| 00:02.42 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
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| 00:09.12 | Katty | hellllllllllloooo all you wonderfully smart peoples! |
| 00:11.33 | Katty | so let's say you have a server on Network A. |
| 00:11.38 | Katty | Network A has a few sip phones. |
| 00:11.54 | Katty | then you take a phone home to Network B. |
| 00:12.00 | Katty | which is a dynamic ip address. |
| 00:12.14 | Katty | let's also assume you don't want to leave port 5060 open for the entire world. |
| 00:12.40 | Katty | short of allowing a port range for network B's phone to connect with Network A. |
| 00:12.54 | Katty | or using a software phone on a computer which is VPNed into network A |
| 00:13.07 | Katty | there any other good ways of handling that? |
| 00:13.55 | niekvlessert | iptables i guess |
| 00:14.00 | carrar | setup a hardware vpn? |
| 00:14.10 | Katty | hardware vpn would work |
| 00:14.17 | carrar | use TLS? |
| 00:14.49 | Katty | let's assume this phone is now moving. |
| 00:14.59 | Katty | i'm in Motel 8 one day, and Burger King the next. |
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| 00:15.12 | Katty | hardware vpn is kinda out of the question... |
| 00:15.15 | niekvlessert | tls then |
| 00:15.27 | florz | "hardware vpn"? WTH? |
| 00:15.39 | Katty | florz: yes. |
| 00:15.51 | Katty | florz: as opposed to authenticating with a piece of software on a computer back to a server. |
| 00:15.54 | Katty | florz: Perminent Tunnel |
| 00:15.56 | florz | Katty: what is that? |
| 00:16.00 | Katty | florz: your mom. |
| 00:16.11 | Katty | infobot: tls? |
| 00:16.12 | infobot | extra, extra, read all about it, tls is Transport Layer Security (ssl) but there could be thread-local storage or, in polish, TrzyLiterowy Skrot, what's mean ThreeLetter Acronym |
| 00:16.12 | florz | Katty: so, "hardware" means "permanent"? |
| 00:16.25 | Katty | infobot: you are so helpful |
| 00:16.25 | infobot | Katty: what are you talking about? |
| 00:16.44 | Katty | infobot: if i knew iw ouldn't be asking you |
| 00:16.58 | niekvlessert | i heard about a phone having integrated vpn |
| 00:17.07 | Katty | that would be nice |
| 00:17.09 | Katty | really really nice |
| 00:17.11 | Get_The_Fish | is it possible to set a variable name by doing something Set(ch-all-INVALID_${CONTEXT}=1) |
| 00:17.19 | niekvlessert | well, tls is better i guess |
| 00:17.22 | niekvlessert | much less overhead |
| 00:17.26 | florz | you can install some openvpn-enabled firmware on snoms |
| 00:17.48 | fink | do any snom phones support iax? |
| 00:17.48 | niekvlessert | because rtp = udp, and vpn = tcp most of the times |
| 00:17.51 | Katty | i'm not sure how you can get a polycom to use TLS, but you know... |
| 00:17.59 | Katty | i'll look into it later |
| 00:18.08 | florz | niekvlessert: what vpn would be tcp? |
| 00:18.13 | niekvlessert | or one should use a udp vpn.... but don't know how that'll work in real live |
| 00:18.25 | carrar | reg.1.server.1.transport="TLS" |
| 00:18.38 | fink | florz: openvpn works over tcp |
| 00:18.42 | niekvlessert | florz: well most of them i guess? |
| 00:18.44 | Katty | carrar: oh hey, how bout that |
| 00:18.44 | upb | lol wtf |
| 00:18.45 | carrar | need to also setup your SSL cetlist stuff in polycom config |
| 00:18.50 | niekvlessert | fink: or udp, you can set that up |
| 00:18.50 | Katty | carrar: there any VPN stuff in there? :P |
| 00:19.02 | fink | niekvlessert: yup |
| 00:19.03 | florz | fink: well, you can configure that if you want to, yes - but I guess that's really more of an exception |
| 00:19.04 | niekvlessert | ssh vpn uses tcp |
| 00:19.05 | carrar | You said you don't want a vpn |
| 00:19.11 | carrar | so TLS should work |
| 00:19.16 | Katty | carrar: but if the phone has a mobbbbileee vpn |
| 00:19.21 | Katty | carrar: :>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> |
| 00:19.24 | carrar | well vpn hardware |
| 00:19.27 | Katty | :< |
| 00:19.32 | niekvlessert | katty: use tls |
| 00:19.37 | carrar | tls is "vpn" |
| 00:19.50 | Katty | i'll go read about tls |
| 00:19.54 | Katty | after i eat this sammich |
| 00:19.55 | niekvlessert | or you could use a softphone with ssh -w :) |
| 00:20.07 | florz | wth would you want to do telephony over TLS?! |
| 00:20.17 | Katty | twitches |
| 00:20.22 | niekvlessert | florz: security i guess |
| 00:20.23 | Katty | resists meme |
| 00:20.30 | florz | I mean, if you don't have an alternative, ok |
| 00:20.46 | niekvlessert | florz: what would think is the best alernative |
| 00:20.56 | Katty | not carrying a phone. |
| 00:21.00 | Katty | that's a great alternative |
| 00:21.05 | niekvlessert | best alternative = the one with lowest overhea |
| 00:21.05 | florz | niekvlessert: some udp based VPN? like, openvpn over udp? |
| 00:21.06 | niekvlessert | d |
| 00:21.54 | niekvlessert | florz: http://tools.ietf.org/html/rfc4347: âhttp://en.wikipedia.org/wiki/Datagram_Transport_Layer_Securityâ specifies a TLS variant that works over datagram protocols (such as UDP). |
| 00:21.59 | niekvlessert | i guess that is what most phones do |
| 00:22.01 | dlynes | Is asterisk capable of playing nicely with tdd/tty? |
| 00:22.27 | Katty | dlynes: if you distract it with chocolate. |
| 00:22.56 | florz | and btw, the usual VPNs are, I would think, ipsec, pptp, or udp-based openvpn |
| 00:23.07 | florz | none of them being based on tcp |
| 00:23.16 | upb | florz: newsflash - openvpn uses TLS aswell |
| 00:23.22 | niekvlessert | yeah, they all suck with nat & port usage except openvpn |
| 00:23.25 | niekvlessert | openvpn ftw!! |
| 00:23.30 | florz | upb: well, but not for the transport |
| 00:24.14 | florz | I mean, tls is all nice and fine, but using a sequential delivery transport for telephony doesn't sound like the best solution to be |
| 00:24.15 | dlynes | Katty, heh |
| 00:24.15 | florz | erm, to me |
| 00:24.54 | niekvlessert | florz: well maybe it works the same as normal rtp, but only the packets are encrypted |
| 00:24.55 | niekvlessert | btw |
| 00:25.07 | niekvlessert | is tls also for rtp? |
| 00:25.14 | florz | niekvlessert: any phones that support tls for media transport?!? |
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| 00:25.22 | niekvlessert | florz: no idea |
| 00:25.29 | niekvlessert | http://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocol |
| 00:25.38 | niekvlessert | this would be the way to go :) |
| 00:25.53 | niekvlessert | tls 4 sip & srtp 4 media |
| 00:26.05 | florz | niekvlessert: usually, voip crypto uses SIPS for the signalling (that's SIP over TLS over TCP) and SRTP for the media stram |
| 00:26.07 | niekvlessert | i already knew that, but i forgot :) |
| 00:26.18 | florz | but that doesn't solve any NAT problems |
| 00:26.23 | niekvlessert | true |
| 00:27.00 | niekvlessert | when will IPV6 be the way to go! :) |
| 00:27.37 | niekvlessert | openvpn with udp, will that give much overhead? |
| 00:27.54 | Get_The_Fish | when ISP's actually adopt it for CPE's and a strategy of ISP IPv6 deployment is solidified |
| 00:27.57 | florz | well, depends ;-) |
| 00:28.25 | niekvlessert | florz: on what? the packet loss? |
| 00:28.46 | niekvlessert | sip conntrack ftw |
| 00:28.47 | florz | no, configuration ;-) |
| 00:29.04 | niekvlessert | florz: so you use openvpn when having nat crap |
| 00:29.25 | florz | that is: you can have compression, which probably would reduce bandwidth overhead to somwhere close to zero - but that causes CPU overhead, of course |
| 00:29.45 | niekvlessert | florz: and echo maybe? |
| 00:29.54 | florz | _echo_? |
| 00:30.38 | niekvlessert | yeah, conversion and/or compression takes time, so latency might cause echo if too much |
| 00:31.38 | niekvlessert | well, anyway, i'm going to bed :) |
| 00:31.40 | florz | well, if you do have any echo in the first place, it might become noticable |
| 00:31.58 | florz | depends on the processors and stuff, obviously |
| 00:32.41 | niekvlessert | florz: well i'm going for openvpn with udp. :) |
| 00:32.42 | niekvlessert | cya |
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| 01:14.45 | UserReg_CL | hi, need execute un test de ivr |
| 01:15.07 | leifmadsen | eh? |
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| 01:20.10 | TJNII | *Hi, I need to execute an IVR test. |
| 01:22.32 | UserReg_CL | ok |
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| 01:35.20 | Get_The_Fish | hey leif, are you watching the room? Can I bounce something off ya? |
| 01:37.07 | leifmadsen | Get_The_Fish: you can ask a question and I will respond if I'm here and I know the answer ;) |
| 01:37.15 | leifmadsen | TJNII: define "asterisk test" |
| 01:37.28 | leifmadsen | TJNII: oh, you were translating :) |
| 01:37.57 | Get_The_Fish | lol... having some issues with attended transfer in 1.6.2.5. I believe that this might have been addressed but I cant find the issue report in Mantis and it doesnt show up in the roadmap |
| 01:38.28 | Get_The_Fish | I know that there have been some regressions in the 1.6.1-1.6.2 versions, specifically with local channels, followme and transfers. |
| 01:38.59 | Get_The_Fish | so, the question is, should I post to the ML, post a bug on Mantis, or just wait as it has been addressed? |
| 01:41.00 | Get_The_Fish | TJNII, you can call into the IVR yourself, or you could write some SIPp tests to do it for you. I recommend the SIPp testing, as it can be done repeatedly and captured for analysis |
| 01:41.21 | Get_The_Fish | I should say TJNII/UserReg_CL |
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| 01:57.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 02:04.57 | leifmadsen | Get_The_Fish: did you get my msg's before? (I changed networks) |
| 02:05.08 | leifmadsen | <leifmadsen> Get_The_Fish: test the latest release candidates at least, or even better, test from the branches directly |
| 02:05.08 | leifmadsen | <leifmadsen> I know a lot of issues in attended transfer and local channels have been fixed -- you should be able to find them under the closed issues in mantis |
| 02:05.08 | leifmadsen | <leifmadsen> you can post to the mailing list, but all anyone is going to say is to test the latest RCs and/or direct from the branches |
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| 02:08.18 | leifmadsen | ctooley: snap! |
