to be in TE mode. That line can be ptp or ptmp, depending on what you ordered. |
| 14:39.29 | WIMPy | napnap: Usually configuring ptp is safe, even if the line is ptmp. |
| 14:39.30 | napnap | WIMPy, ok, thanks. |
| 14:40.52 | qakhan | [TK]D-Fender i know that the reason is i am not passing complete DID number 2847801 in SIP:801@IP |
| 14:41.08 | [TK]D-Fender | qakhan: no |
| 14:41.20 | [TK]D-Fender | you care calling OUT. DID is ***INBOUND*** |
| 14:41.26 | [TK]D-Fender | And you are dialing a BAD NUMBER |
| 14:41.26 | qakhan | i have ext 801 it does not work but if i make ext 2847801 and then make outbound call them it works |
| 14:41.29 | [TK]D-Fender | You dialed "0" |
| 14:41.33 | [TK]D-Fender | JUST "0" |
| 14:41.39 | [TK]D-Fender | no other digits |
| 14:41.50 | qakhan | i meant to say DOD |
| 14:41.59 | [TK]D-Fender | DOD is not a term |
| 14:42.02 | Penguin | hahaha |
| 14:42.05 | Penguin | ~dod |
| 14:42.05 | infobot | rumour has it, dod is Dial on Demand, a way of making the modem dial whenever you try and send info to the inet |
| 14:42.06 | [TK]D-Fender | (in telecom) for this |
| 14:42.30 | [TK]D-Fender | Department Of Defense - Yes... someone SHOULD be shot.... |
| 14:42.45 | qakhan | hahah |
| 14:43.18 | qakhan | i have ext 801 it does not work but if i make ext 2847801 and then make outbound call them it works |
| 14:43.39 | [TK]D-Fender | qakhan: You dialed a number I would never believe ANYONE would ever accept |
| 14:43.45 | Penguin | DOD... that's great. If you say DID and someone tells you that's the wrong direction, it must be switched to DOD. |
| 14:43.50 | [TK]D-Fender | qakhan: And you are loking at HALF of the picture once again. |
| 14:44.15 | qakhan | From: "Ahamed" <sip:2847805@172.29.44.242>;tag=as62ec76e2 |
| 14:44.16 | qakhan | To: <sip:0537707501@10.200.7.157> |
| 14:44.28 | qakhan | check this |
| 14:44.35 | qakhan | sip:2847805@172.29.44.242 |
| 14:44.40 | ghost75 | patch was already there but deleted haha https://issues.asterisk.org/jira/browse/ASTERISK-20841 |
| 14:44.44 | [TK]D-Fender | show COMPLETE debug, not garbage little pieces |
| 14:44.52 | qakhan | i sent you |
| 14:44.59 | qakhan | http://pastebin.com/bNj6B8Bf |
| 14:45.21 | [TK]D-Fender | <--- SIP read from UDP:192.168.1.182:5060 ---> INVITE sip:0@192.168.1.250 SIP/2.0 <---------- THIS is what you dialed |
| 14:45.27 | [TK]D-Fender | "0" |
| 14:46.12 | [TK]D-Fender | And I see another call further on.... |
| 14:46.28 | [TK]D-Fender | Perhaps that's where you're going with this (trim your output) |
| 14:47.03 | jwr__ | Is anyone running asterisk/freepbx, with a pci analog card, in a virtual machine? some googling tells me that it will be echo city, but those results are a few years old. anyone have experience with that? |
| 14:48.15 | [TK]D-Fender | qakhan: So further down I se ANOTHER call that goes out, hits progress, and then gets hung up by your originating phone |
| 14:49.05 | *** join/#asterisk banane_ (528bc51a@gateway/web/freenode/ip.82.139.197.26) |
| 14:49.07 | qakhan | [TK]D-Fender please ignore previos pastebin please check this one |
| 14:49.08 | qakhan | http://pastebin.com/nA7AQtsX |
| 14:49.54 | qakhan | if call go through like this "sip:2847805@172.29.44.242" it works fine |
| 14:50.11 | banane_ | hey guys, does anyone know if it is possible to create a dial-out rule for a special sip peer? i want one user to dial out via isdn and all other users to dial out via sip |
| 14:50.19 | qakhan | but if call go through like this "sip:805@172.29.44.242" it does not work |
| 14:50.36 | WIMPy | banane_: Contexts |
| 14:50.41 | [TK]D-Fender | qakhan: because that is the USER you are identifying as |
| 14:50.50 | [TK]D-Fender | qakhan: This is perfectly normal |
| 14:50.52 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 14:51.20 | [TK]D-Fender | qakhan: When your PBX calls out your phone's callerid is NOT the account you have with your provider |
| 14:51.35 | qakhan | yes now you got me |
| 14:51.56 | qakhan | so how i can add 2847 with 805 when call out |
| 14:51.57 | banane_ | WIMPy: so i have to put the single peer into it´s own context? |
| 14:52.15 | Penguin | its |
| 14:52.19 | [TK]D-Fender | banane_: Certainly one way. Or make better patterns in the same one to match that caller |
| 14:52.32 | WIMPy | banane_: you don't _have to_, but that would be an obvious way to do it. |
| 14:52.46 | [TK]D-Fender | banane_: I'd recommend separating contexts... that's what they're there for |
| 14:53.03 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 14:53.32 | [TK]D-Fender | [10:50]qakhanbut if call go through like this "sip:805@172.29.44.242" it does not work <- so set it in your peer so that * doesn't leave the callerid there. |
| 14:53.33 | banane_ | then i´ll do it that way thanks |
| 14:54.07 | qakhan | how? |
| 14:54.18 | qakhan | example please |
| 14:54.24 | fireman_biff | jwr__: I have at least one virtualized PBX with an analog card (either pci or pci-e, can't remember) without echo problems. But we're moving away from virtualized PBXs that contain pci cards because the pci passthrough seems to cause problems and the card manufactures tend to not support that setup |
| 14:54.39 | [TK]D-Fender | qakhan: Haven't you compared your 2 peer settings yet? There is an obvious difference between them if is is doing it, and the other isn't |
| 14:54.40 | fireman_biff | jwr__: our cards usually have hwec |
| 15:01.10 | *** join/#asterisk fling (~fling@fsf/member/fling) |
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| 15:04.15 | qakhan | [TK]D-Fender here is my dial plan http://pastebin.com/gzpdDLVf |
| 15:04.46 | [TK]D-Fender | qakhan: I did not ask for that. |
| 15:05.51 | [TK]D-Fender | exten => _X.,1,Set(CALLERID(name)=2874${CALLERID(num)}) <- what do you think setting the NAME will do here? |
| 15:06.20 | WIMPy | That was not the line I gave. |
| 15:06.31 | Penguin | OH! OH! ME... PICK ME... I KNOW!! |
| 15:07.25 | qakhan | it just add caller id 2874805 but not make change in SIP:805@IP |
| 15:07.43 | [TK]D-Fender | qakhan: that is NOT the NAME |
| 15:07.48 | qakhan | <WIMPy> CALLERID(num)=2847${CALLERID(num)} |
| 15:07.59 | Penguin | num is different from name |
| 15:08.07 | qakhan | i am sorry WIMPy |
| 15:08.13 | Penguin | SIGNIFICANTLY |
| 15:09.57 | qakhan | working now |
| 15:10.04 | qakhan | Thank you guys |
| 15:10.53 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-dtaxdrvhjainpsec) |
| 15:14.12 | DruidZ | I have a user exhibiting audio issues. Sounds like NAT issue but he has a public IP and is definitely not behind a NAT. |
| 15:14.41 | WIMPy | NAT issues would typically give no audio at all. |
| 15:14.50 | DruidZ | However, I checked his settings and he had NAT mapping enabled. Could that cause problems? |
