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| 01:17.50 | wasanzy | X-Rob: hi |
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| 01:56.11 | ChannelZ | gruvfunk: you plug your hole? |
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| 02:21.05 | drynish | Hi! :) |
| 02:21.08 | drynish | I was in the wrong channel |
| 02:22.08 | gruvfunk | ChannelZ: yessir, thank you for your assistance, and p3nguin too |
| 02:23.40 | gruvfunk | though my brain is now mapping all sorts of stuff... for example on the whole default context thing... |
| 02:25.01 | gruvfunk | in a multiple trunk system (multi-tenant really, contextualized), how should the [general] context= flow, if not to default? |
| 02:25.07 | gruvfunk | create a new context ? (duh) |
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| 02:25.58 | p3nguin | I use a context called misc_calls for the general context. The default context in extensions.conf is present and empty. |
| 02:27.03 | gruvfunk | righton p3nguin |
| 02:35.11 | gruvfunk | question: this same customer is not on the latest 1.8.5, and looks to be compiled from source on a CentOS |
| 02:35.26 | gruvfunk | I'm thinking of reinstalling with yum repos to maintain the package updated |
| 02:35.49 | gruvfunk | I've backed everything up, but not sure how to uninstall the old version ? |
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| 02:50.27 | gruvfunk | ah, simple rm -rf |
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| 03:24.06 | ChannelZ | ah yes, is "repair module really fast" command |
| 03:39.12 | WIMPy | No. It's a root kit. "root me" |
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| 04:07.32 | gruvfunk | hey p3nguin can you give me a hand re: misc_calls |
| 04:08.12 | p3nguin | Yes. |
| 04:09.01 | gruvfunk | so, if I implement same as you described here, well then calls coming in via SIP provider don't flow through to the appropriate context |
| 04:09.20 | gruvfunk | i blanked default context, created a misc_calls, and also have an internal context |
| 04:09.24 | gruvfunk | (for extensions) |
| 04:09.40 | p3nguin | Then you have screwed up sip definitions. |
| 04:09.53 | gruvfunk | in sip.conf, general set to misc_calls, have 2 SIP trunks, each has their own context defined |
| 04:10.12 | gruvfunk | oh, i have misc_calls context blank.... |
| 04:10.16 | gruvfunk | i suppose it needs to not be |
| 04:10.32 | gruvfunk | but i guess i'm not following what I need to do |
| 04:10.40 | p3nguin | Anything explicitly defined as a peer won't be using the misc_calls context. |
| 04:10.57 | p3nguin | Let's start with sip.conf. |
| 04:11.00 | gruvfunk | what if it's defined as "friend" |
| 04:11.25 | p3nguin | The type isn't important in this situation. It's still a peer. |
| 04:11.33 | gruvfunk | right o |
| 04:12.07 | p3nguin | The general section of sip.conf... |
| 04:12.18 | p3nguin | Define the context: context=misc_calls |
| 04:12.36 | gruvfunk | check |
| 04:12.42 | p3nguin | Now any calls that come into the system which do not match a peer entry will go to misc_calls in extensions.conf. |
| 04:12.58 | gruvfunk | yeah problem is all calls seem to fall into that context now |
| 04:13.17 | p3nguin | Your peer definitions are apparently fucked up. |
| 04:13.17 | gruvfunk | even though my SIP peer is set to that peer's specific context |
| 04:13.58 | p3nguin | Each definition needs to have something to make it match. Usually host, port, and/or username. |
| 04:15.23 | gruvfunk | I have a register string, and then I have a context for each of the SIP trunk providers |
| 04:15.38 | gruvfunk | OR ... peer definitions rather |
| 04:15.51 | gruvfunk | within each, I have context=from-providerX |
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| 04:16.09 | p3nguin | You've ensured that the register string is in the general section, right? |
| 04:16.16 | gruvfunk | they are both set to type=friend |
| 04:16.39 | gruvfunk | right |
| 04:16.40 | p3nguin | With type=friend, make sure you define the username/defaultuser in the definition. |
| 04:17.38 | gruvfunk | i have fromuser and defaultuser |
| 04:18.22 | p3nguin | Unless you know that you need fromuser, get rid of it. |
| 04:18.37 | p3nguin | Depending on what you're doing, you may need it. |
| 04:19.57 | p3nguin | You also have host set to the IP address of the provider? |
| 04:20.13 | p3nguin | or at least their hostname. |
| 04:20.16 | Maliuta | Anyone know if you can get dahdi channels to do rfc2833 dtmf rather than inband (which isn't supported by certain codecs/channel types)? |
| 04:20.23 | gruvfunk | replaced with username, get same result: "handle_request_invite: call from (IP : 5060) to extension 's' rejected because extension not found in context 'misc_calls' |
| 04:20.35 | gruvfunk | yes, using host |
| 04:21.07 | p3nguin | Can you pastebin your entire entry which should be matching calls from that peer? |
| 04:22.20 | Maliuta | gruvfunk: that's a simple fix ... put an 's' extension in that context |
| 04:22.24 | p3nguin | No. |
| 04:22.27 | gruvfunk | no |
| 04:22.29 | gruvfunk | :) |
| 04:22.42 | p3nguin | That's not a fix. That's a fux. |
| 04:22.48 | gruvfunk | the trunk works fine using 'default' context in general |
| 04:22.57 | gruvfunk | so we're trying to install some best practices here |