| 02:09.08 | ctooley | leifmadsen, horrible typo... the only excuse I can think of for me ending up here. :-) |
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| 02:14.15 | leifmadsen | ctooley: lol |
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| 02:21.49 | *** join/#asterisk p3rror (~Ali@anonymous-196-206-82-181.broker.freenet6.net) |
| 02:22.31 | p3rror | hello, please i have a question, i seeone thing thata friend compose an USSD and after that he got and menu on his mobile |
| 02:22.42 | p3rror | please can you tell me what technology behind that |
| 02:30.10 | trentcreek | p3rror: I found the asnwer in .00000000001 seconds using google |
| 02:31.48 | p3rror | please can you provide a good link for this |
| 02:32.06 | p3rror | i think that i may look to ussd technology |
| 02:32.08 | trentcreek | sure: http://www/google.com |
| 02:32.27 | p3rror | trentcreek, =) |
| 02:32.35 | trentcreek | oops.. https://www.google.com |
| 02:32.49 | p3rror | trentcreek, i need more than google |
| 02:33.03 | p3rror | i use it 9 years ago |
| 02:33.18 | trentcreek | p3rror: I found the answer in .00000000001 seconds using google |
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| 02:33.54 | p3rror | so what was your query |
| 02:34.30 | trentcreek | this is a channel about asterisk |
| 02:36.50 | p3rror | ok, sorry |
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| 02:37.37 | LemensTS | I . |
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| 03:55.53 | WhIteSidE | Hello all, I've been fighting with app_swift for the past several hours, and I simply cannot get it to compile. I've tried patching, etc, etc. What's the current "best way" to get cepstral working asterisk 1.6 |
| 03:56.05 | WhIteSidE | ? |
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| 05:39.35 | dddh | hm, what does mean Registration from '"1267572396"<sip:1267572396@8.9.10.11>' failed for '79.125.39.13' - No matching peer found |
| 05:39.45 | dddh | 8.9.10.11 is my ip adress |
| 05:40.29 | dddh | are they trying to use my * to call somewhere? |
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| 05:43.45 | [TK]D-Fender | dddh: No, they are trying to register |
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| 05:44.37 | dddh | [TK]D-Fender: ok, but why are they trying to register? :) |
| 05:45.41 | [TK]D-Fender | dddh: probably trying to hack away at you |
| 05:46.11 | dddh | [TK]D-Fender: oh |
| 05:48.27 | dddh | should report abuse to amazon |
| 05:48.50 | dlynes | dddh should configure his box securely, so he doesn't need to worry about it |
| 05:53.00 | dddh | dlynes: you mean I should read more? |
| 05:54.16 | ChannelZ | that's kind of a stupid username to try and register as |
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| 06:00.37 | dddh | they also tried registering as 3918657192@ |
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| 06:47.49 | redax | good morning, |
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| 07:25.02 | Hexie | anyone here know how i can connect to a Siemens PABX ? |
| 07:25.16 | Hexie | i have a Siemens and would like to see the configuration of it? |
| 07:25.33 | Hexie | *without having to get a techy out (R700 call out fee) |
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| 07:32.17 | Maliuta | Hexie: I looked into that a while back ... you can get a VoIP card for them, or you could trunk stuff throufgh via ISDN |
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| 08:10.14 | Hexie | even if i'd just like to see the configuration (IP address's etc?) |
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| 08:25.41 | edbaffei | Hi #asterisk i have just installed asterisk from the iso cd. I was not prompted for a password other than my root psw. Now i cant login to freepbx admin? |
| 08:25.59 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 08:26.11 | edbaffei | The default password maint - password doesn't work? |
| 08:26.47 | edbaffei | how do i login to the system? |
| 08:28.43 | kaldemar | ask in #asterisknow r #freepbx |
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| 08:44.05 | hurdman | hi |
| 08:44.10 | hurdman | i'm looking for something about the Dial(SS7/number:redirecting-number) command into chan_ss7 module |
| 08:44.14 | hurdman | i try to make a transfer without a dial, is-it possible with ss7 ? |
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| 09:30.57 | feature | hi there. trying to setup dymanic conferences in asterisk (not static server-side rooms, but the ones, which can be created 'on-fly' by a random callee). say, person1 is talking to person2 and suddenly wanted some person3 to join them both. |
| 09:32.02 | feature | found 2 modules: meetme and app_conference, but didn't find a word, that would satisfy my needs |
| 09:34.08 | feature | if anyone had a success story with that, a couple of keywords to google for me will be enough |
| 09:34.12 | feature | thanks in advance |
| 09:35.00 | kaldemar | feature: there's app ConfBridge in 1.6.2 that might suit you. |
| 09:36.15 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:37.17 | feature | hmmm, 1.6.0.21 here. i'll try to update it, thanks a lot. |
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| 09:55.07 | J4zen | Hi guys, is there any tool/calculator/formula that can be used to determine the maximum number of concurrent calls on a given DSL connection? I'm using G.711 Ulaw codec on a 1 mbit upload/20 mbit download DSL connection and want to find out what the maximum number of concurrent calls is. |
| 09:55.48 | kaldemar | J4zen: http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 09:56.40 | J4zen | kaldemar: thanks , just what i was looking for :) |
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| 10:16.28 | Katty | :< |
| 10:16.36 | Katty | i has contracted teh ills. |
| 10:17.35 | P424D0X | How I must configure Asterisk, so that a CISCO 7940 works in own subnetz behind the Asterisk server. |
| 10:19.58 | Chainsaw | Katty: Oh noes :( |
| 10:20.13 | Katty | Chainsaw: yes indeed :< |
| 10:20.21 | Katty | Chainsaw: but i can't imagine who :/ |
| 10:20.36 | Chainsaw | Katty: Been on any trains or planes lately? |
| 10:20.44 | Chainsaw | Katty: I'll bet someone sneezed on you. |
| 10:20.58 | Katty | possibility |
| 10:21.05 | Katty | or i walked into a sneeze |
| 10:21.21 | Katty | SOMEONE BREATHED ON ME! /gasp |
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| 10:25.55 | SystemT` | Katty how are u |
| 10:27.05 | Katty | short of throwing up every 20 minutes--i'm just peachy |
| 10:27.23 | Katty | how're you? |
| 10:30.41 | SystemT` | good :) |
| 10:31.07 | SystemT` | Katty i have a littel problem on my asterisk when call is comming is hang up the call immidatly |
| 10:31.17 | SystemT` | and write like answerd |
| 10:31.31 | Katty | that is most unfortunate. i am sorry to hear that. |
| 10:31.42 | SystemT` | can u help me with that? |
| 10:31.53 | Katty | no |
| 10:31.59 | SystemT` | why not? |
| 10:32.13 | Katty | because i will probably be running to the bathroom again here in another 10 minutes |
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| 10:32.28 | SystemT` | aha u sick ? |
| 10:32.36 | Katty | quite. |
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| 10:48.52 | Katty | frowns |
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| 10:55.33 | cuco | i am trying to logon to ajam, but I always get "HTTP Connect attempt from '192.168.0.178' unable to authenticate". I am pretty sure the password is ok. What else can be the problem? |
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| 11:14.07 | feature | kaldemar: regarding ConfBridge advice. have you ever tried or seen its functionality, I've described above in original question? searching docs, but still no success stories about that. going to check out source code already, lol, but decided to ask first. |
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| 11:31.15 | kaldemar | feature: you can for example make an extension that makes and joins conferences. use the exten as confno or something else. depenpds on how your extension will look like. |
| 11:36.05 | Akiraa | Did anyone see or did implement voip servers on dd-wrt or open-wrt capable routers? If so, would you recommend this solution for live small scale (20-ish terminal) applications? |
| 11:48.27 | Chainsaw | Akiraa: It depends on whether any transcoding is required. |
| 11:48.45 | Chainsaw | Akiraa: If you don't need transcoding and are not planning on use MeetMe bridges or other fancy features, it'll probably work fine. |
| 11:59.47 | Chainsaw | Does hint support callgroups/pickupgroups at all? |
| 12:04.24 | tzafrir | cuco, /j #elastix |
| 12:04.40 | tzafrir | err.. wrong channel, I guess |
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| 12:11.05 | sulex | "1.4.23.1" here, is there any chance to log microseconds within logfiles? %q is not interpolated as I've expected if declared in logger.conf dateformat directive. thanks |
| 12:14.39 | tzafrir | sulex, what exactly do you need micro-seconds for? |
| 12:15.09 | tzafrir | I suspect that event if Asterisk logged time in such granularity, you wouldn't get an acurate time |
| 12:15.39 | sulex | tzafrir: we have to calculate the time required for a box to hangup to a call coming from a PRI |
| 12:16.00 | sulex | tzafrir: im fine also in trapping that time by some dahdi trick |
| 12:16.35 | tzafrir | not sure how to do it in dahdi, as dahdi is merely a pipe for d-channel events |
| 12:16.35 | sulex | tzafrir: the hangup is sent immediately, there's not a deliberate answer to the call |
| 12:17.51 | tzafrir | sulex, have you tried looking at the pri-level trace? (pri set debug) |
| 12:17.57 | sulex | tzafrir: Im sure that the time format internally is microseconds aware, but than a conversion is done before logging or showing dates on the cli |
| 12:20.00 | tzafrir | could you be more specific as for the reason / excuse asterisk gives for the hangup? |
| 12:22.43 | sulex | tzafrir: -- Channel 0/1, span 1 got hangup request, cause 16 |
| 12:23.08 | sulex | tzafrir: there's a brutal Hangup() at priority 1 of the extension |
| 12:23.58 | sulex | tzafrir: nothing else, what I'd like to calculate is the time to detect the call and than kill it |