| 15:14.53 | [TK]D-Fender | Actual details would help... |
| 15:15.13 | DruidZ | He had one way audio. |
| 15:15.39 | [TK]D-Fender | that sounds like a NAT configuration issue |
| 15:15.47 | [TK]D-Fender | ALG can also screw things up |
| 15:16.25 | DruidZ | I tirned off all NAT settings in his ATA (SPA122) but I can't check it until tonight when he gets home. |
| 15:16.46 | DruidZ | Just wondering if that is likely the issue or should I keep looking. |
| 15:17.10 | [TK]D-Fender | You should ask when you're in a position to test and show debug. Otherwise it's all just guessing. |
| 15:18.07 | DruidZ | Wouldn't ALG cause registration issues? |
| 15:18.34 | WIMPy | Nope. It only mangles SDP. Usually. |
| 15:18.44 | *** join/#asterisk tony2012 (~Anthony@cpe-72-225-9-31.rochester.res.rr.com) |
| 15:19.16 | DruidZ | And would ALG matter if NAT is not involved? |
| 15:20.01 | WIMPy | How would that happen? |
| 15:20.10 | DruidZ | Both the server and the client are on public IPs in this case. |
| 15:20.15 | WIMPy | ALG is somethign NATting gateways do. |
| 15:20.49 | [TK]D-Fender | DruidZ: You just said they're public.. so NAT shouldn't be an issue.... |
| 15:21.01 | [TK]D-Fender | DruidZ: And guessing is a waste. Come back with some debug we can examine |
| 15:21.26 | DruidZ | Exactly. You suggested ALG. I was just pointing out that NAT wasn't involved. |
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| 15:23.50 | Mango45 | I'm having an issue with calls dropping at exactly 15:30. It seems that the Asterisk server at the far end is telling my Asterisk server the wrong audio IP. Is anyone here aware of a workaround for this issue? |
| 15:23.54 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 15:24.05 | banane_ | how can i dial an extension placed in another context? i tried smth like Dial(SIP/${EXTEN]@second-context) but it says "no such host second-context", am i missing something obvious? |
| 15:24.31 | [TK]D-Fender | banane_: You don't |
| 15:24.35 | Mango45 | banane_: I've used Goto(context,${EXTEN},1) |
| 15:24.38 | [TK]D-Fender | banane_: You point the device to the proper context |
| 15:25.03 | WIMPy | banane_: 1st there is a syntaz error ${] and second, your dialplan is not a sip device. |
| 15:25.42 | WIMPy | But there are local channels that let you Dial extensions as if they were devices. |
| 15:26.03 | WIMPy | But I doubt that it is what you want at this point. |
| 15:26.20 | [TK]D-Fender | WIMPy: I'm pretty sure he doesn't need to "dial" anything there and is having basic comprehension issues with the concept of contexts |
| 15:26.34 | banane_ | ok the syntax error is just in this example not in my dialplan, what i want is one user in context-one to be able to call an extension in another context |
| 15:27.53 | banane_ | thanks @mango45 works fine |
| 15:28.01 | Mango45 | Cool beans. |
| 15:28.30 | *** join/#asterisk lampe2 (~lampe2@v2201209138279141.yourvserver.net) |
| 15:28.54 | WIMPy | banane_: Include the context. |
| 15:32.33 | jeffspeff | i'm installing * in vmware esxi virtual machine of centos. after i run ./configure i get the following. any ideas why it doesn't see the correct details? http://pastebin.com/qeNrvXNY |
| 15:33.39 | file | because the configure script didn't pick it up, but there's nothing to worry about |
| 15:36.15 | jeffspeff | i'm having some weird issues after i finish compiling and i'm trying to figure out the root cause. the issue ocurring when i install from my script is that asterisk starts when using service asterisk start but not all modules load. however, when i start asterisk using "asterisk -cv" it starts fine and loads all modules. |
| 15:37.19 | jeffspeff | also, when running configure, i've found that if i don't specify the libdir to be /usr/lib64 then it will install everything to /usr/lib but look for libasteriskssl.so in /usr/lib64 |
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| 16:42.38 | linagee | is 911 a valid NPA? "[2–9] for the first digit, and [0-9] for the second and third digits." |
| 16:43.17 | linagee | I'd like to say no, I just can't find where its specifically disallowed. |
| 16:43.38 | Penguin | NPA? I would say no and that 911 is a service code. |
| 16:43.56 | linagee | Penguin: N11 codes are not allowed as NPAs? |
| 16:45.44 | Penguin | I'm going to say that it is not an area code. |
| 16:45.51 | linagee | that is... weird how all N11s are reserved in the NPA space.... (Even though they would be differentiated. For instance, 911 versus 1-911-) |
| 16:45.58 | Penguin | If NPA is area code, that indicates routing to a particular switch. |
| 16:46.14 | linagee | Penguin: I guess this list proves that it should not be a valid area code/NPA. http://en.wikipedia.org/wiki/List_of_NANP_area_codes#900.E2.80.93999 |
| 16:47.04 | tony2012 | Anyone have any experience with JABBER_STATUS? I'm having a problem where using a bare jid doesn't work, but full jid does. More full description here: http://forums.asterisk.org/viewtopic.php?f=14&t=89748&start=0 |
| 16:48.10 | linagee | probably some crappy phones allow direct 10 digit dialing without the 1+ prefix. |
| 16:48.16 | linagee | http://en.wikipedia.org/wiki/North_American_Numbering_Plan#Dialing_procedures |
| 16:48.25 | Penguin | Regular phones allow it. |
| 16:48.37 | linagee | I've never really tried it. seems lame to not have 1+ |
| 16:48.56 | Penguin | 1 is the country code, and since I'm already in 1 country, I don't need to dial a 1 before the area and number. |
| 16:49.54 | Penguin | This is in the same sense that some places allow 7-digit dialing when calling within the same NPA. |
| 16:50.07 | linagee | this is crazy... "Most areas permit local calls as 1+10D except for Texas, Georgia, and some jurisdictions in Canada which require that landline callers know which numbers are local and which are toll, dialling 10D for local calls and 1+10D for all toll calls." |
| 16:50.21 | linagee | I hope to never live in texas, georgia, and some jurisdictions in canada..... |
| 16:50.37 | linagee | (so most people in texas dial 10D instead of 1+10D? how weird.) |
| 16:50.50 | Penguin | I always dial 7 or 10 digits. |
| 16:51.06 | Penguin | I am in country 1, so I never dial 1+ 10. |
| 16:51.22 | Penguin | It seems extraneous. |
| 16:51.23 | linagee | Penguin: when you're using a mabell phone or voip? |
| 16:51.36 | linagee | mabell dialing plan. (otherwise known as NANP) |
| 16:51.50 | Penguin | Using wireless service or voip services, I dial 10 digits. |
| 16:52.19 | Penguin | On land-based phones, I dial whatever they force me to dial. |
| 16:52.24 | linagee | :) |
| 16:52.42 | Penguin | If I dial 10 digits and they don't like it, they will play a sound file telling me to dial a 1 before my area code and number. |