| 04:22.58 | p3nguin | I'm sure it does. |
| 04:22.59 | Maliuta | gruvfunk: it's not unusual for people to send sip calls to s@your.host ... even some providers |
| 04:23.07 | p3nguin | That';s not the problem. |
| 04:23.11 | gruvfunk | agree |
| 04:23.23 | p3nguin | The problem is that calls from the peer are not matching the defined SIP account. |
| 04:24.06 | p3nguin | As soon as I see it, unaltered (except for hiding the password if there is one), I'll try to see why it doesn't match. |
| 04:26.06 | p3nguin | Or, on the other hand, if I never see it, I'll eat my soup and watch TV. |
| 04:26.50 | gruvfunk | http://pastebin.com/3HGnUziG |
| 04:27.05 | gruvfunk | altered |
| 04:27.18 | gruvfunk | carefully, of course |
| 04:27.55 | p3nguin | You can't have defaultuser and username. Choose accordingly for your version of Asterisk. |
| 04:28.19 | p3nguin | Get rid of fromuser, fromdomain, and insecure... unless you KNOW that you need them. |
| 04:28.37 | gruvfunk | so, which is it for 1.8.5? |
| 04:28.52 | p3nguin | defaultuser |
| 04:28.53 | gruvfunk | i added "username" on your suggestion |
| 04:30.16 | gruvfunk | are you suggesting removing those 3 items would make it "match" ? |
| 04:30.18 | gruvfunk | hmm |
| 04:30.24 | gruvfunk | trying |
| 04:31.14 | gruvfunk | nah |
| 04:32.04 | gruvfunk | that ain't it |
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| 04:35.22 | gruvfunk | got it |
| 04:35.34 | gruvfunk | back to extensions.conf, my misc_calls was still empty... :) |
| 04:35.50 | gruvfunk | added : include => from-provider1 |
| 04:37.51 | gruvfunk | I think this is the right approach, as you indicated earlier today: not allowing access to the outbound contexts from any inbound contexts |
| 04:39.49 | ChannelZ | do you need anonymous SIP? |
| 04:40.01 | gruvfunk | i do not |
| 04:40.43 | ChannelZ | then allowguests=no and then it doesn't matter |
| 04:41.12 | ChannelZ | allowguest singular sorry |
| 04:42.25 | gruvfunk | true true |
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| 04:45.53 | gruvfunk | does make for a decent lesson though |
| 04:46.19 | gruvfunk | and so does not keeping a full glass of water so close to your shit on your desk.. F |
| 04:46.33 | gruvfunk | i'm sure you'll all get a good laugh.. just spilled 6+ oz of water all over my desk |
| 04:48.01 | p3nguin | It's not the right approach to work AROUND the actual problem of not getting the peer to match. |
| 04:48.37 | p3nguin | Working around it by accepting the calls to the general context and then including the real context where the call should have gone... that's just, well, stupid. |
| 04:50.27 | gruvfunk | hmm |
| 04:50.29 | gruvfunk | i see your point |
| 04:50.42 | gruvfunk | here I thought I was on a fix since you didnt reply |
| 04:51.30 | p3nguin | I lose interest quickly when I ask for an unaltered pastebin and I get an altered paste instead. |
| 04:51.58 | p3nguin | In this case, I'm talking about the false hostname of "provider1.com" |
| 04:53.06 | gruvfunk | i carefully edited |
| 04:53.16 | gruvfunk | replace provider1 with the actual domain |
| 04:53.32 | p3nguin | I have no clue what the actual hostname is. Now I don't even care. |
| 04:53.40 | gruvfunk | it's callcentric.com |
| 04:53.47 | gruvfunk | i don't see how that's relevant |
| 04:54.01 | ChannelZ | it's relevant if your calls come from some other IP |
| 04:54.15 | gruvfunk | and not "callcentric.com" ? |
| 04:54.18 | gruvfunk | a ha |
| 04:54.36 | ChannelZ | or whatever that resolves to |
| 04:54.37 | gruvfunk | yeah, I see calls coming in from IP's not hostnames |
| 04:54.51 | ChannelZ | Asterisk matches peers by a couple of different criteria depending on its type |
| 04:55.01 | gruvfunk | the IP's seem to change from time to time, there are also different DID's on that same trunk |
| 04:55.11 | p3nguin | I use host=callcentric.com |
| 04:55.13 | ChannelZ | See 'Naming devices' in the sample sip.conf |
| 04:56.01 | gruvfunk | p3nguin: same here |
| 04:56.23 | p3nguin | type=peer, fromuser=myCCphonenumber, username=myCCphonenumber |
| 04:56.32 | p3nguin | My callcentric peer entry is VERY minimal. |
| 04:58.26 | p3nguin | I still can't understand why everyone thinks their ITSPs' IP addresses and hostnames are super duper top secret. |
| 04:58.57 | gruvfunk | sorry p |
| 04:59.22 | p3nguin | You're not the first person to have that idea. |
| 04:59.31 | p3nguin | And you won't be the last. |
| 04:59.39 | gruvfunk | when ignorant, best be safe |
| 05:00.36 | coppice | when ignorant, best enter politics |
| 05:00.48 | gruvfunk | lol |
| 05:00.48 | ChannelZ | I hear Google's IPs are top secret |
| 05:00.52 | p3nguin | But now that you've divulged the national secret, I can say it's probably not the hostname that's causing the problem directly. Are you using SRV lookups? CallCentric actually uses SRV correctly. |
| 05:01.39 | gruvfunk | no |
| 05:03.13 | gruvfunk | wait yes |
| 05:03.16 | p3nguin | They have nine host addresses, all on port 5080. |
| 05:03.43 | gruvfunk | srvlookup=yes |
| 05:03.48 | gruvfunk | so is it a port thing? |
| 05:04.16 | gruvfunk | maybe share your CC context? |
| 05:04.26 | gruvfunk | i mean peer def. |
| 05:05.34 | p3nguin | http://pastebin.com/hLEJgyny |