| 12:23.59 | tzafrir | sulex, so why do you expect Asterisk not to hang up the call? |
| 12:24.15 | tzafrir | Why do you need to kill the call? |
| 12:25.13 | sulex | tzafrir: because the test is to calculate how long asterisk would take to kill the call after it receive it. now I've a value of one second due the pour granulairty of the logger |
| 12:25.52 | sulex | tzafrir: this is why I'm asking if there's a way to log microseconds. as far as i can see in 1.6 you can use %q from strftime() |
| 12:26.03 | sulex | tzafrir: but in 1.4 it seems not working |
| 12:27.11 | tzafrir | there are various ioctl calls at the DAHDI levels. e.g. the one that sets the channel in audio mode |
| 12:27.38 | *** join/#asterisk fleebailey33 (fleebailey@unaffiliated/fleebailey33) |
| 12:27.56 | sulex | tzafrir: ok, so basically ive to defocus from asterisk and go down to the "hardware" level... that makes more sense in what Im doing |
| 12:28.10 | sulex | tzafrir: cheers mate |
| 12:29.29 | *** join/#asterisk scardinal (~supreme@0905ds1-rdo.0.fullrate.dk) |
| 12:30.05 | tzafrir | sulex, have you looked into patching the logging functions of asterisk? |
| 12:30.17 | tzafrir | main/logger.c , basically |
| 12:31.01 | sulex | tzafrir: not yet in effect, maybe it's faster than play with ioctl or upgrade to 1.6 |
| 12:31.15 | tzafrir | I don't suppose you can get a real micro-second granularity, but at least something close to ms granularity |
| 12:31.16 | sulex | tzafrir: yeah i think it's easier. thanks for the suggestion |
| 12:31.39 | sulex | tzafrir: yes indeed... for sure better than the "second" i alway get ;) |
| 12:32.05 | fleebailey33 | is asterisk a good place to start, what i want to do is call myself every morning, sorta like an alarm |
| 12:32.38 | sulex | fleebailey33: yep, use callfiles |
| 12:33.00 | fleebailey33 | sulex, thanks for pointing the direction |
| 12:33.01 | fleebailey33 | :) |
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| 12:39.52 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:50.07 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 12:53.27 | ijpalmer | hi I'm trying to install asterisk 1.4.20 on a fedora 12 box, I've installed all the prerequisites but when I try to install the wanpipe-3.4.9 drivers I get a message saying SMG Failed to install. I've posted a pastebin here http://pastebin.com/fpbbDAVA any help would be appreciated |
| 12:54.43 | Chainsaw | Does hint support callgroups/pickupgroups at all? |
| 12:55.17 | Chainsaw | ijpalmer: It's asking you to contact Sangoma support though. That's not us. |
| 12:55.48 | Chainsaw | ijpalmer: (The shared library "libsangoma" appears to be missing) |
| 12:55.59 | ijpalmer | chainsaw: I've emailed them but they're on a different time zone to us, just wondered if anyone here had any ideas that's all |
| 12:57.51 | ijpalmer | thanks chainsaw, how do i check this and add it if necessary, I'm no linux expert sorry |
| 12:59.17 | Chainsaw | ijpalmer: Is there a sangoma library in your list of prerequisites? |
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| 13:03.13 | ijpalmer | chainsaw: i followed the prereqs from the sangoma site here http://wiki.sangoma.com/Requirements and couldn't see this mentioned |
| 13:04.04 | Chainsaw | ijpalmer: And you ran yum install lksctp-tools-devel? |
| 13:05.12 | fleebailey33 | sulex, still around? |
| 13:05.40 | fleebailey33 | would i need any additional hardware besides of course a computer setup in order to use callfiles? |
| 13:07.01 | SystemT` | ~itsp |
| 13:07.01 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 13:08.17 | ijpalmer | chainsaw: yeah I didn't have it installed at first but it throws up a message asking for it |
| 13:08.47 | Chainsaw | ijpalmer: Alright. Nothing obvious I can think of then. |
| 13:09.01 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 13:09.19 | ijpalmer | chainsaw: ok thanks anyway, guess I'll have to wait until Sangoma start work |
| 13:09.31 | wonderworld | hi, could someone point me to information on what SMDI is? |
| 13:09.57 | wonderworld | i need to debug an asterisk box, the logs are filled with "No SMDI interfaces are available to listen on, not starting SDMI listener" |
| 13:10.33 | wonderworld | is it SDMI or SMDI? |
| 13:12.01 | coppice | simple messy desk interface |
| 13:13.09 | tzafrir | smdi |
| 13:13.21 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 13:13.24 | tzafrir | res_smdi.so, which is a dependency of some other modules |
| 13:13.38 | wonderworld | thanks, do you know what it does? |
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| 13:14.08 | tzafrir | wonderworld, if you don't know what it does, you don't need it |
| 13:14.47 | wonderworld | i am debugging an asterisk box and the logs are filled with smdi-errors |
| 13:15.05 | wonderworld | so i just want to know what it's good for, because i never stumbled upon it before |
| 13:16.07 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:16.18 | wonderworld | is this the right definition for it -> http://www.webopedia.com/TERM/S/SMDI.html ? |
| 13:17.53 | sulex | fleebailey33: where are you going to receive the call? |
| 13:17.58 | Chainsaw | wonderworld: No, try: http://en.wikipedia.org/wiki/SMDI |
| 13:18.09 | fleebailey33 | sulex, USA |
| 13:18.25 | sulex | fleebailey33: on a mobile phone? |
| 13:18.36 | fleebailey33 | maybe, maybe not |
| 13:19.23 | *** join/#asterisk seba (~seba@p57BDE1DC.dip.t-dialin.net) |
| 13:19.40 | seba | hi |
| 13:21.06 | fleebailey33 | why? |
| 13:21.21 | fleebailey33 | sulex, answer is probably, a mobile, but how does that matter? |
| 13:21.21 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23) |
| 13:21.34 | sulex | fleebailey33: the easiest way that comes to my mind. 1) set up a VoIP service with a provider (no clue on who in the US) 2) setup asterisk to place calls over SIP 3) create a sound file with some sort of alarm "wake the fuck up it's time to to work" 4) create a callfille to setup a call to your "maybe, maybe not" 5) create a cronjob that place the call files in $spooldir/outgoing at a given our |
| 13:22.24 | fleebailey33 | sounds easier enough |
| 13:22.25 | *** join/#asterisk caim (~user@unaffiliated/caim) |
| 13:22.28 | seba | i have 4 computers behind a nat, all of them connected to a vpn, which i want to use one sip account to dial out to normal telephones. i want to setup a server on my vpn gateway. is asterisk the right thing for me or would something like siproxd/openser better for me? |
| 13:22.32 | fleebailey33 | the last 3 i already planned on doing |
| 13:23.39 | sulex | fleebailey33: the destination of the calls matters, as if it's just a VoIP utency you just need asterisk. if it's mobile or PSTN and you are not connected to the telephone line with your asterisk box the easier way is to subscribe to a VoIP provider |
| 13:23.41 | tzafrir | sulex, s/crond/atd/ ? |
| 13:24.10 | sulex | tzafrir: that just depends if you want the call placed on a regular basys or not |
| 13:24.24 | fleebailey33 | makes sense |
| 13:24.38 | sulex | tzafrir: cron to go to job every morning, at if you go just once in a while :P |
| 13:25.15 | tzafrir | right |
| 13:26.51 | [TK]D-Fender | seba: depends on what else you might want to do other than just proxy out to your ITSP |
| 13:27.47 | seba | at the moment, not much.. just use all my computers from all over the world connected to my vpn as my normal home telephone |
| 13:28.12 | [TK]D-Fender | seba: Which has nothing to do with phones. You asked about * |
| 13:28.54 | [TK]D-Fender | seba: So if the only thing you want to do is dumb-proxy a bunch of phones as registered to a single account, then sipproxd might be fine for you. SER is probably overkill. |
| 13:29.20 | [TK]D-Fender | seba: however if you ever think of doing more, then * iclearly offers a lot |
| 13:29.59 | seba | hm okay, thx, then I'll take a look at siproxd and come back to * if i grow out of it ;) |
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| 13:35.08 | Skeeter- | [TK]D-Fender, in the same way of seba, or maybe not, if i have a 500+ extension and i would like to get a sip proxy, SER would stil be overkill, i nvr heard pf siproxd |
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| 13:38.44 | [TK]D-Fender | Skeeter-: SER wouldn't be considered overkillin that case |
| 13:38.55 | *** join/#asterisk seba (~seba@p57BDBF5E.dip.t-dialin.net) |
| 13:39.10 | Skeeter- | [TK]D-Fender, ok thanks |
| 13:39.45 | *** part/#asterisk _saya (~kamanashi@202.79.19.96) |
| 13:41.00 | *** join/#asterisk WinZ (~winz@82.146.61.218) |
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| 13:45.48 | WinZ | guys, my Asterisk server has a static public IP (207.42.130.*). My SIP phone is on the same network as Asterisk (192.168.0.0). In [general] I put nat=no and canreinvite=yes. In [mysipphone] I put nat=yes and canreinvite=no. Am I doing it right? |
| 13:46.18 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
| 13:47.40 | tzafrir | WinZ, set externip and localnet |
| 13:47.49 | tzafrir | (see the sample ocnfig file) |
| 13:48.02 | *** join/#asterisk mbranca (~matteo@host139-217-static.224-95-b.business.telecomitalia.it) |
| 13:48.06 | tzafrir | (sip.conf , that is) |
| 13:49.44 | WinZ | yes, it's set, good |
| 13:49.50 | WinZ | For my trunks, can I also use nat=no and canreinvite=yes? |
| 13:50.06 | [TK]D-Fender | win [mysipphone] I put nat=yes (doesn't matter) and canreinvite=no (always good) |
| 13:50.22 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 13:50.59 | WinZ | [TK]D-Fender, so, it's always good to process voice through asterisk, right? |
| 13:51.27 | *** join/#asterisk mattwilliams (~mattwilli@s15365979.onlinehome-server.com) |
| 13:51.47 | hluesea | HEllo |
| 13:52.17 | mattwilliams | Hello, does asterisk have ASR/TTS support? Or do you suggest to use something like Loquendo, LumenVox, etc? |
| 13:52.22 | hluesea | how can i collect incoming calls in cdr (i can see only internal numbers flow in cdr) ? |
| 13:53.49 | [TK]D-Fender | mattwilliams: No, * does not. 3rd party add-ons offer this however |
| 13:53.55 | hluesea | mean internal numbers dialled and used numbers in cdr db |
| 13:54.02 | mattwilliams | [TK]D-Fender: Thanks |
| 13:54.40 | elred_ | WinZ, you lost performance if you process voice |
| 13:54.50 | [TK]D-Fender | hluesea: They do hit CDR |