| 16:53.08 | linagee | and that negative reinforcement doesn't cause you to always dial 1+10D? |
| 16:53.24 | Penguin | Of course not. It's extraneous and generally wrong. |
| 16:53.25 | linagee | or you just think *they* are lame for not allowing direct 10D dialing? hah |
| 16:53.37 | Oggu | Is there an easy way to get set up with Asterisk on Mac OS X? |
| 16:53.46 | Penguin | Just because one service requires the country code does not mean I need to dial it everywhere. |
| 16:54.18 | WIMPy | Oggu: ./configure;make;make install, just as on any other OS. |
| 16:54.44 | Penguin | There are places that require 10-digit dial to call across the street, even when they are using the same NPA. |
| 16:55.48 | linagee | WIMPy / Oggu: or do some homebrew type thing? (I am not a Mac.) |
| 16:55.50 | Penguin | And there are places that have so many different area codes that they require 10-digit dial because of that. But not 11 digits, since it's in the same country. |
| 16:56.01 | Oggu | I tried the homebrew thing. And the make install way |
| 16:56.12 | Oggu | After like 10 tries it sorta works. But no pjsip |
| 16:57.46 | linagee | Penguin: I think its funny in states where the area codes have been fractured from "the main one" and people tie personal feelings/associations to "the main area code" and think everyone else is somehow an outsider. :) |
| 16:57.47 | *** join/#asterisk DruidZ (~darcy@dilbert.druid.net) |
| 16:57.52 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 16:58.05 | linagee | Penguin: "you're not one of us!" :) |
| 16:58.14 | Penguin | I remember when that happened. |
| 16:58.23 | linagee | Penguin: I still see the effects of it |
| 16:58.30 | Penguin | Like when St. Louis needed more than just the 314 area code. |
| 16:58.47 | DruidZ | Now I am trying to get a SIP redirect working. The following works fine: |
| 16:58.47 | DruidZ | exten => *51,1,Dial(SIP/thetestcall@sip.antisip.com,1) ; France |
| 16:58.55 | DruidZ | But this does not: |
| 16:58.55 | DruidZ | exten => 4165551111,1,Dial(SIP/4165551111@sip.voip.ms,1) |
| 16:59.08 | DruidZ | Numbers are sanitized. |
| 16:59.24 | DruidZ | The SIP debug log is at http://www.vex.net/~darcy/voip.ms.failure.txt |
| 16:59.29 | Penguin | druidz: Configure a peer for your ITSP and then dial using the proper syntax. |
| 16:59.45 | linagee | Penguin++ |
| 17:00.09 | linagee | DruidZ: install freepbx and let a gui do all the hard work |
| 17:00.25 | linagee | (I hope that's not a ban-able offense in here. to use the easy way out.) |
| 17:00.30 | navaismo | wat! |
| 17:00.34 | Penguin | It should be. |
| 17:00.47 | DruidZ | Penguin: But I want to avoid the ITSP. That's why I am redirecting. |
| 17:00.56 | navaismo | someone kick that guy for using that kind of suggestion |
| 17:01.20 | Penguin | I don't see anything redirecting. All I see is a Dial to a URI which is that of an ITSP. |
| 17:01.20 | *** part/#asterisk linagee (~linagee@about/linux/regular/linagee) |
| 17:01.28 | *** join/#asterisk linagee (~linagee@about/linux/regular/linagee) |
| 17:01.52 | Penguin | Dial() is not a redirection. |
| 17:02.00 | Penguin | And voip.ms is an ITSP. |
| 17:02.20 | Penguin | So create a peer for the ITSP and then dial using SIP/peername/phonenumber |
| 17:02.50 | DruidZ | Right. What I mean is that I have an extension 4155551111 which tries to dial a SIP address. |
| 17:03.04 | Penguin | And then? |
| 17:03.06 | DruidZ | I didn't have to do that for sip.antisip.com |
| 17:03.07 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 17:03.29 | DruidZ | Why do I have to do it for sip.voip.ms? What's the difference? |
| 17:03.30 | Penguin | Perhaps antisip allows anonymous calls? |
| 17:04.04 | DruidZ | The voip.ms one worked in FS and it works directly in Twinkle. |
| 17:04.44 | linagee | if I have a phone service dependant on voip/my ITSP, is there a good way to test it stays up and running? have something on a hard land line calling it every few hours and record the line and do audio analysis against a known good? |
| 17:04.54 | linagee | (shorter: how do I do end to end testing for a voip service) |
| 17:05.25 | navaismo | calling |
| 17:05.54 | linagee | navaismo: automated though |
| 17:06.11 | navaismo | call files/AMI |
| 17:06.43 | linagee | navaismo: end to end |
| 17:07.02 | navaismo | and? |
| 17:07.05 | Penguin | druidz: If you can make calls to them, I have to assume you are providing the credentials required by them. |
| 17:07.35 | Penguin | druidz: If you then cannot make calls to them on another device, such as asterisk, I have to assume that you are not providing the credentials. |
| 17:08.00 | linagee | navaismo: I can test to make sure my server is up, test to make sure the connection to my ITSP is up, but how can I test beyond that? what if my ITSP shows as a good connection but their PSTN connection is down? |
| 17:08.27 | linagee | navaismo: wait for people to complain is one way, but that kind of sucks. |
| 17:09.10 | Penguin | If your ITSP's gateway to the PSTN is down that much, you need to consider a better ITSP. |
| 17:09.16 | navaismo | If the bridge between the ITSP and PSTN is down then you have shitty ITSP. If that can happen then your ITSP will advice about that issue. |
| 17:09.57 | Penguin | Any reputable ITSP will have multiple routes for redundancy. |
| 17:10.06 | linagee | navaismo: so... I have to put an alert against their blog and hope their update it when something goes wrong? |
| 17:10.23 | linagee | Penguin: are you saying a good ITSP will *never* go down? |
| 17:10.27 | *** join/#asterisk ulogic (421e6b4f@gateway/web/freenode/ip.66.30.107.79) |
| 17:10.31 | Penguin | No, not saying that at all. |
| 17:10.52 | Penguin | But a good ITSP will have very minimal outages because of the redundancy. |
| 17:10.59 | linagee | Penguin: I have conexiant. I thought they were pretty good. And they did go down for a bit a while back. (At least, one time they let me know about.) |
| 17:11.11 | navaismo | if you have a crappy itsp do not expect much about their service, but if you want to stay with that ITSP then use call files and AMD to test the service |
| 17:11.16 | DruidZ | Penguin: I certainly didn't add any credentiols vor voip.ms in my Twinkle. I doubt that it comes with any. |
| 17:11.36 | DruidZ | s/credentiols/credentials/ |
| 17:11.51 | linagee | navaismo: what do you mean by call files? Maybe you were answering me all along and I just didn't understand. Do you mean call them from another point and record what's going on into files? |
| 17:12.03 | navaismo | DruidZ, i have a voip.ms account and it came with credentials |
| 17:12.10 | Penguin | ~call files |
| 17:12.10 | infobot | ACTION looks around and then screams out files as loudly as possible |
| 17:12.15 | Penguin | haha |