| 05:14.22 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 05:17.02 | gruvfunk | stumped, tried trimming down to match yours, and even switched to peer to make sure |
| 05:18.10 | gruvfunk | this is 1.8.5, not sure if different |
| 05:19.40 | ChannelZ | well username is now defaultuser |
| 05:20.07 | gruvfunk | yep, that's what was set to begin with |
| 05:20.29 | ChannelZ | pastebin some verbose console output when a call comes in |
| 05:22.36 | gruvfunk | under the current setup (trying to allow guest calls and still separate things out contextually), i'm just getting one line, can I paste here? |
| 05:23.00 | gruvfunk | NOTICE[24294]: chan_sip.c:21870 handle_request_invite: Sending fake auth rejection for device <sip:MYPHONENUMBER@66.193.176.35>;tag=3521683299-755097 |
| 05:24.01 | gruvfunk | so it's not even matching it seems |
| 05:24.41 | ChannelZ | is 66.193.176.35 you? |
| 05:24.49 | gruvfunk | callcentric servers |
| 05:24.53 | gruvfunk | i'm calling form a mobile phone |
| 05:25.05 | ChannelZ | I don't see that in the list of callcentric.com IPs |
| 05:25.06 | gruvfunk | into one of the CC DID's that ring into this PBX |
| 05:25.16 | ChannelZ | theirs are all 204.* |
| 05:25.36 | gruvfunk | a whois shows it belongs to them |
| 05:26.09 | gruvfunk | or at least references their name |
| 05:26.11 | gruvfunk | Callcentric Wholesale, Inc. TWTC-NYCL-C-ACCAT-00 (NET-66-193-176-0-1) 66.193.176.0 - 66.193.177.255 |
| 05:26.46 | ChannelZ | doing a nslookup here I don't see it |
| 05:27.29 | gruvfunk | hmm |
| 05:28.00 | gruvfunk | you're right, whois gives reference |
| 05:28.03 | gruvfunk | nothing else works |
| 05:29.53 | ChannelZ | I see a nameserver entry for ns1.telengy.net that is in the same netblock but that's about it. None of their SRV records point back to that IP either |
| 05:30.43 | gruvfunk | what are we saying? |
| 05:30.59 | ChannelZ | in any case if your type=peer it should be able to pick it up by the fromuser though |
| 05:31.10 | gruvfunk | tried that |
| 05:31.18 | ChannelZ | assuming it's right. Post a sip debug of a call attempt |
| 05:33.02 | ChannelZ | brb need water |
| 05:43.30 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:47.22 | ChannelZ | Bollocks |
| 05:47.39 | ChannelZ | I just accidentally firewalled my asterisk box off... *completely* |
| 05:50.13 | ChannelZ | guess it's road-trip time |
| 05:53.50 | gruvfunk | rough |
| 05:54.10 | gruvfunk | good luck ChannelZ |
| 05:54.19 | gruvfunk | i'm just going to not allow guest calls, since don't need to anyway |
| 05:54.38 | gruvfunk | but will be back at it tomorrow on a different box trying to get a good thing going |
| 05:54.44 | ChannelZ | right - although that doesn't seem to be your primary problem |
| 05:54.52 | gruvfunk | exactly |
| 05:54.59 | ChannelZ | without guest you can't receive anything at the moment right? |
| 05:55.18 | gruvfunk | I get |
| 05:55.27 | gruvfunk | but |
| 05:55.55 | gruvfunk | my general context=misc_calls, in extesions.conf has include => from-callcentric :) |
| 05:56.10 | ChannelZ | yeah but that doesn't do shit unless guest calls are on |
| 05:57.06 | gruvfunk | oh right, sorry.. yeah getting tired |
| 05:57.09 | ChannelZ | Your original problem, people exploting your system, was a dialplan problem. If that's fixed, the whole guest calls thing doesn't really make any difference |
| 05:57.29 | ChannelZ | But your second problem is that ALL your calls are guest calls because they aren't matching your desired peer |
| 05:57.29 | gruvfunk | so I am allowing guest calls |
| 05:57.35 | gruvfunk | right |
| 05:57.49 | gruvfunk | and I can't figure out why callcentric behaves that way when voip.ms does not |
| 05:58.06 | gruvfunk | also, interestingly, this is what callcentric posts as a config |
| 05:58.07 | gruvfunk | http://www.callcentric.com/support/device/asterisk/ |
| 05:58.19 | gruvfunk | they actually expect you to use context-from-callcentric in general |
| 05:58.40 | gruvfunk | but we know p3nguin has it setup and working differently |
| 05:59.05 | ChannelZ | the context doesn't matter |
| 06:00.29 | ChannelZ | anyway I gotta run off down to work to type 2 lines into a console in person *grrph* |
| 06:01.43 | gruvfunk | thx ChannelZ |
| 06:02.00 | p3nguin | The context they say to use is just a suggestion. Any context name will work, as long as that's the context you want to send calls from that peer into. |
| 06:02.22 | p3nguin | It could be any arbitrary name you choose. They chose from-callcentric. |
| 06:02.45 | gruvfunk | what I meant was that they expect you to use it [general] as teh default context |
| 06:03.03 | gruvfunk | which is how this box was setup (by somebody else) |
| 06:03.44 | gruvfunk | but then it's not matching at the peer level, it's just a default match where all calls will go that way right? |
| 06:03.45 | p3nguin | That part is silly, but see the second part of step 1. |
| 06:03.57 | p3nguin | That's right. |
| 06:04.37 | p3nguin | Maybe tomorrow I will do some sip debugging with callcentric calling. |
| 06:04.47 | gruvfunk | which is how it's setup |
| 06:05.18 | p3nguin | What you said was right, but the concept of it is not the correct way to handle it. |
| 06:05.24 | gruvfunk | with the exception of defaultuser |
| 06:05.41 | gruvfunk | agreed |