| 13:55.01 | elred_ | but in nat scenario, i don't see any other solution |
| 13:55.38 | [TK]D-Fender | wizYou can allow reinvites only with your same-lan phones, and block them everywhere else, and in general if you like |
| 13:55.44 | [TK]D-Fender | WinZ: ^ |
| 13:56.08 | [TK]D-Fender | WinZ: But I'd just as soon prevent any issues and run everything through it |
| 13:56.25 | WinZ | [TK]D-Fender, thanks a lot |
| 13:58.56 | Kobaz | everybody wants a rock to wind a piece of string around |
| 14:01.22 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
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| 14:08.15 | jblack | kobaz: I wonder what happened to chess piece face. |
| 14:08.43 | *** join/#asterisk ACiDV (~acidv@97-170.dr.cgocable.ca) |
| 14:08.47 | Kobaz | ? |
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| 14:12.06 | ACiDV | Anyone have experience with SIP "call simulator" (ex. sipp, pbx-test (atis)) ? What I need to do is simulate a SIP REFER (blind xfer), any hints ? |
| 14:13.24 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
| 14:13.49 | *** join/#asterisk petern_ (~petern@lachesis.fuzzle.org) |
| 14:14.41 | hluesea | <[TK]D-Fender> as if it so i have a problem on my cdr. I am looking into cdr db but i can't see any incoming calls in it. I chechked all cdr conf files but only internal calls storaging |
| 14:15.04 | [TK]D-Fender | hluesea: I don't see you showing us anything.... |
| 14:15.53 | hluesea | <[TK]D-Fender> ok i am pasting bin, sorry for that :) |
| 14:19.57 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
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| 14:23.33 | petern_ | i'm using fastagi. is there a way to specify where the digit sounds come from within the script? |
| 14:23.45 | petern_ | (or beforehand) |
| 14:24.15 | [TK]D-Fender | petern_: Huh? |
| 14:24.32 | *** join/#asterisk pawz (~pawz@ppp118-208-191-223.lns20.bne4.internode.on.net) |
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| 14:27.56 | tzafrir | petern_, "digit sounds" are normal sound files |
| 14:28.23 | tzafrir | digit/<NUM> , normally |
| 14:29.40 | petern_ | yes, and i'd like to be able to play back from say custom/script1/digit/ in one script and custom/script/2/digit/ in another, for example |
| 14:31.21 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 14:31.22 | [TK]D-Fender | petern_: You can explicitly playback whatever sound file you want |
| 14:31.23 | hluesea | my cdr cof files, |
| 14:31.24 | hluesea | <PROTECTED> |
| 14:31.57 | hluesea | and i have correct db connection informations on cdr.conf and right cdr table on database |
| 14:32.20 | petern_ | yes, but then i'd have to reimplement SAY DIGITS and SAY NUMBER |
| 14:32.34 | tzafrir | petern_, you could play ugly games with LANGUAGE |
| 14:32.46 | tzafrir | and maybe some extra symlink tricks |
| 14:33.32 | tzafrir | petern_, but why do you need that? |
| 14:33.50 | petern_ | i need localised digits |
| 14:34.07 | petern_ | by localised i mean things like local accents, not different languages |
| 14:34.14 | tzafrir | then it's not really a dirty trick with LANGUAGE |
| 14:34.22 | petern_ | each localisation would be per-script |
| 14:34.25 | petern_ | i guess not :) |
| 14:34.32 | tzafrir | put the sounds under sounds/YOUR_LANGUAGE |
| 14:34.35 | [TK]D-Fender | petern_: then go download some from someone who has recorded them the way you like |
| 14:34.40 | petern_ | language=westcountryfarmer :D |
| 14:34.46 | WinZ | [TK]D-Fender, just one more question please.. canreinvite option in a trunk section -- does it affect only asterisk<->trunk connection? and, same, canreinvite in a friend section -- does it affect only friend<->asterisk? |
| 14:34.57 | *** join/#asterisk smooth_penguin (~smoove@59.95.28.192) |
| 14:35.17 | [TK]D-Fender | WinZ: if 2 ends fo a acll allow it, * will try. If either says no, then it won't |
| 14:35.52 | tzafrir | petern_, the convention for language directory names is <lang>_<COUNTRY>_<m/f>_<name> |
| 14:36.10 | WinZ | [TK]D-Fender, thumbs up! |
| 14:36.36 | tzafrir | The <lang>_ prefix is significant, because Asterisk will fall back to look under <lang> if it cant find it in <lang>_* |
| 14:36.48 | *** join/#asterisk garymc (~chatzilla@host81-148-109-86.in-addr.btopenworld.com) |
| 14:37.14 | tzafrir | So you can e.g. keep the system ones under en/ and put your own under en_<something> |
| 14:37.24 | Skeeter- | asterisk 1.6.2, stop now got changed or sth? |
| 14:37.24 | petern_ | sounds good |
| 14:37.34 | tzafrir | Just make sure you set LANGUAGE to the same en_<something> |
| 14:37.52 | tzafrir | Skeeter-, core stop now |
| 14:38.02 | Skeeter- | tzafrir, thanks |
| 14:41.53 | tzafrir | Skeeter-, feel free to edit /etc/asterisk/cli_aliases and rem-out template=asterisk14 |
| 14:42.25 | Skeeter- | tzafrir, wow, thanks that is good to know |
| 14:42.41 | tzafrir | hmm... unrem, that is |
| 14:42.57 | [TK]D-Fender | Skeeter-: You might reading the upgrade docs. |
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| 15:15.19 | Skeeter- | Can you make press 0 from a queue(just like voicemail) to reach the operator? |
| 15:15.47 | Kobaz | sure... set context in queues.conf |
| 15:17.03 | Skeeter- | context=ivr-3 |
| 15:17.26 | *** join/#asterisk ManxPower-work (~manxpower@36.sub-75-216-1.myvzw.com) |
| 15:17.52 | Kobaz | Skeeter-: look at the queues.conf.sample |
| 15:18.24 | *** join/#asterisk radic (~radic@p4FDAABB9.dip0.t-ipconnect.de) |
| 15:18.33 | radic | hi |
| 15:18.47 | [TK]D-Fender | Kobaz: ... FreePBX <- |
| 15:19.00 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 15:19.06 | radic | I get "407 Proxy Authentication Required" from an SIP-server on every outgoing call |
| 15:19.06 | Kobaz | ? |
| 15:19.37 | Kobaz | radic: authenticate with your proxy then |
| 15:19.55 | radic | Kobaz: and how I do that? |
| 15:20.02 | Kobaz | read the documentation in sip.conf |
| 15:20.28 | [TK]D-Fender | radic: That behaviour is perfectly normal |
| 15:20.40 | ManxPower-work | Yup. |
| 15:21.22 | Kobaz | 4xx is an error though |
| 15:21.52 | Kobaz | radic: are you having problems placing calls? or you just get that item in your log? |
| 15:22.33 | radic | Kobaz: I get that error if I do an call |
| 15:22.43 | Kobaz | 'a call' |
| 15:22.47 | Kobaz | not 'an call' |
| 15:22.56 | ManxPower-work | pastebin just the Dial line you are seeing in the CLI. |
| 15:23.11 | Kobaz | an is only used when you have a vowel as the first letter of the next word |
| 15:23.17 | Kobaz | or a vowel sound |
| 15:23.29 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 15:24.53 | radic | Kobaz: a, e, i, o, u? |
| 15:25.04 | Kobaz | sounds about right |
| 15:26.32 | Kobaz | a call, a bird, a floor, a pbx, a tkfender... an apple, an orange, an independant contractor |
| 15:29.17 | beek | a pain in the ass |
| 15:29.22 | beek | :D |
| 15:29.42 | Kobaz | totally |
| 15:29.57 | coppice | an eye for an eye, a tooth for a tooth |
| 15:30.06 | beek | coppice++ |
| 15:32.45 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 15:33.26 | radic | and "an Iamnacho"?! |
| 15:33.32 | radic | hmm |
| 15:34.16 | Iamnacho | hi :) |
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| 15:36.03 | Kobaz | radic: "a tkfender" was a joke... but yeah that would be kinda correct |
| 15:38.24 | ManxPower-work | I thougbht it was about 1 million dollars for an eye these days |
| 15:38.53 | Skeeter- | sorry to everyone |
| 15:39.02 | Skeeter- | i didnt notice i was in the asterisk channel |
| 15:40.03 | Kobaz | Skeeter-: what asterisk channel? |
| 15:40.17 | tzafrir | Skeeter-, this is a * channel |
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| 15:41.24 | tzafrir | It seems everything is on-topic. Except GUIs |
| 15:42.50 | [TK]D-Fender | tzafrir: Correct! |
| 15:43.24 | ManxPower-work | Skeeter-, Stop lying. You know perfectly well this is not a GUI channel. |
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| 15:45.04 | Skeeter- | ManxPower-work, i know that isnt the gui channel, but i didnt notice i was typing in the asterisk channel instead of the gui channel |
| 15:45.23 | ManxPower-work | Skeeter-, Oh! Ok. sorry. |
| 15:46.11 | Skeeter- | ManxPower-work, usually if i ask a question here, i always ask if it is gui related or no first, before wasting ppl time |
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| 15:48.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:53.20 | radic | http://pastebin.com/TsPz3MYa |
| 15:53.21 | radic | http://pastebin.com/QB2n0JkH |
| 15:53.58 | radic | where is the problem? |
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| 15:58.36 | sbrath | How can I allow someone to get more than 1 call if it's a DID call to their extension, but have Queue() consider them busy if they are InUse ? |
| 15:58.59 | [TK]D-Fender | radic: problem is you aren't showing us the complete call |
| 15:59.16 | [TK]D-Fender | sbrath: * will call whatever you tell it to. |
| 15:59.57 | [TK]D-Fender | sbrath: If you want * to check if their on their phone first, then call ChanIsAvail beforehand. If you just want to check after they fail to answer for whatever reason, then do that. |
| 16:00.06 | [TK]D-Fender | sbrath: either way this is all up to you and your dialplan |
| 16:00.28 | radic | http://pastebin.com/k1Mu1kcZ |
| 16:00.54 | sbrath | But the Call is being delivered to the person inside the Queue App |
| 16:01.01 | [TK]D-Fender | radic: Including verbose 10 <_ |
| 16:01.09 | radic | yes |
| 16:02.27 | sbrath | strategy strict dosen't consider InUse as Unavailable since call-limit is higher than 1 |
| 16:02.39 | [TK]D-Fender | radic: insecure=no <- |
| 16:03.15 | [TK]D-Fender | sbrath: Feel free to show an actual sample call with CLI status dumps & configs to match |
| 16:03.45 | sbrath | cli status dumps ? |
| 16:03.52 | sbrath | I have verbose to 10 |
| 16:04.07 | hardwire | crank it to 11 |
| 16:04.08 | radic | [TK]D-Fender: same as before |
| 16:05.34 | [TK]D-Fender | radic: They are also be "nat=no" |
| 16:07.02 | tzafrir | hardwire, I don't think there's any message with verbose level > 5 |
| 16:07.47 | hardwire | aww |
| 16:07.58 | hardwire | 11 should be a magic number that starts shouting things at you |
| 16:08.21 | *** join/#asterisk tkrn (~tkrn@WS1-DSL-208-102-253-13.fuse.net) |