| 17:12.28 | navaismo | haha |
| 17:12.30 | Penguin | That's not quite what I was looking for. |
| 17:12.33 | linagee | ~call whats a call file is that a recording of an audio file |
| 17:12.34 | infobot | ACTION looks around and then screams out whats a call file is that a recording of an audio file as loudly as possible |
| 17:12.45 | navaismo | ~call file |
| 17:12.45 | infobot | ACTION looks around and then screams out file as loudly as possible |
| 17:12.51 | file | WHAT |
| 17:12.55 | Penguin | hahaha |
| 17:12.58 | linagee | lol |
| 17:13.13 | navaismo | file your bot is joking around |
| 17:13.18 | navaismo | ~callfile |
| 17:13.19 | infobot | i guess callfile is a text file that when placed in the correct directory makes Asterisk make an outgoing call. See http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out#Callfiles |
| 17:13.38 | file | I hold no power over infobot |
| 17:13.41 | file | infobot, botsnack |
| 17:13.41 | infobot | aw, gee, file |
| 17:13.57 | linagee | ~botsmack |
| 17:13.57 | infobot | OWW! |
| 17:13.59 | [TK]D-Fender | ~areyouadog? |
| 17:13.59 | infobot | Bark! Bark! |
| 17:14.03 | [TK]D-Fender | infobot: Good boy! |
| 17:14.03 | infobot | aw, gee, [TK]D-Fender |
| 17:14.07 | [TK]D-Fender | ~botsnack |
| 17:14.07 | infobot | [TK]D-Fender: :) |
| 17:14.40 | linagee | I can't believe the wiki actually lists this, lol. "Prank call programs." |
| 17:14.48 | linagee | good way to get booted off the PSTN. lol. |
| 17:15.05 | Penguin | IANAL |
| 17:15.13 | navaismo | i wish the stupid frog from freepbx was so cute as the infobot here |
| 17:15.36 | navaismo | Penguin, you what? |
| 17:15.40 | navaismo | O_o |
| 17:15.42 | DruidZ | navaismo: Sure. So does the person I am trying to call. |
| 17:15.50 | Penguin | ~IANAL |
| 17:15.50 | infobot | [ianal] "I am not a lawyer". See also IANALBIPOOTV. |
| 17:15.56 | linagee | not sure what's worse. a prank call or a telemarketer. at least with a prank call, you know they're malicious. with a telemarketer, they're just trying to hide that. also they will take your money. |
| 17:16.44 | navaismo | Penguin, i guess ive seen a lot pr0n in my life to missundertood those words |
| 17:17.59 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:19.08 | napnap | My "first" attempt to make a call at an analog device throught the card TDM400P fail, this is the output of asterisk ( http://pastebin.fr/33109 ) an idea ? |
| 17:20.16 | navaismo | napnap, using freepbx? |
| 17:20.18 | napnap | I use dahdi, I added my opermode to wctdm then... I don't know what to do |
| 17:21.23 | navaismo | napnap, seems like you dialed only 49, do you have outbound routes defined? better move to #freepbx |
| 17:21.47 | [TK]D-Fender | napnap: -- Executing [s@macro-dial-one:37] Dial("SIP/80-00000003", "DAHDI/15,"",tr") in new stack <- you aren't dialing a number at all |
| 17:22.11 | [TK]D-Fender | And have junk quotes in your dial command. This looks like a crappy old version of FreePBX |
| 17:22.23 | napnap | navaismo, I use Elastix, but if I well understand it's above freepbx |
| 17:22.25 | Oggu | How does Playback search for sounds? I tried to give it an absolute path but still couldn't find the file |
| 17:22.42 | navaismo | vomit then eat it again |
| 17:22.49 | [TK]D-Fender | napnap: not "above", they run a hacked up forked version. |
| 17:22.51 | Penguin | "above freepbx" |
| 17:23.09 | napnap | [TK]D-Fender, oh, ok |
| 17:23.16 | [TK]D-Fender | Oggu: Then you did it wrong. Show us. |
| 17:23.20 | navaismo | Oggu, are you using the extension at the end of the sound? |
| 17:23.30 | navaismo | like mysound.wav |
| 17:24.05 | napnap | navaismo, yah only 49 it's an internal analogic device |
| 17:24.15 | Oggu | navaismo: No |
| 17:24.19 | Oggu | [TK]D-Fender: http://pastebin.com/Hk4t0afr |
| 17:24.37 | Penguin | oggu: It typically uses the sounds dir as the relative path, and selects the file extension based on the codec used on the channel. |
| 17:24.49 | Penguin | To play "hello" you use Playback(hello) |
| 17:24.49 | navaismo | napnap, so fxs device |
| 17:24.54 | Oggu | I just started out with asterisk. And trying things out. |
| 17:24.55 | [TK]D-Fender | Oggu: Show us the actual call, and the dump of that full path |
| 17:25.14 | navaismo | maybe the space in the path |
| 17:25.19 | [TK]D-Fender | perhaps |
| 17:25.30 | Oggu | http://pastebin.com/2Y8F3cWv |
| 17:26.04 | Oggu | For some reason the install put the files in appl support |
| 17:26.12 | [TK]D-Fender | " Resource temporarily unavailable" |
| 17:26.15 | Oggu | I can move them |
| 17:26.21 | Penguin | Move them or change the sounds dir. |
| 17:26.34 | [TK]D-Fender | Im wondering if your modules are all loaded |
| 17:26.51 | Penguin | Library should not be hanging off of FS root anyway. /Library should not exist. |
| 17:27.01 | [TK]D-Fender | Penguin: OS-X |
| 17:27.15 | Penguin | throws his hands in the air and goes to get a Hot Pocket |
| 17:27.20 | navaismo | napnap, show us the output of dahdi show channels in asterisk cli and lsdahdi in linux shell |
| 17:27.23 | Oggu | Is it better to just get a virtualbox? |
| 17:27.45 | [TK]D-Fender | Oggu: It's best to install native on "whatever" and do it right |
| 17:27.46 | navaismo | Oggu, depends on your needs |
| 17:28.10 | [TK]D-Fender | Oggu: Go verify your modules have loaded right |
| 17:28.21 | [TK]D-Fender | "module show like format" |
| 17:28.36 | [TK]D-Fender | And show us the full dump of the path & faile |
| 17:29.39 | Penguin | http://i.imgur.com/gLhhRb3.jpg |
| 17:29.48 | napnap | navaismo, http://pastebin.fr/33111 |
| 17:30.19 | Oggu | [TK]D-Fender: module show like format says "0 modules loaded". But I can still connect with SIP.. |
| 17:31.18 | Oggu | But I don't get any errors during startup |
| 17:31.21 | [TK]D-Fender | go check your asterisk.conf and modules.conf |
| 17:31.33 | [TK]D-Fender | You won't get errors for things it's not trying to do... |
| 17:31.45 | Oggu | It says [Mar 20 18:28:52] NOTICE[2567]: loader.c:1323 load_modules: 12 modules will be loaded. |
| 17:31.58 | Oggu | And then it starts loading modules.. |
| 17:32.08 | file | that's not very many modules. |
| 17:32.27 | [TK]D-Fender | Pretty screwed up |
| 17:32.30 | Oggu | I disabled autoload as pjsip broke things |
| 17:33.02 | [TK]D-Fender | Oggu: then you should have simply disabled ONLY that one |
| 17:33.39 | navaismo | napnap, amm you have fxo ports so you connect landlines there |
| 17:33.50 | navaismo | are you trying to send a call to a PSTN number? |
| 17:34.10 | [TK]D-Fender | navaismo: he isn't\ |
| 17:34.15 | [TK]D-Fender | navaismo: which I pointed out. |
| 17:34.43 | navaismo | yea but the he said " yah only 49 it's an internal analogic device" |
| 17:35.06 | navaismo | s/the/then/ |