| 06:05.52 | gruvfunk | i'd like to use the way you have setup |
| 06:06.03 | gruvfunk | i don't know it all, but I'm also not a total twat |
| 06:06.12 | gruvfunk | so this one is frustrating for sure |
| 06:06.16 | p3nguin | If you know for a fact that calls from callcentric are coming from a different address, create a second peer for that host. |
| 06:06.58 | p3nguin | When I get around to testing my calling with callcentric, then maybe I can figure out why it is giving you grief. |
| 06:07.57 | p3nguin | I'll have to call my callcentric number or something and watch the debug. |
| 06:08.43 | p3nguin | It has always just worked for me, so I never bothered looking at the sip details of a call. |
| 06:10.15 | gruvfunk | yeah same here |
| 06:10.32 | p3nguin | Apparently it has never worked for you, though. :/ |
| 06:12.08 | p3nguin | Anyway, I'm off for the night. I'll debug it tomorrow to see what they are doing here. |
| 06:12.25 | gruvfunk | right |
| 06:35.57 | *** join/#asterisk myp (~andrew@mx2.admin.tomsk.ru) |
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| 06:42.46 | ChannelZ | well that was sure exciting |
| 07:01.21 | *** join/#asterisk timahvo1 (~rogue@41.212.123.197) |
| 07:07.58 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
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| 07:56.20 | *** join/#asterisk [gnubie] (~marvin.pa@117.6.131.116) |
| 07:56.25 | [gnubie] | waves |
| 07:57.56 | [gnubie] | hears nothing.. :( |
| 08:00.33 | ChannelZ | The sound of one hand clapping |
| 08:01.28 | [gnubie] | echo test fails.. :( |
| 08:02.47 | ChannelZ | Is that an asterisk issue or are we still talking about waving? |
| 08:05.34 | [gnubie] | asterisk issue |
| 08:05.59 | [gnubie] | i'm running asterisk v1.8.5 from digium's binary deb packages |
| 08:06.44 | ChannelZ | did you play a sound before doing the Echo()? |
| 08:07.12 | [gnubie] | yes |
| 08:07.20 | [gnubie] | and i don't hear anything |
| 08:07.50 | [gnubie] | my asterisk box is behind dmz having a private ip |
| 08:08.06 | [gnubie] | my sip client is also behind a nat |
| 08:11.30 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 08:13.43 | ChannelZ | You mean your * box is behind NAT, or is it actually DMZ'd (all traffic gets forwarded to that box) |
| 08:15.17 | [gnubie] | DMZ'd |
| 08:15.46 | [gnubie] | my client is behind a nat |
| 08:18.19 | ChannelZ | did you set externip and localnet in sip.conf and set nat=yes for the client's peer entry? |
| 08:18.29 | [gnubie] | yes |
| 08:18.55 | [gnubie] | externip=<public_ip> |
| 08:20.40 | ChannelZ | is your client registering? |
| 08:27.49 | *** join/#asterisk radic (~radic@tmo-096-143.customers.d1-online.com) |
| 08:34.18 | [gnubie] | yes |
| 08:38.10 | [gnubie] | ChannelZ: kindly check http://www.pastie.org/2333633 |
| 08:39.04 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 08:40.28 | [gnubie] | is the line 185 correct? it doesn't look one for me. i have opened only from 10000 to 10100 |
| 08:41.50 | ChannelZ | did you modify a bunch of IPs in here? |
| 08:42.31 | [gnubie] | only my asterisk's public ip address.. |
| 08:42.50 | ChannelZ | is that what you changed to 11.22.33.44? |
| 08:42.54 | [gnubie] | yes |
| 08:43.05 | [gnubie] | 601 is my echo test |
| 08:43.07 | ChannelZ | (and "sip.example.com") |
| 08:43.32 | [gnubie] | yes |
| 08:45.32 | ChannelZ | Do you actually have some g729 licenses? |
| 08:45.50 | [gnubie] | yes |
| 08:46.04 | [gnubie] | 2 x g.729a licenses |
| 08:47.10 | ChannelZ | well I don't necessarily see anything wrong in the dialog. Turn on RTP debug and see where it thinks it's sending packets to and from |
| 08:47.22 | [gnubie] | ok |
| 08:49.05 | [gnubie] | i will set sip debug off |
| 08:49.12 | [gnubie] | but rtp debug to on |
| 08:49.52 | ChannelZ | yup |
| 08:50.58 | [gnubie] | these are the lines i'm getting when trying to call my echo test |
| 08:51.11 | [gnubie] | <PROTECTED> |
| 08:51.12 | [gnubie] | <PROTECTED> |
| 08:51.12 | [gnubie] | <PROTECTED> |
| 08:51.12 | [gnubie] | <PROTECTED> |
| 08:51.47 | ChannelZ | nothing unusual |
| 08:52.35 | ChannelZ | did you turn on rtp with 'rtp set debug on' ? |
| 08:53.05 | [gnubie] | yup |
| 08:53.46 | ChannelZ | and |
| 08:54.03 | ChannelZ | it should have spit tons of lines out when you made a test call |
| 08:57.32 | [gnubie] | ok. i set it previously to the public ip address where i am coming from.. now, i just set it to "rtp set debug on" and now the result has a line: Sent RTP packet to 192.168.1.113:49280 (type 18, seq 044721, ts 038400, len 000020) |
| 08:58.12 | [gnubie] | it looks like i don't have rtp received on my asterisk box from my sip client |
| 08:58.23 | ChannelZ | ok so it's sending packets to the wrong place, although based on your SIP debug the device is asking for the correct RTP |
| 08:58.29 | [gnubie] | 192.168.1.113 |
| 08:58.40 | ChannelZ | that's its LAN address yes |
| 08:58.46 | [gnubie] | and i also do not hear the audio coming from my asterisk |
| 08:58.59 | ChannelZ | but it's really at 117.6.131.116 correct? |
| 08:59.00 | [gnubie] | basically, for the line: Sent RTP packet to 192.168.1.113:49280 (type 18, seq 044721, ts 038400, len 000020) |
| 08:59.41 | [gnubie] | the audio from the asterisk box sends it to my sip client at 192.168.1.113 but i don't hear anything |