| 16:08.59 | radic | changed bothe settings but still "407 Proxy Authentication Required" |
| 16:12.04 | [TK]D-Fender | radic: Comment out the fromdomain |
| 16:13.14 | radic | nothing changed |
| 16:15.28 | [TK]D-Fender | radic: You're trying to use * witha provider that generally insists on your using their ATA, right? |
| 16:16.07 | [TK]D-Fender | radic: Also, reverse you dial. Should be Dial(SIP/peer/number) |
| 16:16.20 | tzafrir | hardwire, actually you're right. it goes up to 11: |
| 16:16.23 | tzafrir | http://pastebin.ca/1830188 |
| 16:16.46 | P424D0X | How I must configure Asterisk, so that a CISCO 7940 works in own subnetz behind the Asterisk server. |
| 16:17.35 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:18.41 | tzafrir | to complete the picture: debug levels as well: http://pastebin.ca/1830191 |
| 16:18.45 | Naikrovek | P424D0X: have asterisk listen on 0.0.0.0 |
| 16:19.29 | idespinner | P424D0X, is this with NAT or no NAT? |
| 16:19.33 | P424D0X | Naikrovek: The connect to the sipgate.de does work.. |
| 16:20.05 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:21.08 | timeshell | Greetings. Is there a way to see which files are being served by phoneprov? |
| 16:22.31 | P424D0X | Here is my network: Linksys (192.168.0.1) listen on 0.0.0.0, PC3 (Asterisk-Server) with two ethernet-devices (eth0: 192.168.0.3 connected to 192.168.0.1) and (eth1: 192.168.1.1) connected to Cisco 7940 (192.168.1.30).. Netmask is /24.. |
| 16:23.25 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 16:24.47 | *** part/#asterisk petern_ (~petern@lachesis.fuzzle.org) |
| 16:26.22 | P424D0X | I can call the asterisk-server from my Phone, but there is not outgoing and incoming calls from sipgate.de.. My local asterisk-Server is connected to sipgate.de |
| 16:26.39 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:27.26 | *** part/#asterisk ManxPower-work (~manxpower@36.sub-75-216-1.myvzw.com) |
| 16:27.37 | idespinner | P424D0X, can the cisco phone sucessfully route out to the internet? you may not have the routes on both sides of your network |
| 16:28.27 | *** join/#asterisk DJF5 (~email@84-105-183-83.cable.quicknet.nl) |
| 16:28.28 | *** join/#asterisk SAngeli (~SAngeli@host242-77-static.82-94-b.business.telecomitalia.it) |
| 16:30.05 | [TK]D-Fender | idespinner: It shouldn't have to. The phone has no reason to touch the internet. |
| 16:30.15 | [TK]D-Fender | idespinner: * is in between |
| 16:30.35 | P424D0X | I have not tried out this yet. How do I do this?! |
| 16:30.55 | [TK]D-Fender | P424D0X: Haven't tried what? |
| 16:31.16 | Gugge | do i need AGI if i want to run a bash script from the dialplan, and parse the output |
| 16:31.24 | Kobaz | bash? |
| 16:31.26 | Kobaz | why bash? |
| 16:31.33 | Gugge | then whatever script :) |
| 16:31.34 | Kobaz | but yeah... i suppose you could |
| 16:32.10 | Gugge | in this case i just need to know if theres files in a given directory |
| 16:32.27 | P424D0X | idespinner: route out to the internet from my phone.. |
| 16:32.35 | Gugge | i guess i can just check the returncode from system() |
| 16:32.59 | Kobaz | well, keep in mind that program return codes are a short, so it's 0-255 |
| 16:33.29 | *** join/#asterisk dmast (~dmast@exchange.newpointe.org) |
| 16:33.29 | Kobaz | at least on linux anyway |
| 16:33.56 | *** join/#asterisk freezey (~trees@static-64-61-84-174.isp.broadviewnet.net) |
| 16:34.03 | Gugge | yes, thats why i wanted a more general sollution, where i could parse stdout from the script |
| 16:34.10 | Kobaz | but you could return a 1/0 and check the result in dialplan |
| 16:34.10 | hardwire | tzafrir: lol |
| 16:34.59 | P424D0X | How can i test, that my phone route out to the internet.. |
| 16:35.24 | freezey | i have my phone company saying that we currently have 2 T1's running into our asterisk server 1 for in 1 for out. So i took a look in the phone room and the cisco IAD has 2 t1 connections going into my digium card. However their is only 1 HTUR card on the wall. so how is it possible to have 2 t1's if their is only 1 HTUR card? |
| 16:35.32 | idespinner | P424D0X, not 100% familiar with cisco, but I'm pretty sure you can telnet to them |
| 16:35.32 | Nugget | telnet is eeeeeeevil! |
| 16:35.34 | freezey | so i am thinking its not possible |
| 16:35.55 | [TK]D-Fender | P424D0X: again why is your phone going directly out to the internet? |
| 16:35.59 | tzafrir | Gugge, use TrySystem ? |
| 16:36.44 | Gugge | tzafrir: that also only returns the SYSTEMSTATUS var |
| 16:37.24 | tzafrir | Gugge, you want the equivalent of var=`command` ? |
| 16:37.26 | P424D0X | [TK]D-Fender: My phone is not able yet to route in internet. |
| 16:37.36 | Gugge | tzafrir: something like that yes |
| 16:37.38 | tzafrir | Try the function SHELL |
| 16:37.54 | [TK]D-Fender | P424D0X: why should it? * should be doing that |
| 16:37.57 | Gugge | ohh, that seems right |
| 16:38.31 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 16:38.52 | radic | [TK]D-Fender: now I get a 403 forbidden |
| 16:38.54 | freezey | [TK]D-Fender: you usually have all answers what do you think about that? |
| 16:39.18 | P424D0X | [TK]D-Fender: I thought that all that the asterisk server does. |
| 16:39.31 | Kobaz | his answer-machine is getting hungry |
| 16:39.36 | [TK]D-Fender | freezey: No idea |
| 16:39.56 | [TK]D-Fender | P424D0X: Phone talks to *. * talks to internet. Therefor the phone does not talk to the internet. |
| 16:40.00 | freezey | well is it possible for 1 HTUR card to handle multiple t1's? |
| 16:40.04 | [TK]D-Fender | P424D0X: So what are you going on about? |
| 16:40.09 | P424D0X | [TK]D-Fender: Does go also without which is connected my phone directly with the Internet?! |
| 16:40.16 | Kobaz | what the heck is a htur card |
| 16:40.43 | [TK]D-Fender | freezey: HDSL can eb > 1.5mbps, no? |
| 16:41.00 | freezey | yes |
| 16:41.09 | freezey | thats just 1 t1 lie |
| 16:41.11 | freezey | line |
| 16:41.42 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:41.42 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:41.53 | P424D0X | [TK]D-Fender: Thus I would like to have this. Only this does not function surely. I can reach the asterisk server, but not be called or call myself. |
| 16:42.31 | [TK]D-Fender | freezey: Well the IAD could be used for a link supporting more channels overall and just pumping out multiple T1's to span. If thats appropriate to your actual service... well that's another matter |
| 16:42.45 | [TK]D-Fender | P424D0X: From where, to where? |
| 16:43.57 | freezey | [TK]D-Fender: so even though the HTUR cards have 1 handoff your saying it could still be pumping more bandwidth and then just split it at the IAD? |
| 16:44.45 | [TK]D-Fender | freezey: could be. |
| 16:44.58 | freezey | highly unlikely though |
| 16:45.16 | freezey | let me check into a few things |
| 16:45.49 | Gugge | check if you can make more calls than a single t1 can handle, if you can ... you have more than one :P |
| 16:45.49 | P424D0X | I can call my * directly.. But i havn't connection to pstn.. My SIP-Gate is sipgate.de.. I can see, the * is logged on, but i can't do outgoing and incoming calls from and to pstn.. |
| 16:46.15 | Gugge | P424D0X: and the errors logged in the console when you try are pasted where? |
| 16:46.39 | P424D0X | Gugge: Moment.. |
| 16:48.24 | [TK]D-Fender | PASTEBIN! |
| 16:48.26 | [TK]D-Fender | ~pb |
| 16:48.27 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 16:48.29 | [TK]D-Fender | P424D0X: ^^^^^^^^^^^^6 |
| 16:49.07 | P424D0X | [TK]D-Fender: 6?! |
| 16:49.17 | [TK]D-Fender | ... |
| 16:50.24 | P424D0X | [TK]D-Fender: I can see a message: "SIP/2.0 404 Not found" |
| 16:51.26 | [TK]D-Fender | P424D0X: perhaps you should pastebint he entire call with SIP DEBUG so we can tell you where you wen't wrong |
| 16:54.30 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
| 16:59.19 | *** join/#asterisk korihor (~korihor@190.205.251.97) |
| 17:03.11 | P424D0X | [TK]D-Fender: http://pastebin.com/5A6N749v |
| 17:04.39 | P424D0X | [TK]D-Fender: Should I post my configs?! |
| 17:06.25 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 17:06.57 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 17:07.05 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 17:12.48 | Kobaz | configs are always useful |
| 17:17.06 | *** join/#asterisk SAngeli (~SAngeli@host242-77-static.82-94-b.business.telecomitalia.it) |
| 17:19.33 | P424D0X | Kobaz: Is added.. |
| 17:20.10 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 17:20.53 | titter | Is there a way to handle warnings/result codes in my dialplan when using app_fax |
| 17:21.16 | *** join/#asterisk hfb (~hfb@pool-96-247-108-157.lsanca.dsl-w.verizon.net) |
| 17:23.05 | kfife | If I am IAX trunking between 2 Asterisk instances, and ultimately connecting to SIP endpoints on BOTH ends of the call, can I let the ENDPOINTS do ALL the jitterbuffering (even though I'm switching protocols), or must the iax-trunk do its own jitterbuffering? I know the answer is yes if it's all IAX all the way through. |
| 17:24.39 | kfife | I'm actually asking whether voice frames are being 'repackaged' by asterisk on new time bounaries when switching from IAX to SIP. If so, it would seem that iax would need to do ALL of its own |
| 17:24.39 | kfife | jitterbuffering to prevent incremental losses from out-of-order packets. |
| 17:25.51 | [TK]D-Fender | P424D0X: Line 154 : Contact: <sip:2473867@192.168.0.3> <----------- You clearly have not set your * up to work properly from behind NAT. Go follow the guide : |
| 17:25.53 | [TK]D-Fender | ~sipnat |
| 17:25.53 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:25.55 | [TK]D-Fender | ^^^^^^^^ |
| 17:27.51 | P424D0X | [TK]D-Fender: Okay.. Many thanks.. I'll look the guide.. |
| 17:33.51 | titter | nvm, found the variables being set by app_fax |
| 17:37.30 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
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| 17:59.10 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 18:02.36 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 18:03.57 | ariel_ | Hello everyone |
| 18:06.34 | *** join/#asterisk mikealeonetti (~mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