| 17:35.28 | napnap | navaismo, landlines ? what do you mean ? Yeah I connect only internal lines |
| 17:36.01 | navaismo | ok, as [TK]D-Fender said check your dialed number and check your outbound routes you aren't sending any number |
| 17:36.42 | napnap | navaismo, no I call only internal device throught internal analogic line... |
| 17:36.54 | navaismo | what |
| 17:36.58 | napnap | I use t0 for external call, not this card |
| 17:37.22 | navaismo | well if you want to connect analog phones use FXS modules |
| 17:37.27 | navaismo | not FXO |
| 17:37.38 | napnap | navaismo, erf, I have the wrong card :-s |
| 17:37.50 | navaismo | wrong module* |
| 17:38.11 | napnap | oh ok, module in soft way..ok |
| 17:38.22 | Penguin | hard module |
| 17:38.31 | *** join/#asterisk kemmler (tully@74.195.67.126) |
| 17:39.34 | Oggu | Redid my modules.conf. No only noload-ing pjsip-stuff. Now it says i have loaded modules. But still no sound |
| 17:39.48 | napnap | Penguin, hard module arf :-\ I will check that |
| 17:40.08 | navaismo | google fxs module |
| 17:40.17 | [TK]D-Fender | Oggu: Is it complaining about the file? |
| 17:40.27 | napnap | navaismo, yeah, I read the manual of my card... |
| 17:40.30 | navaismo | the bot need a plugin for lmgtfy |
| 17:40.38 | navaismo | napnap, seems like you dont |
| 17:40.57 | Penguin | ~lmgtfy fxs module |
| 17:40.57 | infobot | ACTION thinks you should look here: http://lmgtfy.com/?q=fxs module |
| 17:41.08 | navaismo | awesome |
| 17:41.20 | Oggu | [TK]D-Fender: Yeah. Tried SayDigits instead now. It stops on saying a digit |
| 17:41.25 | navaismo | well the url miss the + |
| 17:41.33 | [TK]D-Fender | Oggu: then you have a networking issue |
| 17:41.40 | Penguin | ~lmgtfy fxs+module |
| 17:41.40 | infobot | ACTION thinks you should look here: http://lmgtfy.com/?q=fxs+module |
| 17:41.44 | kemmler | Hey, I'm getting an error when trying to use the RealTime feature. Here's the error. No database handle available with the name of 'astdb' (check res_odbc.conf). If I run isql -v asterisk-mysql I'm connected fine. Also, if I run odbc show in the asterisk cli it says Connected: Yes. There's definately the database astdb and not a typo. |
| 17:41.45 | napnap | navaismo, no, I meant I'm reading |
| 17:42.22 | Oggu | [TK]D-Fender: Ok. But im connecting from a sip phone on localhost... |
| 17:42.41 | [TK]D-Fender | doesn't mean you don't have a local firewall on that PC |
| 17:43.00 | Oggu | Now sound files work with just hello-world. But stops on that instead |
| 17:43.13 | [TK]D-Fender | meaning? |
| 17:43.23 | navaismo | 42 |
| 17:43.42 | Oggu | http://pastebin.com/gKPZJGxC |
| 17:43.44 | Oggu | And stops |
| 17:45.03 | [TK]D-Fender | I don't see the call ending... |
| 17:45.17 | Oggu | It doesn't |
| 17:45.21 | Oggu | It just stops there |
| 17:45.38 | Oggu | Until I hang up in the softphone |
| 17:45.46 | Oggu | So playback never "returns" |
| 17:47.08 | [TK]D-Fender | Oggu: "dialplan show from-internal" |
| 17:47.22 | [TK]D-Fender | and pastebin another call : "sip set debug on" |
| 17:47.59 | navaismo | i'm afraid that the Job's stuff made a bad installation of asterisk, is not pjsip required to chan_sip work correctly? |
| 17:48.23 | file | chan_sip does not use pjsip |
| 17:48.38 | Oggu | http://pastebin.com/WNEyL8YA |
| 17:49.12 | Oggu | dialpan http://pastebin.com/Fbsg6KVV |
| 17:49.38 | navaismo | file, ok, for what is installed ? |
| 17:49.51 | file | chan_pjsip uses it |
| 17:50.18 | navaismo | hmmm in 11? or I saw something wrong |
| 17:50.26 | file | in 12 |
| 17:50.27 | navaismo | stupid drugs |
| 17:51.08 | [TK]D-Fender | <--- SIP read from UDP:127.0.0.1:60384 ---> |
| 17:51.15 | [TK]D-Fender | I see the start of a read... and none of that packet |
| 17:51.26 | file | probably a keepalive |
| 17:52.00 | Oggu | I have some more general questions as well. The reason I'm experimenting with Asterisk in the first place. |
| 17:52.25 | Oggu | If asterisk is the right choice |
| 17:52.27 | Oggu | I want to use asterisk as a server which I control from an other service. For example I want to be able to tell asterisk to call a number. Notify the service when it is answered and then send back which of the connected "agents" to connect the call to. |
| 17:53.01 | napnap | navaismo, oh yeah, fuck, I checked my orders "FXO/FXS : 4 FXO, 0 FXS" :-\ . I wish I can change it.. thanks for your help. |
| 17:53.04 | navaismo | ugh another dialer |
| 17:54.01 | Oggu | I figured that is what AMI is for |
| 17:55.04 | [TK]D-Fender | file: the packet should still end |
| 17:55.05 | navaismo | Oggu, yea many asterisk are slaughtered every year to server as annoying dialers, also there are many versions like vicidial, elastix-callcenter and web scripts to make that |
| 17:55.32 | navaismo | and AMI is not for make automated calls only |
| 17:55.56 | Oggu | Annoying dialers? |
| 17:56.02 | *** join/#asterisk dorphalsig (b532ffa2@gateway/web/freenode/ip.181.50.255.162) |
| 17:58.00 | Oggu | I also need to be able to ask the other service who an incoming call should be directed to |
| 17:58.16 | navaismo | dialplan |
| 17:58.20 | navaismo | ~book |
| 17:58.20 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:00.43 | dorphalsig | Hi. I'm running asterisk 11 and have a discrepancy with my telco about some calls they claim were made through a vulnerability in my system. The only way to access the SIP trunks are through the asterisk box. There are no record in my cdr of the calls in question, is there any way to make a call through my asterisk box ignoring my diaplan and ignoring the CDR? |
| 18:02.18 | Oggu | We have already have a seperate system for all of this. And would want to avoid moving that logic into asterisk |
| 18:03.30 | rrittgarn | dorphalsig: Calls that are redirected via AMI don't show up in CDR... found that out the hard way. They do however show up in CEL if you have it enabled |
| 18:04.53 | Oggu | navaismo: Please explain further. This is the way we are doing it today but we are looking at using asterisk instead. Do you think there is a better way to do it? |
| 18:05.52 | file | CDRs were redone in 12, that statement may no longer be true |
| 18:05.58 | file | rrittgarn, ^^^ |
| 18:06.24 | navaismo | Oggu, there is no 'better way', but first aof all you need to install correctly asterisk then have some experience with asterisk then you can use plain asterisk and make the logic via dialplan or use a packed distro |
| 18:06.38 | rrittgarn | File: erhmagerd really? you have no idea how happy that makes me if AMI Redirect doesn't break it |
| 18:07.09 | file | rrittgarn, CDRs don't try to be smart and condense everything to a single record - so a call may have multiple now (which means you may need to do post-processing) |