| 08:59.53 | ChannelZ | but you've already told me your client is behind NAT |
| 09:00.00 | ChannelZ | on a different network |
| 09:00.11 | [gnubie] | now, if i try to talk hear, i do not see any other lines from the asterisk shell for the rtp traffic |
| 09:00.17 | [gnubie] | yes |
| 09:00.27 | ChannelZ | and is that client's external IP really 117.6.131.116? |
| 09:00.59 | [gnubie] | yes |
| 09:01.34 | ChannelZ | ok so unless you went and changed some of the references in your SIP debug paste earlier, the remote end is asking Asterisk to send to the correct IP. |
| 09:02.13 | ChannelZ | Which means that either the packets aren't making it out of the firewall your Asterisk is behind, or they aren't making it IN to the client's (probably the latter.) |
| 09:02.56 | ChannelZ | The client that is behind NAT either needs to support STUN to figure out how to drill a hole in the firewall to allow the traffic IN, or you need to port-forward manually to allow it to happen |
| 09:04.03 | [gnubie] | i tried to set my client using stun and ice but both has the same result |
| 09:06.38 | ChannelZ | well I'm pretty sure that is the problem. I discovered the other day that even though Asterisk is being told to send its traffic to the right place, if it's blocked for some reason it falls back to sending it all to the wrong IP. |
| 09:07.00 | [gnubie] | i see.. |
| 09:08.24 | ChannelZ | Why I don't know. And in your case it seems like it has to be getting it from the SIP header as opposed to the actual body (don't know what else to call it) which states the correct IP. But I haven't done extensive tests to figure out exactly how it's behaving |
| 09:10.30 | ChannelZ | You not getting any RTP back is a similar problem, either the client's firewall is blocking the outgoing traffic, or they're being blocked on the receiving end (which shouldn't be happening if needed it is fully DMZ'd as you say.) Or I guess the client could be sending them off the wrong IP too but probably not, they're just getting blocked on either end |
| 09:10.58 | [gnubie] | ok. thanks anyway.. i will try to connect to another access point.. |
| 09:11.24 | [gnubie] | i'll be back here.. i need to go out and look for another access point.. |
| 09:11.26 | [gnubie] | waves |
| 09:11.34 | [gnubie] | thanks ChannelZ |
| 09:13.21 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 09:14.31 | ChannelZ | good luck |
| 09:18.27 | [gnubie] | thanks.. |
| 09:18.28 | [gnubie] | brb |
| 09:40.59 | *** join/#asterisk sip123 (~sip123@196.1.219.211) |
| 09:41.06 | sip123 | goooooooooooood moring |
| 09:54.04 | *** join/#asterisk hariom (~hariom@117.195.160.22) |
| 09:56.35 | hariom | Anybody experienced in PRI line? I want to know what will be the destination number visible on my Asterisk if a caller dials, say on 5001, and PRI identifies that 5001 is occupied by some other caller and allots another channel (say 5002). Will asterisk see 5002 as the destination number or 5001 as the destination number (which is actually the destination number dialed by the caller) |
| 10:01.30 | *** join/#asterisk BuenGenio (~BuenGenio@203.145.92.194) |
| 10:05.13 | *** join/#asterisk djuhl30 (~quassel@121.135.82.142) |
| 10:05.41 | djuhl30 | -- Auto fallthrough, channel 'SIP/flowroute-00000023' status is 'UNKNOWN' |
| 10:05.50 | djuhl30 | What does that mean? |
| 10:07.02 | hariom | Anybody experienced in PRI line? I want to know what will be the destination number visible on my Asterisk if a caller dials, say on 5001, and PRI identifies that 5001 is occupied by some other caller and allots another channel (say 5002). Will asterisk see 5002 as the destination number or 5001 as the destination number (which is actually the destination number dialed by the caller) |
| 10:07.55 | djuhl30 | I can call out but I can't receive a call. |
| 10:12.45 | djuhl30 | <--- Transmitting (no NAT) to 169.254.79.13:5061 ---> |
| 10:12.45 | djuhl30 | SIP/2.0 489 Bad Event |
| 10:28.10 | djuhl30 | http://pastebin.com/24yvVh9K |
| 10:46.42 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 10:47.10 | *** join/#asterisk wonderworld (~ww@port-92-201-244-196.dynamic.qsc.de) |
| 10:50.29 | *** join/#asterisk irroot (~irroot@41.28.48.50) |
| 11:01.47 | singler | hariom: destination number and channel ar not related |
| 11:03.51 | *** join/#asterisk irroot (~irroot@197.105.178.196) |
| 11:39.29 | *** join/#asterisk skrusty (~skrusty@83.166.171.74) |
| 11:50.33 | *** join/#asterisk cerberus_za (~coert@196-210-206-227.dynamic.isadsl.co.za) |
| 12:05.27 | hariom | singler: thank. So you mean to say destination number will remain the same what the caller has dialed irrespective of what the channel a PRI line choose. |
| 12:07.22 | singler | yes |
| 12:23.53 | *** join/#asterisk djuhl30 (~quassel@121.135.82.142) |
| 12:25.03 | djuhl30 | I set up a phone on port 5061, so if I am using a sip account from flowroute.com that is on 5060 it should forward to port 5061 right? |
| 12:25.14 | djuhl30 | or am I just confused? |
| 12:26.14 | weinerk | Please help: from inside an AGI script - I need to play music while executing some activity. |
| 12:28.48 | djuhl30 | I can call out, but I can't receive calls |