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| 18:10.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 18:11.11 | *** join/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 18:11.20 | moos3 | how can I drop a connection to MeetMe |
| 18:12.28 | afink | Hello everyone, I am having an issue where audio is getting distorted after about 3 minutes of being on the phone. I have a SIP connection to the PSTN. Could someone please give me some suggestions on what to do to resolve? |
| 18:16.53 | jaskew | afink: What codec are you using (e.g. ulaw, G.729, etc) and what does the distortion sound like (e.g. dropouts, gurgling, etc)? |
| 18:17.19 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:91:49d9:2ac0:762d) |
| 18:17.20 | cusco | hi |
| 18:17.42 | jaskew | afink: and is it always 3 minutes into the call, or does it vary? Has it ever started up right away? |
| 18:17.52 | *** join/#asterisk b14ck (~comradeb1@cpe-24-24-136-239.socal.res.rr.com) |
| 18:18.00 | afink | I am not using any specific codec, I let * chose on its own. My provider says they do not use g.729 b/c of the distortion. Someone here suggested I try g.729 across the board, but I could not get that to work |
| 18:19.02 | afink | jaskew: No, it is not always 3 minutes that is a rough average, and sometimes it doesn't happen at all. |
| 18:19.57 | jaskew | afink: What does the distortion sound like. Is it dropouts, noice or garbling? |
| 18:20.02 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
| 18:20.17 | jaskew | s/noice/noise |
| 18:20.34 | afink | I haven't heard for myself, but the best description would probably be garbling |
| 18:21.11 | jaskew | afink: so the distortion is heard on the far end, but you hear the other person fine? |
| 18:21.26 | *** part/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 18:21.44 | afink | I'm not local, the distortion is on both ends and the customer says "It is very delayed" |
| 18:22.22 | jaskew | afink: what kind of internet connection do they have. How is the asterisk box connected - is it though a router? |
| 18:22.23 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:23.00 | afink | They have 3mb Metro Ethernet from the sip provider |
| 18:24.00 | afink | It is connected on the lan, the lan goes through a firewall out to the provider |
| 18:25.12 | afink | try some quality control on the firewall? |
| 18:25.40 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 18:27.23 | [TK]D-Fender | afink: On a single call will it happen regardless? What other codecs have you etsted? |
| 18:27.34 | [TK]D-Fender | afink: This sounds like a "ITSP=suck" issue |
| 18:27.47 | afink | hmm, that would suck |
| 18:28.47 | jaskew | afink: sorry - got a call. |
| 18:29.04 | afink | [TK]D-Fender: no, it will not, it is very sporadic. I haven't tested any other codecs. Someone told me long ago that * would choose the best codec. |
| 18:29.26 | jaskew | afink: I was thinking maybe a QoS issue, but if the SIP provider is handling the interent connection, they *should* have that covered. |
| 18:29.43 | afink | maybe I should check with them |
| 18:29.54 | [TK]D-Fender | afink: No, * will settle on the fist common match and always pick the same regardless if each side allows 10 dcodes |
| 18:30.21 | afink | ok, any suggestions on which ones work best? |
| 18:30.36 | jaskew | afinnk: DS0 |
| 18:30.52 | Skeeter- | What would be the best solution for a customer that got multiple AP, and when he walks a long way with his phone, switches AP, drop the call. Is there any way to counter that? |
| 18:31.12 | jaskew | sorry - that was an attempt at humor |
| 18:32.16 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 18:32.46 | jaskew | afink: ulaw (or alaw) is probably best quality, but has 0 tolerance for bandwidth limitation or latency |
| 18:33.20 | jaskew | afink: other compressed codecs have various tradeoffs btwn bandwidth & quality. |
| 18:34.06 | jaskew | [TK]D-Fender: any sage advice on codec quality? |
| 18:34.39 | sbrath | [TK]D-Fender: I figured out the Queue() problem with it ringing inuse, it seems that in 1.6.2 that ringinuse default is true, and even thou I had it configured in general as ringinuse=no, it didn't take.. I added it to the specific queue, and it's perfect. |
| 18:35.26 | afink | ok, well looking at my firewall graphs they almost never hit 3 mb so I could give alaw and ulaw a shot |
| 18:36.03 | jaskew | afink: Is it symmetric - 3Mb up and down? |
| 18:36.10 | afink | yes |
| 18:36.11 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:37.12 | jaskew | afink: OK. Keep in mind it can be latency/jitter too. A full size packet takes a certain amount of time to transmit. Even with perfect QoS, you can end up with a good amount of jitter |
| 18:37.45 | jaskew | I think Cisco has some trickery to fragment packets and keep the jitter down. Not sure about other routers |
| 18:38.54 | jaskew | Skeeter: You might need APs that support roaming (I think that's the term) |
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| 18:43.58 | mmj_nix | what SIP data should an ata be sending for hook-flash ? |
| 18:45.23 | *** join/#asterisk cuco (~elcuco@IGLD-84-228-194-95.inter.net.il) |
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| 18:47.37 | jaskew | mmj_nix: I think that depends on the ata. I wouldn't think it would send anything until you invoke a feature. I could be wrong tho - I'm somewhat new. |
| 18:49.26 | mmj_nix | <PROTECTED> |
| 18:51.41 | jaskew | mmj_nix: what happens when you flash - do you get a second dial tone? |
| 18:53.01 | mmj_nix | <PROTECTED> |
| 18:53.37 | [TK]D-Fender | mmj_nix: Go read its manual |
| 18:54.42 | jaskew | mmj_nix: As far as I know (and I may be totally incorrect) there is no SIP code for a hook-flash. Now, after you flash and dial another number and flash again, the ATA should send the commands to effect a transfer. |
| 18:54.52 | jaskew | for instance |
| 18:55.12 | mmj_nix | [TK]D-Fender: did that, it's so lame you would not believe it, it says "see your provider for feature support" :( |
| 18:55.37 | [TK]D-Fender | mmj_nix: Quality item... |
| 18:56.03 | *** join/#asterisk Madoc (~Madoc@bas8-toronto63-1177708113.dsl.bell.ca) |
| 18:56.57 | Madoc | I'm trying to develop an application that be controlled via AMI, it sounds like UserEvent is what I want, I can see easily how to send them via AMI, but where can I find some documentation on how to recieve them in the application? |
| 18:59.11 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 18:59.36 | mmj_nix | [TK]D-Fender: too true LOL ;), sadly I have 10 of these so getting it figured out is still a worthwhile cause. The ATA's web config page has SIP "calling features" area for hold, flash, transfer, conference. They are currently blank. |
| 19:01.58 | mmj_nix | jaskew: so what does Asterisk use for hook-flash if not SIP on a SIP channel? |
| 19:02.13 | *** join/#asterisk jconnolly (~jconnolly@firebug.buglabs.net) |
| 19:04.20 | [TK]D-Fender | mmj_nix: no such thing for * |
| 19:07.41 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
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| 19:13.00 | *** join/#asterisk WhIteSidE (~chatzilla@uawifi-nat-210-16.arizona.edu) |
| 19:13.08 | WhIteSidE | Hello all, I've been fighting with app_swift for the past several hours, and I simply cannot get it to compile. I've tried patching, etc, etc. What's the current "best way" to get cepstral working asterisk 1.6 |
| 19:13.14 | WhIteSidE | ? |
| 19:14.02 | mmj_nix | [TK]D-Fender: you mean no such thing as feature codes for those events in *? |
| 19:14.21 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
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| 19:17.15 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
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| 19:21.56 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:22.22 | jaskew | mmj_nix: Maybe this will help - Think about dialing a number. You pick up the phone and dial 10 (+/-) digits. SIP does not send messages for going off-hook and each digit. It only sends one message (INVITE) once the dialing is complete |
| 19:23.26 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 19:25.11 | jaskew | mmj_nix: So it's the same kind of thing with a flash. The ATA doesn't send anything until you've activated a feature. If it's call waiting, then the flash itself is sufficient - the ATA sends an OK to answer the other line |
| 19:25.31 | jaskew | mmj_nix: and whatever other message(s) to put the first call on hold. |
| 19:26.16 | jaskew | mmj_nix: In the case of a transfer or conference, no messages are sent until the ATA has collected all of the digits and there is something to do |
| 19:26.27 | *** join/#asterisk Dibri (~gavit@pop1.isgroup.sr) |
| 19:27.43 | mmj_nix | <PROTECTED> |
| 19:29.37 | *** join/#asterisk WhIteSidE (~chatzilla@uawifi-nat-210-16.arizona.edu) |
| 19:29.42 | WhIteSidE | Cepstral? Anyone? |
| 19:30.02 | mmj_nix | <PROTECTED> |
| 19:31.00 | jaskew | WhiteSide: I dunno Cepstral - maybe someone else does. |
| 19:31.52 | WhIteSidE | It seems like with * 1.6 all of the text to speech engines no longer have support from the dialplan |
| 19:32.01 | jaskew | mmj_nix: Inthe case of Call Waiting, I would expect the ATA to send SIP messages to put the existing call on hold (can't remember what that is) and then send an "OK" to the incoming call to connect it. |
| 19:32.07 | WhIteSidE | Are there any TTS solutions that are "workable" with 1.6? |
| 19:33.50 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 19:34.38 | jaskew | Whiteside: I don;t know that either. Hang around, hopefully someone will pick up on the question |
| 19:34.47 | WhIteSidE | :) |
| 19:35.08 | WhIteSidE | That's why I came here, the * group has always provided excellent help |
| 19:35.15 | mmj_nix | <PROTECTED> |
| 19:36.06 | jaskew | mmj_nix: Are those drop-downs or are they expecting a number string? |
| 19:38.32 | mmj_nix | jaskew: they are just empty text fields right now. I have tried things like *0 #0 #72 and the like with out any luck. |
| 19:42.14 | jaskew | mmj_nix: Hmm. So my guess would be if you put *0 in the "hold" box, then you could flash and hit *0 to put the call on hold. Not sure how you retrieve (maybe *0 again)? I'm confused by the "flash" item though. |