| 18:07.34 | Oggu | navaismo: What can these packaged distros do? |
| 18:07.39 | rrittgarn | that's still better than 'Oh you wanted a CDR... too bad' |
| 18:07.49 | file | there's even a specification now |
| 18:08.01 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification |
| 18:08.03 | rrittgarn | my post processing now involves looking through the CEL table and creating CDRs from there |
| 18:08.05 | file | it only covers 12 though |
| 18:08.19 | Oggu | We are not looking for a simple "call center" as we need much deeper integration with our remaining systems |
| 18:08.21 | navaismo | Oggu, take a look on the vicidial page goautodial, elastix and so on |
| 18:08.58 | file | rrittgarn, seriously though... that page tells you everything you need to know about them in 12 :D it's nifty |
| 18:09.20 | file | and AND examples |
| 18:09.22 | rrittgarn | File: to whom do I owe a beer for this one? |
| 18:09.31 | file | Matt Jordan, mjordan |
| 18:09.43 | file | he sacrified himself to rewrite them |
| 18:10.20 | rrittgarn | I'll be at Astricon this year... if he's there I will be sure to procure him a beverage |
| 18:10.40 | Oggu | navaismo: Maybe I understand you wrong but the systems you mentioned are not as flexible and detailed as what I was looking for |
| 18:10.43 | file | he will definitely be there unless he quits being my (sort of a stretch of the imagination) boss |
| 18:11.02 | rrittgarn | i'll buy you both beers if you're both there haha |
| 18:11.45 | file | that would ruin my quest to get all of my drinks at every AstriCon going forward put on marketing's tab |
| 18:12.06 | rrittgarn | Well who am I to interrupt a quest |
| 18:14.34 | navaismo | Oggu, you need to configure it all even plain asterisk |
| 18:14.36 | leifmadsen | file: lol |
| 18:14.57 | file | leifmadsen, it pays to have friends in marketing |
| 18:14.58 | leifmadsen | file: "my (sort of a stretch of the imagination) boss" |
| 18:15.00 | navaismo | Oggu, but anyway first you need to install correctly and play with it(asterisk) |
| 18:15.04 | file | leifmadsen, oh, that, yes :P |
| 18:15.07 | leifmadsen | no one can control file! |
| 18:15.15 | leifmadsen | my doppelganger |
| 18:15.23 | leifmadsen | file... leif... suspicions arise! |
| 18:15.28 | Oggu | navaismo: Of course. |
| 18:15.50 | leifmadsen | file: look out! I'm hacking on C code for Asterisk! |
| 18:15.56 | *** join/#asterisk ThatCantBe (~irc@dsl.dyn-206.53.182.172.tbinet.bm) |
| 18:15.57 | file | leifmadsen, don't want it. |
| 18:15.59 | leifmadsen | so far it compiles |
| 18:16.49 | *** join/#asterisk retentiveboy (~retentive@74-95-28-34-Atlanta.hfc.comcastbusiness.net) |
| 18:16.53 | Oggu | navaismo: But before spending to much time hitting my head agains a wall. Is asterisk a good choice if you want to control it with outside logic? Or should I look for something else? |
| 18:17.35 | Oggu | I currently use a custom service built with pjsip |
| 18:19.20 | [TK]D-Fender | Oggu: What do you mean "control it"? |
| 18:19.23 | leifmadsen | asterisk 12 uses pjsip :) |
| 18:19.31 | [TK]D-Fender | Oggu: You need to be specific about your intentions |
| 18:20.54 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 18:23.19 | Oggu | I basically need to be able to have a lot of "internal" phones (agents) connected to a server. And then programatically tell that server to call outside people, connect calls (one, two or more in one call), receive incoming call and decide what do do about them based on information we have about the customer. (like who his contact person is) |
| 18:23.45 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 18:24.15 | rrittgarn | Oggu: we do something along those lines with astersk 11, NodeJS, and the AMI |
| 18:24.31 | Oggu | We are also currenly using Node |
| 18:24.36 | Ice_Strike | <PROTECTED> |
| 18:24.39 | Ice_Strike | You around? |
| 18:24.48 | Oggu | rrittgarn: How is your experience? |
| 18:24.51 | file | 12 you could do that with the new REST interface |
| 18:25.15 | rrittgarn | man file you're full of all sorts of nifty features i need to invent time to play with |
| 18:25.21 | rrittgarn | Oggu: our customers love it |
| 18:25.40 | Oggu | How is it to work with asterisk in that way? |
| 18:25.41 | file | Asterisk 12: We put a 12 on the box! Except we didn't, we actually spent tons of time doing cool new stuff. |
| 18:26.53 | file | rrittgarn, https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI?src=search |
| 18:27.11 | rrittgarn | well with asterisk 11, it was a bit of a learning curve, but its not the most difficult project we've encountered |
| 18:30.02 | file | and now I go home |
| 18:30.04 | file | runs off |
| 18:33.44 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 18:33.49 | *** join/#asterisk imcdona (~Thunderbi@209.181.91.201) |
| 18:36.57 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 18:39.32 | jeffspeff | fresh compile on a fresh install. keep getting error when doing "asterisk -crv" Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) the file does not exist, but the .pid file is there. here's exactly the cli output of what i'm doing prior and the checks done after error http://pastebin.com/9FeBs2C3 |
| 18:39.57 | *** join/#asterisk imcdona (~Thunderbi@209.181.91.201) |
| 18:40.46 | pabelanger | asterisk dead but subsys locked |
| 18:40.47 | [TK]D-Fender | jeffspeff: then you aren't having it look in the right place... |
| 18:41.22 | [TK]D-Fender | check your asterisk.conf vs your init script |
| 18:43.50 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 18:44.22 | jeffspeff | [TK]D-Fender, where can i find the init script? |
| 18:45.57 | navaismo | or maybe selinux enabled |
| 18:46.33 | jeffspeff | selinux is disabled prior to compiling |
| 18:49.49 | jeffspeff | [TK]D-Fender, i just verified that the asbindir, astvarrundir and astvarlogdir are the same in both asterisk.conf and /usr/sbin/safe_asterisk |
| 18:50.14 | jeffspeff | *astsbindir |
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| 18:51.52 | *** join/#asterisk imcdona1 (~Thunderbi@209.181.91.201) |
| 18:56.48 | Katty | HELLO NURSE |
| 18:57.17 | newtonr | Katty, Animaniacs ? |
| 18:57.22 | Katty | of course. |
| 18:57.24 | jeffspeff | lol |
| 18:57.38 | WIMPy | Katty: Please hold on. The next free nurse is reserved for you. |
| 18:57.55 | Katty | nurse? free?! |
| 18:58.09 | Katty | there are no free nurses in 'murica |
| 18:58.24 | WIMPy | Ok, I guess that fshould have been "vacant". |
| 18:59.01 | MaliutaLap | Katty: I'm not a nurse, but I've spent enough time in hospital "fixing" my infusion rates and correcting nurses on procedure that I can probably do the job :) |
| 18:59.27 | MaliutaLap | Katty: also have heaps of first aid training :) |
| 19:01.55 | Katty | how do you deal with a hypochondriac? |