| 12:39.12 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 12:51.44 | hariom | djuhl30: do you have appropriate entry in your extensions.conf and sip.conf ? |
| 12:52.04 | djuhl30 | As far as I can tell |
| 12:52.06 | hariom | your asterisk should make a call to your phone on 5061 from extensions.conf |
| 12:52.12 | djuhl30 | Using Skype to check |
| 12:52.40 | djuhl30 | To: <sip:+821064409475@flowroute.com>;tag=gK0227a470 |
| 12:52.55 | djuhl30 | oops |
| 12:52.58 | djuhl30 | something else |
| 12:53.11 | *** join/#asterisk agnogenic (~agnogenic@c-67-176-218-28.hsd1.il.comcast.net) |
| 12:53.24 | hariom | Just think how would your asterisk know how to call to your phone ? |
| 12:54.02 | djuhl30 | register flowroutes sip address as a client |
| 12:54.37 | djuhl30 | and when it receives a call from the did I rented route it locally would be my guess |
| 12:55.07 | hariom | [SIP service provider] -> Asterisk (sip.conf that says pass this call to context xyz) -> extensions.conf (that as a context [xyz] and that context makes a dial to your phone says 1000) -> sip.conf (which has an entry for your phone with number 1000) |
| 12:55.10 | djuhl30 | It tries to but just hangs up |
| 12:55.21 | djuhl30 | huh? |
| 12:56.38 | djuhl30 | Yeah basically |
| 12:56.40 | hariom | djuhl30: your asterisk will figure out what channel to use. If you are calling from a sip provider (or says external sip service to your phone connected to asterisk or somewhere else), then asterisk will use chan_sip as a channel |
| 12:57.02 | djuhl30 | Ok |
| 12:57.17 | djuhl30 | But as far as I can tell, the diaplan has it. |
| 12:57.46 | hariom | it goes to sip.conf and from there to extensions.conf to find what action to do and if you say dial to your phone, it will find where is your phone define (channel). if it is sip phone, then it again goes to sip.conf and make the relavent action. |
| 12:58.25 | hariom | First try it out with two locally connected softphones |
| 12:58.49 | hariom | Google to know how to connect asterisk to sip provider. |
| 12:59.56 | djuhl30 | hariom: what do you think I was doing all this time. Even followed the instructions from flowroute |
| 13:00.09 | djuhl30 | sip.conf: http://pastebin.com/jHZGnWCh |
| 13:00.28 | hariom | djuhl30: I don't know what you were doing. :) |
| 13:00.40 | djuhl30 | hariom: I can get two softphones connected locally |
| 13:00.47 | djuhl30 | I've done it. |
| 13:01.08 | hariom | gr8. Try more on the lines I have explained. |
| 13:01.53 | djuhl30 | Hold on |
| 13:01.54 | hariom | may be your firewall or nat configuration is not correct if you have done things right till now. |
| 13:03.15 | *** join/#asterisk Guest8383 (~Geek@unaffiliated/cain) |
| 13:03.52 | djuhl30 | Asterisk box has no firewall |
| 13:04.15 | djuhl30 | And I got ports 5060/5061 udp open on the router I am using |
| 13:04.36 | agnogenic | So in theory would a 300mhz processor be able to handle to simultanious calls? |
| 13:04.49 | agnogenic | two* |
| 13:04.54 | djuhl30 | Now my number keeps ringing.. |
| 13:05.13 | djuhl30 | But nothing from the softphone |
| 13:08.55 | djuhl30 | Calls go out fine |
| 13:09.01 | djuhl30 | talking with my sister |
| 13:09.23 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 13:14.29 | *** join/#asterisk sourcode (~code@ppp-58-11-112-110.revip2.asianet.co.th) |
| 13:29.15 | weinerk | Please help: from inside an AGI script - I need to play music while executing some activity. |
| 13:30.44 | *** join/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 13:30.59 | jmls | afternoon all |
| 13:31.03 | djuhl30 | hi |
| 13:31.06 | jmls | <PROTECTED> |
| 13:32.16 | jmls | curl command line equivalent is curl -X DELETE "http://localhost/yadayada" |
| 13:33.27 | Gugge | my guess is you dont |
| 13:34.01 | jmls | bugger |
| 13:34.02 | Gugge | but you could use SHELL() to call that command : |
| 13:34.05 | Gugge | :) |
| 13:34.19 | jmls | true. expensive on resources, but true. |
| 13:34.38 | Gugge | and if you dont need feedback, you could use SYSTEM() |
| 13:35.23 | Gugge | or you could modify your webservice to use a normal GET url |
| 13:35.45 | jmls | it's not my webservice, it apache's ;) |
| 13:36.44 | Gugge | Im pretty sure CURL() only does GET's |
| 13:38.16 | jmls | no, it does POSTS as well |
| 13:38.28 | jmls | [Description] |
| 13:38.28 | jmls | <PROTECTED> |
| 13:38.28 | jmls | <PROTECTED> |
| 13:38.38 | jmls | (core show function CURL) |
| 13:39.15 | jmls | -X only modifies the method keyword in a request, it doesn't change the underlying behavior. Thus if you use "curl -XDELETE [URL]" it acts like a GET but sends DELETE instead. If you use "curl -d foo -XDELETE [URL]" it acts like a POST but sends a DELETE instead. |
| 13:39.42 | jmls | (curl.haxx.se) |
| 13:42.20 | Gugge | You are right, GET and POST |
| 13:46.57 | djuhl30 | extenstions.conf: http://pastebin.com/CubT6UVL |
| 13:48.28 | djuhl30 | So if it is a firewall, I need to know what to open up. All I know is I am talking to my dad right now |
| 13:48.37 | djuhl30 | Everything goes out. |
| 13:52.02 | djuhl30 | What is the port range for asterisks? |
| 14:00.43 | gravin | Hello, is it possible to have to ATAs - PAP2t, one connected to credit card terminal, another to the credit card controller, all connected to asterisk, enabled t.38 pass through |
| 14:00.52 | gravin | two* ATAs |