| 19:42.30 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 19:44.05 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 19:44.08 | jaskew | See http://www.voip-info.org/wiki/view/SIP for info on SIP protocol. |
| 19:44.59 | mmj_nix | jaskew: I was thinking something similar, but until I can get hook-flash to actually accepted by *, in call features are not possible :( |
| 19:45.36 | Madoc | Is there documentation on developing custom Asterisk Applications, like a function reference at least? |
| 19:45.58 | jaskew | I think the only time * accepts "hook flash" is if you have a phone connected to an FXS port 9or is it FXO - I get them mixed up) |
| 19:47.01 | jaskew | Other thatn that, it's the ATA's job to handle a hook flash and decide what it means. |
| 19:50.15 | ruben23 | hi |
| 19:50.45 | mmj_nix | <PROTECTED> |
| 19:51.04 | ruben23 | any suggestion on this error log--> http://pastebin.com/KeVE9pac |
| 19:52.12 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 19:52.34 | *** part/#asterisk SAngeli (~SAngeli@host242-77-static.82-94-b.business.telecomitalia.it) |
| 19:53.49 | ruben23 | i got more of this on my logs-----> http://pastebin.com/r3FJK0CF |
| 19:55.00 | LemensTS | anyone offer a sip trunk for faxing i can put on an ata? |
| 19:55.14 | LemensTS | just for faxing no calls |
| 19:56.48 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 19:57.08 | *** join/#asterisk dzup (~alex@201-130-207-130-cable.cybercable.net.mx) |
| 20:00.32 | *** join/#asterisk IBC_JKENNEY (~jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 20:00.54 | IBC_JKENNEY | I know this is not the topic for this room however does anyone have any experience with hylafax and faxmerge |
| 20:00.58 | IBC_JKENNEY | hylafax room is dead |
| 20:03.35 | *** part/#asterisk jconnolly (~jconnolly@firebug.buglabs.net) |
| 20:10.52 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
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| 20:25.35 | miamiseb | I have returned |
| 20:27.53 | chuckf | sorry to hear that miamiseb |
| 20:30.10 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 20:30.29 | Katty | drags in |
| 20:30.52 | spenguin[work] | hey Katty |
| 20:31.12 | Katty | hello |
| 20:31.14 | spenguin[work] | anyone here use the skype module with asterisk? |
| 20:31.18 | spenguin[work] | sup Katty :> |
| 20:31.22 | Katty | ill :< |
| 20:31.35 | spenguin[work] | :< |
| 20:31.41 | *** join/#asterisk dzup (~alex@unaffiliated/dzup) |
| 20:31.44 | spenguin[work] | take a holiday |
| 20:31.48 | Katty | how're you mister penguin |
| 20:32.11 | spenguin[work] | sleepy - still alive on coffee |
| 20:32.39 | IBC_JKENNEY | I know this is not the hylafax room however nobody is ever there |
| 20:32.47 | IBC_JKENNEY | does anyone have any experience with it |
| 20:33.01 | Katty | nope |
| 20:33.05 | IBC_JKENNEY | :( |
| 20:33.15 | Katty | maybe that should tell you something |
| 20:33.44 | IBC_JKENNEY | hey now. It works really well i'm trying to get it to do faxblasting |
| 20:33.53 | IBC_JKENNEY | thru open office |
| 20:34.03 | IBC_JKENNEY | so i expect some crickets |
| 20:34.16 | IBC_JKENNEY | but i am sure i'm not the only one that has attempted it |
| 20:34.42 | Katty | good luck. |
| 20:35.55 | carrar | Write a doc when you have it working |
| 20:36.05 | spenguin[work] | lol |
| 20:36.22 | Katty | good suggestion |
| 20:36.32 | spenguin[work] | meanies |
| 20:36.35 | carrar | Hows the zoo Katty? |
| 20:37.18 | Katty | fine fine, i guess |
| 20:37.31 | Katty | maybe i should turn ferretcam on |
| 20:37.38 | spenguin[work] | yay |
| 20:37.45 | carrar | Guess!?!? You need tracks on the ceiling at home so you have remote movable cameras |
| 20:38.18 | Katty | ferret cam or squirrel cam? |
| 20:38.19 | carrar | and remote lighting and door control |
| 20:38.29 | Katty | you guys pick |
| 20:38.29 | spenguin[work] | the poor critters think they are always on TV, but katty turns them off randomly |
| 20:38.30 | carrar | Accident Cam |
| 20:38.39 | spenguin[work] | squirrel |
| 20:38.48 | Katty | k, squirrel it is |
| 20:38.53 | spenguin[work] | \0/ |
| 20:39.53 | Katty | it's on |
| 20:40.01 | spenguin[work] | Katty: I was thinking - you should tie a rope from the tray to the tree, get the squirrels to traverse the rope :p |
| 20:40.12 | spenguin[work] | ~critters |
| 20:40.14 | spenguin[work] | ~critter |
| 20:40.15 | infobot | Perl/Tk flat text search script. URL: http://www.mainmatter.com/critter/ |
| 20:40.18 | spenguin[work] | hrm |
| 20:40.23 | spenguin[work] | whats the url |
| 20:40.25 | Katty | infcrittercam |
| 20:40.31 | Katty | infobot: crittercam |
| 20:40.32 | infobot | [crittercam] Katty's live broadcast of The Nut House @ http://ustre.am/8H5d and The Fuzzy Ferret Flat @ http://ustre.am/bEBU |
| 20:40.50 | carrar | and there is a squrrel |
| 20:40.58 | Katty | one on the left tree too |
| 20:41.04 | *** join/#asterisk titter` (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 20:41.18 | Katty | one on the ground |
| 20:41.30 | Katty | 2 on the ground |
| 20:41.31 | carrar | plethora of squirrels |
| 20:41.48 | Katty | omnomnomnomnom |
| 20:42.02 | *** join/#asterisk BCS-Satori (~BCS-Sator@75-148-21-113-WashingtonDC.hfc.comcastbusiness.net) |
| 20:42.33 | Katty | bunny? |
| 20:42.45 | carrar | bottem laft? |
| 20:42.46 | carrar | left |
| 20:42.48 | Katty | yeah |
| 20:42.54 | Katty | i think it was a bunny |
| 20:43.13 | carrar | or a chipmonk |
| 20:43.52 | Katty | too big to be a chipmonk |
| 20:44.17 | spenguin[work] | will pay to watch a live penguin |
| 20:44.33 | spenguin[work] | s/will/would |
| 20:45.38 | Katty | isn't there a live penguin webcam at monteray bay? |
| 20:45.47 | Katty | http://www.montereybayaquarium.org/efc/efc_splash/splash_cam.aspx |
| 20:45.56 | hardwire | spenguin[work]: how much? |
| 20:46.02 | spenguin[work] | yeah? |
| 20:46.04 | hardwire | do you just wanna watch? |
| 20:46.14 | carrar | Katty you need like a Canon VB-C50 |
| 20:46.24 | hardwire | for no touching.. just watching.. that'll be pretty cheap.. I just gotta find a few penguins that won't mind. |
| 20:46.25 | spenguin[work] | hardwire: no, watch plus remote control :p |
| 20:46.39 | Katty | carrar: no i don't |
| 20:46.51 | carrar | we can move it around and see what that was |
| 20:46.51 | Katty | carrar: that webcam does full HD recording, but i don't have the bandwidth for it |
| 20:47.07 | Katty | yeah so you can turn the camera around and watch inside the window? |
| 20:47.09 | Katty | i don't think so |
| 20:47.19 | carrar | put paper behind it |
| 20:47.28 | BCS-Satori | I am having an issue with presence in asterisk not indicating a state change. When I make a phone call from extension 6101 and do a "core show hints" i show the phones state in idle; to which the watching phone does not see them on the call. I have to be missing something to change the state? |
| 20:47.28 | Katty | spenguin[work]: how do you like the penguin camera? |
| 20:47.31 | carrar | or a Digim white paper |
| 20:47.54 | Katty | or not |
| 20:47.54 | spenguin[work] | Katty: hold on, looking at it now |
| 20:48.09 | hardwire | splash_cam sounds fun |
| 20:48.39 | Katty | they look bored |
| 20:49.11 | spenguin[work] | heh throw in some fish :P |
| 20:49.20 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 20:49.35 | Katty | feeding time is 3PM PST |
| 20:49.57 | Katty | so in about 2 more hours it'll be fish time |
| 20:50.00 | miamiseb | got fop2 popups working |
| 20:50.04 | LemensTS | . |
| 20:50.04 | miamiseb | had to unpack and fix the JS though |
| 20:50.06 | miamiseb | blah |
| 20:50.17 | Katty | fop...2? |
| 20:50.18 | hardwire | MOVE YOU STUPID PENGIES! |
| 20:50.25 | Katty | flash operator panel? |
| 20:50.26 | Katty | 2? |
| 20:50.27 | miamiseb | Katty: yeah, his next version |
| 20:50.30 | miamiseb | shrugs |
| 20:50.31 | hardwire | fopfopfop |
| 20:50.34 | Katty | interesting |
| 20:50.37 | Katty | i will have a look at it |
| 20:50.38 | miamiseb | it works for my screen pops, thats all I care about |
| 20:50.38 | spenguin[work] | heh thanks Katty |
| 20:50.43 | miamiseb | but as I said |
| 20:50.47 | miamiseb | it was broken in it's default state |
| 20:50.48 | Katty | what is a 'screen pop' |
| 20:50.48 | spenguin[work] | ill be watching this cam too :> |
| 20:51.17 | miamiseb | screen pop is something that will popup a url based on an incoming call with a custom url that includes * parameters |
| 20:51.26 | miamiseb | now if theres a bot watching, he can use that too |
| 20:51.37 | carrar | BCS-Satori, did you search for setting up BLF on asterisk? |
| 20:51.54 | Katty | http://www.penguin.net.nz/viewing/webcams.html <- more penguin webcams |
| 20:52.18 | Katty | miamiseb: does it just pop up a url? |
| 20:52.23 | miamiseb | saw tons of penguins in the aviary in pitts, very nice. |
| 20:52.31 | Katty | miamiseb: or can it be "this string of text" |
| 20:52.38 | miamiseb | thats all a screen pop is, fop does more stuff |
| 20:52.44 | miamiseb | it'll do whatever you want via ajax |
| 20:52.56 | Katty | mmmyeah i don't work with ajax. |
| 20:52.59 | miamiseb | in my case, it doing a window.open |
| 20:53.03 | Katty | but that's still interesting to hear. |
| 20:53.15 | miamiseb | its pretty prebuilt, but I can work with JS just fine |
| 20:53.35 | miamiseb | I dunno how routine unpacking and fixing a JS var through debugging is though |
| 20:53.50 | Katty | is going to go lay down for awhile. |
| 20:54.02 | Katty | must get well :< laters. |
| 20:54.16 | miamiseb | feel better |
| 20:54.25 | carrar | drink lots of carrot juice |
| 20:58.04 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:00.02 | *** join/#asterisk WhIteSidE (~chatzilla@uawifi-nat-210-24.arizona.edu) |
| 21:00.27 | WhIteSidE | Cepstral or other TTS on 1.6? Anyone? |
| 21:01.12 | russellb | res_cepstral is supported in asterisk 1.6, if that was the question ... |
| 21:02.39 | spenguin[work] | poor Katty |
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| 21:40.11 | kfife | Sanity check: appears I can't use a DNS name as in 'permit=sip.myserver.net', is that right ? |
| 21:41.17 | kfife | or is there there a dns-friendly equivalent like exterip and externhost? I'm not seeing it if there is. |
| 21:41.47 | kfife | ... and Tivo1 needs to get a Tivo series 4 |
| 21:46.07 | *** join/#asterisk asteriskmonkey (~philip@69.77.169.14) |