| 19:03.16 | WIMPy | By writing bills? |
| 19:03.21 | Katty | nods |
| 19:03.23 | Katty | so it seems. |
| 19:03.34 | Katty | ALSO. on topic today, which IS surprising |
| 19:03.40 | *** join/#asterisk karlh626 (~karlh626@addr-199.21.193.173.nptpop-cmts-cable-sub.rdns-bnin.net) |
| 19:03.46 | Katty | in what order are things supposed to go in regards to numbers, letters and symbols |
| 19:03.48 | WIMPy | shrugs |
| 19:04.01 | Katty | a, 1, and * for example |
| 19:04.09 | WIMPy | Where? What? |
| 19:04.19 | Katty | in extensions.conf under say [outbound] |
| 19:04.29 | Katty | what order are you /supposed/ to put things in |
| 19:04.36 | WIMPy | The one you can handle. |
| 19:04.41 | [TK]D-Fender | Katty: * sorts them anyway |
| 19:04.53 | [TK]D-Fender | Katty: Doesn't matter how you do your extensions.conf |
| 19:04.54 | Katty | but is there a Best Practice(tm) |
| 19:04.59 | Katty | k |
| 19:05.00 | WIMPy | Might make sense to sort them alphabetically as * does. |
| 19:05.14 | [TK]D-Fender | Katty: It's all aesthetic at that point. |
| 19:05.16 | WIMPy | That way you wuld see easily which extension would match. |
| 19:05.21 | WIMPy | Hopefullt. |
| 19:05.23 | Katty | i will appease my ocd then |
| 19:05.41 | [TK]D-Fender | Katty: but since I use "s" for IVR's etc, I tend to do those at the top, numbered ones in order, then lettered |
| 19:06.06 | [TK]D-Fender | Katty: OCD only requires standardization. Pick whatever cooperates best with yours :) |
| 19:06.07 | Katty | [TK]D-Fender: what extension do you generally use for Page() |
| 19:06.15 | Katty | [TK]D-Fender: or do you access it via an IVR? |
| 19:06.23 | Katty | or anyone else for that matter. |
| 19:06.28 | Katty | i believe walmart uses *10 |
| 19:06.35 | [TK]D-Fender | Katty: Never really used it personally... and I'm running FreePBX at home & work... |
| 19:06.50 | [TK]D-Fender | Katty: And using default feature code settings |
| 19:07.09 | Katty | there's a code in features.conf for page? |
| 19:07.13 | Katty | goes to look |
| 19:07.13 | [TK]D-Fender | Katty: So were we ever to get around to it ... it'd be something-or-other ;) |
| 19:07.20 | [TK]D-Fender | Katty: No, FreePBX <- |
| 19:07.30 | Katty | well i'm certainly not going to go dig through that. |
| 19:07.32 | Katty | i'd need a shower. |
| 19:07.39 | Katty | also probably a nap. |
| 19:07.45 | Katty | oh look, google |
| 19:08.21 | Katty | google claims it's 00 |
| 19:08.29 | Katty | and by google i mean the wiki.freepbx.org page |
| 19:08.50 | Katty | what do the rest of you use? |
| 19:09.50 | [TK]D-Fender | looks like *80 |
| 19:10.09 | Katty | link? |
| 19:10.24 | [TK]D-Fender | No link... off my config screen |
| 19:10.32 | [TK]D-Fender | actually, that *80 + Ext for direct single-set |
| 19:10.45 | [TK]D-Fender | In the Paging Groups section, you pick the memebers and invent your own code... |
| 19:10.51 | [TK]D-Fender | So I guess there isn't really a standard there |
| 19:11.01 | [TK]D-Fender | So "N/A" |
| 19:18.22 | navaismo | do you guys know a good upgrade mysqldb tool, I was trying with mysqldiff to migrate an old db with data to the new schema but didn't work |
| 19:19.17 | Katty | pstgres? |
| 19:19.20 | Katty | postgres :> |
| 19:20.02 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:23.46 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 19:24.29 | navaismo | :'( |
| 19:24.49 | DruidZ | Regarding my previous question, I found the answer. |
| 19:25.20 | Katty | consoles navaismo |
| 19:25.24 | DruidZ | My question: Why doesn't "exten => 4165551111,1,Dial(SIP/4165551111@sip.voip.ms,1)" work? |
| 19:26.02 | DruidZ | Answer: the ",1" timeout. I was giving up before voip.ms dealt with the call. |
| 19:26.12 | DruidZ | exten => 4165551111,1,Dial(SIP/4165551111@sip.voip.ms) |
| 19:27.04 | DruidZ | I was confused because "exten => *51,1,Dial(SIP/thetestcall@sip.antisip.com,1)" did work. They must be quicker. |
| 19:27.18 | DruidZ | Not sure where I got the example from. |
| 19:28.40 | [TK]D-Fender | "core show application dial" <- |
| 19:28.52 | [TK]D-Fender | Always be sure to read the instructions for your apps & understand them |
| 19:34.09 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 19:38.24 | *** join/#asterisk bulkorok (~Adium@053d9311.dynamic.tele-ag.de) |
| 19:44.32 | *** join/#asterisk qakhan (~qakhan@50-204-254-12-static.hfc.comcastbusiness.net) |
| 19:45.43 | DruidZ | It's been a rough week. My FS switch died and would not work. Since I was planning on switching to * anyway I did that once I realized that it was going to take a lot of debugging to get FS back up. |
| 19:46.15 | Mango45 | What makes you prefer Asterisk, DruidZ? |
| 19:46.22 | Mango45 | hasn't learned FS yet. |
| 19:46.35 | DruidZ | Overall I was quite please with how quickly I was able to understand the basics and get my users back up under Asterisk. |
| 19:47.49 | DruidZ | Mango45: I need to sit down and make a list. |
| 19:48.19 | DruidZ | It just seemed so fragile. Asterisk is already showing more stability. |
| 19:48.26 | Mango45 | Ah, interesting. |
| 19:49.14 | *** part/#asterisk fireman_biff (~biff@208.0.98.13) |
| 19:49.37 | DruidZ | And I can restart it without rebooting the server. For some reason a restart on FS would fail to recognize any extensions and there was nothing on any of the lists or wikis explaining why. |
| 19:50.19 | DruidZ | Since my servers tend to take five minutes to reboot that made experimenting painful. |
| 19:51.03 | WIMPy | I know that feeling. Might make more sense to use the better hardware for testing and the old shit for production :-) |
| 19:51.07 | DruidZ | Also, FS seems to be very Linux oriented. If you run BSD they seem to think that the answer is "too bad." |
| 19:51.18 | Mango45 | Good plan WIMPy. :) |
| 19:51.31 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 19:52.16 | WIMPy | Trying FS on my test box was horribly slow. |
| 19:53.10 | qakhan | i cannot hear voice on incoming or outgoing through sip trunk but its works fine on elastrix |
| 19:53.25 | qakhan | here is my elastix sip debug |
| 19:53.56 | qakhan | http://pastebin.com/u3RiQ6dK |
| 19:54.11 | navaismo | keep using elastix please |
| 19:54.18 | qakhan | and here is my asterisk sip debug http://pastebin.com/8EEaJPnY |
| 19:54.35 | DruidZ | The server was fast enough. Multi CPU HP server with 4GB RAM. It was just not optimized for booting. It was optimized for running. |
| 19:56.23 | DruidZ | FS also has some ugly code. When I tried to suggest improvements I got "compiles on Linux so it must be OK." |
| 19:57.09 | DruidZ | It compiles on Linux only because they ignore the thousands of warnings. |
| 19:58.32 | *** part/#asterisk bulkorok (~Adium@053d9311.dynamic.tele-ag.de) |