| 14:01.12 | gravin | something like FOIP, we dial the extension |
| 14:01.26 | gravin | it talks to each other, just the message is too short and it disconnects |
| 14:04.22 | Gugge | does the creditcard terminal send a fax? |
| 14:05.32 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 14:05.33 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 14:05.34 | [sr] | hi |
| 14:10.21 | [sr] | who wants to came ride bikes with me? need company :p |
| 14:11.35 | *** join/#asterisk djuhl30 (~quassel@121.135.82.142) |
| 14:11.53 | djuhl30 | One day I'll get it. |
| 14:12.02 | djuhl30 | But I suppose I need sleep now. |
| 14:12.18 | djuhl30 | I guess calling out is half way |
| 14:23.01 | *** join/#asterisk Tim_Toady (~fuzzy@79.103.14.172) |
| 14:24.30 | coppice | garvin: two problems with that idea. one is that A PAP2T doesn't handle T.38. the other is T.38 only works for FAXes |
| 14:27.36 | jmls | so, no go with the -X DELETE then ? |
| 14:29.26 | Gugge | jmls: only with SHELL or SYSTEM |
| 14:35.53 | jmls | hmm. I think that the only thing that needs adding would be CURLOPT_CUSTOMREQUEST to the CURLOPT function |
| 14:36.42 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 14:44.28 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 15:00.26 | *** join/#asterisk BuenGenio (~BuenGenio@cm61-10-82-188.hkcable.com.hk) |
| 15:17.28 | *** join/#asterisk radic (~radic@tmo-097-108.customers.d1-online.com) |
| 15:27.32 | *** join/#asterisk eugeneoden (~goden@99-62-173-93.lightspeed.austtx.sbcglobal.net) |
| 15:30.57 | *** join/#asterisk rethus (~suther@p549A6764.dip.t-dialin.net) |
| 15:31.11 | rethus | ikts some time ago, that i write this line... |
| 15:31.17 | rethus | exten => auth,1,GotoIf($["${CALLERID(num)}" == "dev1"]?auth,auth,pinRequest:auth,auth,congRequest) |
| 15:32.17 | rethus | my exten is named auth... but is this an error, that first if-condition has 3 ","? |
| 15:32.24 | rethus | i'm not sure anymore |
| 15:34.23 | rethus | ahh, i remember... first auth is context, second one is extension and last value is a label "pinRequest) |
| 15:35.22 | rethus | if i change something in extensions.conf have i to restart asterisk with "core restart now" or core reload now works too? |
| 15:39.50 | rethus | i got no messeages anymore on my asterisk cli, if someone call in , or if i connect with my sipphone. |
| 15:39.56 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:40.11 | rethus | how can i activate this? |
| 15:40.53 | p3nguin | core set verbose 1 |
| 15:40.55 | Tim_Toady | rethus: diaplan reload will make it |
| 15:41.38 | p3nguin | And no you don't have to do core restart now to reload extensions.conf. Just use dialplan reload. |
| 15:41.59 | rethus | k, i try both |
| 15:43.53 | rethus | <PROTECTED> |
| 15:44.09 | rethus | even if accepted connection |
| 15:44.27 | p3nguin | SIP messages can be viewed by enabling sip debug. sip set debug on |
| 15:45.32 | rethus | k, but that was not the messages i mean. |
| 15:45.55 | rethus | before, if someone call me, i got some input on asterisk cli... not such detailed like sip-debug |
| 15:47.29 | p3nguin | The core verbose messages normally show that. If core set verbose 1 doesn't show it, turn it up to 4 to see the maximum verbosity. |
| 15:47.42 | p3nguin | core set verbose 4 will show you everything. |
| 15:48.22 | rethus | great, that was what i searching for. thanks a lot |
| 15:48.44 | p3nguin | I thought 1 would show you the calls, but I guess it wasn't enough. |
| 15:49.32 | rethus | 4 shows me the single executing steps.. thats what i need |
| 15:50.21 | p3nguin | I'll give you another tip: if you connect to the CLI using asterisk -r, use core set verbose 4 to increase verbose messages to level 4; or you can connect to the CLI using asterisk -rvvvv to do the same thing upon connecting. |
| 15:50.52 | rethus | good hint, thanks |
| 15:59.41 | rethus | if i have in extension exten => callin,2,NoOp(Pin is ${PINENTRY}) |
| 16:00.03 | rethus | normaly this string "Pin is <number>" should be output to asterisk-cli ? |
| 16:00.43 | p3nguin | It's a NoOp(), so you can only see it if your verbose it turned up enough to see the steps of the dial plan executing. |
| 16:01.00 | p3nguin | The correct app to see it printed to the CLI is Verbose(). |
| 16:01.35 | rethus | ok, how to use... Verbose("mystrin ${VAR}") ? |
| 16:01.37 | *** join/#asterisk cusco (~tralala@a83-132-168-178.cpe.netcabo.pt) |
| 16:01.41 | p3nguin | yes |
| 16:01.55 | rethus | great, thanks |
| 16:02.31 | rethus | but verbose is shown each time, while NoOp only shown on "set verbose 4" ? |
| 16:02.52 | p3nguin | And if you need to specify the minimum core verbose level in which it will actually print it, you can use something like Verbose(2,say stuff here) to make it print on level 2 and up. |
| 16:03.09 | rethus | k, thats a good hint |
| 16:03.10 | p3nguin | The NoOp() will only be shown when verbose it set high enough to see dial plan steps executing. |
| 16:03.30 | rethus | in such way i can do a little debug-level |
| 16:03.53 | p3nguin | Verbose(4,stuff here) would make it only print on core verbose 4 and up. Et cetera. |
| 16:04.17 | p3nguin | For verbose level 9000, Verbose(9000,stuff here). |
| 16:04.21 | p3nguin | 9000 and up |