| 21:46.33 | asteriskmonkey | I see dahdi_dummy loaded but im getting : app_meetme.c:803 build_conf: Unable to open pseudo device |
| 21:46.39 | asteriskmonkey | any pointers for fix? |
| 21:48.31 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 21:49.05 | elliot98 | slowly sticks his head into the channel |
| 21:49.13 | elliot98 | anybody home? |
| 21:49.45 | elliot98 | I am trying to upgrade from an older version of asterisk that uses ZAPTEL to the newer ones that use DAHDI |
| 21:50.30 | Kobaz | sounds fun |
| 21:50.39 | elliot98 | should I first remove ZAPTEL or does that not make a difference? |
| 21:50.40 | Kobaz | gets some popcorn |
| 21:50.51 | Kobaz | doesn't matter... but at the minimum, unload the zaptel modules |
| 21:53.24 | *** join/#asterisk titter` (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 21:53.57 | elliot98 | I see...and when I update asterisk, all I do is install it over the existing version? |
| 21:54.20 | Kobaz | remove the old asterisk modules from /usr/lib/asterisk/modules |
| 21:54.27 | Kobaz | and then yeah, you can make install over the old one |
| 21:57.11 | elliot98 | thanks! |
| 21:57.26 | asteriskmonkey | anyone got this before? app_meetme.c:803 build_conf: Unable to open pseudo device |
| 21:57.56 | elliot98 | on other things, how's the asterisk crew coming along lately? |
| 21:58.48 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:58.54 | Kobaz | they are coming... |
| 21:59.29 | *** join/#asterisk fish-bulb (~cstewart@nat/digium/x-rubuecddmacionup) |
| 22:00.21 | elliot98 | do I need to upgrade my Libpri too? |
| 22:00.26 | Kobaz | you should |
| 22:00.31 | miamiseb | and now I've got my destination number added to the URL, yay! pretty much done now |
| 22:00.33 | Kobaz | it's probably pretty old if you're using zaptel |
| 22:00.40 | Kobaz | lots of bug fixes |
| 22:01.25 | Kobaz | and some great new bugs as well |
| 22:01.27 | Kobaz | good stuff |
| 22:02.38 | elliot98 | oh, yummy! |
| 22:03.10 | Kobaz | speaking of t1 |
| 22:03.11 | Kobaz | er |
| 22:03.13 | Kobaz | wrong window |
| 22:03.13 | elliot98 | talking about bugs...does 1.4.29 (latest, I think) have any regression bugs? |
| 22:03.20 | Kobaz | probably |
| 22:03.42 | elliot98 | I don't want to upgrade just to find out it's more buggy than what I have |
| 22:03.58 | Kobaz | that's kinda the only way |
| 22:04.02 | Kobaz | hopefully you have unit tests |
| 22:04.49 | Kobaz | there it is |
| 22:04.52 | Kobaz | er |
| 22:05.10 | carrar | install it in your test lab |
| 22:05.22 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 22:06.02 | elliot98 | I wish I have a test lab for this server |
| 22:06.07 | elliot98 | *had |
| 22:06.56 | elliot98 | what is spandsp used for? |
| 22:07.44 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 22:08.07 | elliot98 | do I need it? |
| 22:09.17 | asteriskmonkey | Anyone know why id get this ? Error inserting dahdi_dummy (/lib/modules/2.6.18-92.1.22.el5PAE/dahdi/dahdi_dummy.ko): Unknown symbol in module |
| 22:10.32 | miamiseb | whats dmesg show? |
| 22:11.18 | nightrid3r | whats uname -a show |
| 22:12.29 | asteriskmonkey | miamiseb: its barfing lots of lines like dahdi_dummy: Unknown symbol dahdi_receive |
| 22:13.30 | *** part/#asterisk rttrey (~trey@209.208.18.121) |
| 22:13.47 | asteriskmonkey | nightrid3r: 2.6.18-92.1.22.el5PAE |
| 22:13.58 | tzafrir | asteriskmonkey, how do you load it? |
| 22:14.26 | miamiseb | asteriskmonkey: do you have zaptel loaded? pastebin an lsmod |
| 22:14.27 | asteriskmonkey | tzafrir tried the service dahdi start and then insmod dahdi_dummy |
| 22:14.48 | asteriskmonkey | miamiseb: no zaptel on the box |
| 22:15.02 | miamiseb | can I see an lsmod anyway? |
| 22:15.14 | tzafrir | asteriskmonkey, a different version of dahdi.ko is (still?) loaded |
| 22:15.25 | Zhad | Are you sure you haven't made any changes to your kernel since compiling it? |
| 22:15.41 | asteriskmonkey | here is the pastebin http://pastebin.ca/1830765 |
| 22:15.48 | asteriskmonkey | Zhad: no not touched kernel |
| 22:15.51 | Zhad | ie, the sources are current wrt the version running. |
| 22:16.04 | miamiseb | you've got dahdi loaded |
| 22:16.14 | asteriskmonkey | dummy is not loading though |
| 22:16.41 | asteriskmonkey | do i need to peel out dahdi from he kernel and try again? |
| 22:17.06 | miamiseb | pastebin cat /proc/kallsyms |
| 22:17.11 | miamiseb | or better yet |
| 22:17.14 | miamiseb | grep that for dahdi |
| 22:17.14 | tzafrir | asteriskmonkey, what version of dahdi is it? |
| 22:17.23 | asteriskmonkey | 2.2.1 |
| 22:17.25 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 22:18.02 | Qwell | asteriskmonkey: How did you install DAHDI? |
| 22:18.26 | asteriskmonkey | make, make install |
| 22:18.37 | Qwell | What distribution? |
| 22:18.48 | asteriskmonkey | centos 5.4 |
| 22:18.57 | Qwell | That isn't a 5.4 kernel |
| 22:19.36 | asteriskmonkey | hang on wrong box centos 5, kernel 2.6.18-92.1.22.el5PAE |
| 22:20.21 | asteriskmonkey | mmm wait... if i upgraded from a prior version would i have to completley unload it from memory, the service dahdi restart would do that would it? |
| 22:22.40 | asteriskmonkey | Qwell: if i upgraded and left a running copy of an older dahdi in memory, the service restart on dahdi dosnt free the old kernel modules does it? |
| 22:23.52 | Kobaz | lsmod |
| 22:23.55 | Kobaz | unload as needed |
| 22:25.34 | asteriskmonkey | Kobaz: thanks, tried there inuse :) guess i have to wiat until people go home |
| 22:26.40 | Kobaz | /etc/init.d/asterisk stop |
| 22:26.45 | Kobaz | that'll make them not in use |
| 22:27.08 | Kobaz | asteriskmonkey: you're trying to upgrade dahdi during hours? |
| 22:28.31 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 22:29.14 | asteriskmonkey | no: did it after hrs a couple days back however i noticed conferencing was busted and paging so i tracked it back to the timer :/ |
| 22:30.38 | asteriskmonkey | system was stopped at that time, so i guess for one reason or another it might not have kicked out the old dahdi from the kernel :/ will try manually, thanks guys.. usually the stupid mistakes come to light after asking others :") |
| 22:31.55 | *** join/#asterisk b14ck (~comradeb1@cpe-24-24-136-239.socal.res.rr.com) |
| 22:38.24 | elliot98 | I am getting this error: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref |
| 22:38.52 | elliot98 | how do I resolve this error? |
| 22:40.05 | *** join/#asterisk brezular (~brezular@adsl-dyn-116.95-102-190.t-com.sk) |
| 22:40.09 | asteriskmonkey | elliot: you just upgrade your dahdi? |
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| 23:21.53 | timeshell | Which current version of asterisk is most stable in the 1.6.0 series? |
| 23:23.41 | *** join/#asterisk Cobraz (~cobraz@ti0011a380-1449.bb.online.no) |
| 23:23.44 | Cobraz | Hello :) |
| 23:23.48 | *** part/#asterisk asteriskmonkey (~philip@69.77.169.14) |
| 23:24.52 | Cobraz | I've just installed my AsteriskNow box, its running smoothly. However, i was interested in the function "Skype Gateway" that was mentioned in the description at AsteriskNow.com - how do I get this to work? :) |
| 23:26.17 | timeshell | You have to buy the chan_skype channel from digium |
| 23:26.52 | Cobraz | Okey, why does this cost so much? |
| 23:27.23 | Chainsaw | Cobraz: Because they're in it for the money. |
| 23:27.47 | timeshell | lol |
| 23:28.28 | Cobraz | Well, if they are - its allways some guys hooking up their computers in the garage for a weekend to make an opensource free version of it - does it exist? |
| 23:29.27 | timeshell | No |
| 23:29.35 | Cobraz | Sure? |
| 23:29.38 | Chainsaw | Cobraz: Skype isn't open-source, remember? |
| 23:29.42 | timeshell | Skype and Digium have a working agreement. Skype assisted Digium to make the skype channel |
| 23:30.15 | Chainsaw | Cobraz: I can tell you that it will never be. A lot of people use it "because it's free!". |
| 23:30.34 | Chainsaw | Cobraz: And they will capitalise on that with paid specialty products like this. Skype is not your friend. |
| 23:30.50 | Cobraz | Well, Skype is awesome for work. |
| 23:30.54 | timeshell | repeats "Skype is not your friend" |
| 23:31.00 | timeshell | Skype stinks. |
| 23:31.03 | Cobraz | We use it at our business for collaberation. |
| 23:31.12 | timeshell | Go use googletalk or callcentric |
| 23:31.26 | Cobraz | And they are supported in Asterisk? |
| 23:31.30 | timeshell | Yes |
| 23:31.35 | timeshell | Because they are SIP based. |
| 23:32.09 | sneaker | Hello folks. |
| 23:37.59 | *** join/#asterisk titter` (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 23:38.39 | miamiseb | Night everyone, have a great one |
| 23:42.32 | sneaker | So, had some toll fraud through my Asterisk 1.4 box the other day. I have 3 extensions configured on it and have verified that they were not compromised. |
| 23:43.16 | sneaker | So, I am left thinking that it was either a bug in asterisk or a problem with my dialplan that was exploited. |
| 23:48.06 | sneaker | My question for the fine folks in this channel is where do you look for info on these kinds of vulnerabilities. I'm not gonna ask you to fix my problem for me. Just a little nudge in the right direction. |
| 23:51.16 | timeshell | Look at the CDR log first. |
| 23:51.24 | timeshell | See where the call originated from. |
| 23:51.45 | timeshell | If there's nothing in your log, perhaps they have the login/password for your SIP provider. |
| 23:51.50 | Chainsaw | sneaker: Usually it means you haven't used contexts to properly separate out "the inside" from "the outside". |
| 23:51.56 | timeshell | Check the call logs with your SIP provider also. |
| 23:52.15 | timeshell | Oh.... lol |
| 23:52.19 | timeshell | misunderstood the question |
| 23:52.43 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 23:52.43 | timeshell | http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt |
| 23:55.05 | spenguin[work] | does anyone knwo anything about the skype module for asterisk? |
| 23:55.40 | sneaker | One other thing, in the asterisk cdr rather than one of my authenticated extensions it shows 'SIP/<fraudsters ip addr>' |
| 23:56.27 | sneaker | Thanks for the info/tips. Mucho appreciado.. |
| 23:56.42 | spenguin[work] | can 1 channel carry only 1 call? |
| 23:56.55 | spenguin[work] | or does it depend on what type of channel it is? |