| 19:58.48 | file | in developer mode you can't build Asterisk if there are any warnings |
| 19:59.17 | DruidZ | likes "-Wall -Werror" at a minimum. |
| 19:59.45 | DruidZ | file: See, that's the attitude that I like. |
| 20:00.24 | WIMPy | Oh, I know other kinds of ugly code and guess where I found that? |
| 20:00.35 | DruidZ | In NetBSD (where I am a developer) you can't build and install packages in developer mode. |
| 20:01.24 | DruidZ | That is, you can't build incorrect packages even though they might build for users if they made it into the wild. |
| 20:01.29 | *** join/#asterisk startledmarmot (~startledm@cpe-76-172-72-246.socal.res.rr.com) |
| 20:01.46 | startledmarmot | quick q: looking for a freelance asterisk con for an old client of mine |
| 20:02.23 | startledmarmot | standard hosted pbx/freepbx style stuff - shoot me a pm if you're in that game |
| 20:03.58 | DruidZ | startledmarmot: I just started with Asterisk on Monday so I am no expert yet. Can you wait until tomorrow? :-) |
| 20:04.26 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:04.30 | startledmarmot | DruidZ: ;) Welcome though! |
| 20:04.34 | qakhan | [TK]D-Fender can you please check my both pastebin |
| 20:04.45 | WIMPy | DruidZ: :-) |
| 20:05.32 | Mango45 | DruidZ: Welcome. :) |
| 20:06.42 | *** join/#asterisk tony2012 (~Adium@quar13195.rit.edu) |
| 20:07.29 | DruidZ | Thanks. I suppose I came off as a whiny noob over the last few days but I was under the gun. I did do a lot of research before asking dumb questions here but sometimes I really needed the quick answer. |
| 20:08.02 | DruidZ | I hope to be feeding back very soon. |
| 20:09.35 | DruidZ | goes to re-read the docs to see what suggestions he can make now. |
| 20:27.44 | *** join/#asterisk triplef (~IPTfrank@QUBCPQ0336W-LP140-03-2925327618.dsl.bell.ca) |
| 20:28.18 | triplef | hey all , is there a way to route calls on peer doing it ? exten => _3XXX,n,Dial(SIP/${EXTEN},300) but i would want if peer = 3001 then something else |
| 20:28.31 | triplef | so if im user 3001 , and i dial 3002 i want a busy |
| 20:28.46 | triplef | but if im user 3004 and i dial exten => _3XXX,n,Dial(SIP/${EXTEN},300) its ok |
| 20:30.21 | Mango45 | Set 3001 and 3004 up in different contexts perhaps? |
| 20:30.25 | triplef | i can't lol |
| 20:30.28 | triplef | too easy |
| 20:30.41 | triplef | but can i use like callerid ? |
| 20:30.56 | Mango45 | Sure, as long as the user doesn't have the ability to spoof it. |
| 20:30.57 | triplef | exten => _3XXX/3001,n,Dial(SIP/${EXTEN},300) its ok |
| 20:31.14 | triplef | nah they my little hamsters working, i just want to restrict one who calls ALL THE TIME |
| 20:31.58 | triplef | exten => _3XXX/3001,1,Busy() |
| 20:32.02 | triplef | like this i guess |
| 20:32.28 | triplef | else i add a context ;) |
| 20:33.02 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
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| 20:33.26 | Penguin | You don't need a different context. There is nothing wrong with the one you already have. |
| 20:33.38 | Penguin | And a different context won't provide you any other options anyway. |
| 20:34.04 | Penguin | What you do is add an extension. |
| 20:34.19 | Penguin | or add some tests to the current extension. |
| 20:35.00 | triplef | yeah could do that but then it gets complicated |
| 20:35.15 | Penguin | Not really. |
| 20:35.19 | triplef | gotoif macro with {$peer but i guess not sure what variables i can use |
| 20:35.19 | navaismo | gotoif evaluating ${CALLERID(NUM)} should work |
| 20:35.21 | triplef | oh accountid |
| 20:35.22 | Penguin | It's very clear to me. |
| 20:35.33 | Penguin | No macro required. |
| 20:35.45 | triplef | accountcode |
| 20:35.57 | Penguin | You can use callerid number or account code. or both. |
| 20:36.02 | triplef | i already have a macro before the dial i could do in there ;) |
| 20:36.11 | triplef | nah ill just do there |
| 20:36.21 | Penguin | There's no macro needed. |
| 20:37.16 | triplef | GotoIf($[${accountcode} == 3001]?n:something,1) |
| 20:38.02 | Penguin | If the accountcode variable is 3001 for that phone, that would evaluate true. |
| 20:38.33 | Penguin | You could also use ExecIf() to execute the Busy() application rather than using GotoIf() to send it somewhere else. |
| 20:38.34 | triplef | GotoIf($[${CDR(accountcode)} == 3001]?n:something,1) |
| 20:38.36 | triplef | better ;) |
| 20:38.51 | triplef | ah execif let's read on this ;) |
| 20:39.23 | triplef | ExecIf(expression?appiftrue(args) |
| 20:39.33 | Penguin | ExecIf($[x${CDR(accountcode)} = x3001]?Busy()); |
| 20:39.48 | triplef | so ExecIf($[${accountcode} == 3001]) |
| 20:39.50 | triplef | lol was doing it thanks |
| 20:40.30 | triplef | why the x ? |
| 20:40.42 | triplef | to force a string cmp ? |
| 20:40.54 | Penguin | In your example, if the accountcode variable is ever null value, it will die horribly. |
| 20:41.02 | triplef | man didnt touch asterisk in like 4-5 years now lol |
| 20:41.15 | triplef | used to devel back in the days when anthm etc where areound |
| 20:41.22 | Penguin | The x on both sides forces it to a string in the event there is a null value. |
| 20:41.42 | Penguin | You could also use double quotes to effect the same thing. |
| 20:41.54 | Penguin | ExecIf($["${CDR(accountcode)}" = "3001"]?Busy()); |
| 20:42.51 | triplef | tks updated watching on cameras to see when she calls now lol |
| 20:42.57 | triplef | fun to work in retail surveillance |
| 20:43.31 | triplef | nice to have a no cell policy and they ALL get phones on counter ;) tried to import jammers once but LAE stopped me straight lol |
| 20:48.51 | leifmadsen | LAE? |
| 20:49.26 | *** join/#asterisk slicknick5181 (~slicknick@207-255-116-208-dhcp.unt.pa.atlanticbb.net) |
| 20:50.23 | triplef | law enforcement Agency |
| 20:50.25 | triplef | LEA |
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| 21:35.06 | dym | Can anyone recommend a decent (germany based) sip trunk provider (except sipgate)? |
| 21:35.22 | dym | Also "deutsche telefon" doesnt seem to be that great |
| 21:50.51 | WIMPy | Well, that's understood if they use SIP. |
| 21:50.58 | WIMPy | What do you want? |
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| 22:13.33 | file | moo |
| 22:14.27 | navaismo | are you on drugs again? |
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| 22:19.04 | file | no. |
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| 22:37.09 | tony2012 | Anyone good at res_xmpp? I'm having an issue where JABBER_STATUS isn't returning the correct status when a client is offline. It returns 7, should return 6. |
| 22:39.39 | tony2012 | I can see the unavailable presence data come in via xmpp debug, but then a call to JABBER_STATUS returns 7. |
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