| 16:05.10 | p3nguin | I almost always use Verbose(stuff here) without a level number so it is printed every time regardless of the verbose level I am using. |
| 16:05.17 | rethus | other question... if i call via my telephone the sip-number, i didn't hear any sound... asterisk directly answer. |
| 16:05.31 | rethus | how can i set to her two till trhee times the beep |
| 16:05.40 | p3nguin | ringing sounds? |
| 16:05.46 | rethus | yes |
| 16:06.36 | p3nguin | In the US, the ringing sound is 2 seconds on 4 second off, for a total of 6 seconds for each ring cycle. To hear three ring cycles, you can put a Wait(18) before the app that answers in your dial plan. |
| 16:07.07 | rethus | you mean in front of Answer(); ? |
| 16:07.14 | p3nguin | yes |
| 16:07.45 | p3nguin | If you are not using US ringing sound, you'll need to know how long your ring cycles are to calculate three of them. |
| 16:08.14 | cusco | there is a coment on dahdi zonedata.c about those |
| 16:08.27 | rethus | and the number are mili-seconds? |
| 16:08.33 | p3nguin | seconds |
| 16:08.39 | rethus | ah, ok |
| 16:08.53 | p3nguin | Wait(18) = 18 seconds, or three ring cycles with US ringing sound. |
| 16:09.53 | p3nguin | I'd probably reduce it to 14 if I wanted three rings, because I'd want two full cycles, plus the third ring sound without the 4-second pause after. |
| 16:10.43 | rethus | k, i have the german ringtone, its only a ring for 3 seconds on, and 3 seconds off |
| 16:10.56 | rethus | does the exten (if not used n but 1,2,3,4) to be incremental by 1 or can i also use 1,10,20 to add some lines later without rewriting each following linenumber? |
| 16:11.36 | rethus | soon i want to switch to ael, but i for now i have the old format... |
| 16:11.38 | p3nguin | Now is probably a good time to let you know that you don't have to renumber your entire dial plan... unless you are using a really old Asterisk. Only priority 1 needs to be explicitly written; all other priorities can be n to increment each time. |
| 16:11.59 | p3nguin | Oh, you were just mentioning that when I was trying to type it. |
| 16:13.04 | p3nguin | You'll either use n or numbers incrementing by 1. 1,10,20 will not increase from 1 to 10 to 20 without apps that do jumping or using Goto(). |
| 16:13.47 | p3nguin | The correct thing to do to prevent having to renumber is to use 1 for the first line and n for every other line. |
| 16:14.19 | rethus | whats with jumping-points? now i switch into a php-agi-script check something and rejumb tu callin,5 |
| 16:14.34 | rethus | if all are n, i can't jump to it |
| 16:14.38 | rethus | is there another way |
| 16:15.08 | p3nguin | Some apps do jumping, like n+100, where you actually have to write 101 for example. |
| 16:15.18 | p3nguin | Jumping is pretty ancient, though. |
| 16:15.57 | rethus | i have in auth something like that: |
| 16:16.05 | rethus | ;exten => auth,n(pinRequest),Read(PIN,conf-getpin,5); |
| 16:16.07 | p3nguin | The proper way to arrive at a specific priority when they are all n is to use labels. |
| 16:16.22 | rethus | and with a GotoIf, i jump to "pinRequest" |
| 16:16.23 | p3nguin | The label on what you just showed me is 'pinRequest' |
| 16:16.30 | p3nguin | That's the right way to do it. |
| 16:16.42 | rethus | is this possible with a normal execute statement |
| 16:17.01 | rethus | i only know this in context with gotoif |
| 16:17.03 | p3nguin | Those are called priority labels, and they are used to go to priorities in the extension where you're using n on them. |
| 16:17.38 | p3nguin | With numbered priorities, you can just goto the number; with n, you use a label and goto the label. |
| 16:18.02 | rethus | k, i'll try this.. would be the best choice i think |
| 16:19.19 | p3nguin | Any app that requires moving execution to a specific priority should be able to go to the lable. |
| 16:19.23 | p3nguin | label |
| 16:20.17 | rethus | btw. now i have wait (10), nothing happend... completly silent... for 10 seconds... than the system go on |
| 16:21.28 | p3nguin | Okay, that's strange. Maybe you have to play fake ringing sound. Use Answer(), then Ringing(), then Wait(10)... then everything else. |
| 16:24.32 | rethus | k, thats works. |
| 16:24.38 | rethus | but like u say, its strange |
| 16:25.01 | p3nguin | I wonder if it just doesn't work the other way in your country. |
| 16:25.13 | p3nguin | Maybe the telco doesn't like it or something. |
| 16:25.30 | p3nguin | I'm not familiar with how it works outside of USA. |
| 16:26.07 | rethus | it may be a missconfiguration of my asterisk. But i'll ask some other asterisk-users from germany in a forum, to get a clean solution. But for now, it works... thanks for this hint. |
| 16:26.50 | p3nguin | Okay. Yeah, normally the telco rings while your Asterisk does Wait(10), then you Answer(). |
| 16:27.18 | p3nguin | But if you can get it to answer right off, you can then play your own ringing sound while you wait. |
| 16:27.40 | WIMPy | What are you up to? |
| 16:28.00 | p3nguin | He's trying to get three ringing cycles before his Asterisk answers. |
| 16:28.21 | WIMPy | BRI? |
| 16:28.24 | rethus | i don't hear ringsound here in germany if i do wait (10), only if i call Ringing() after Answer() |
| 16:28.27 | p3nguin | I thought it was SIP. |
| 16:28.41 | rethus | from my normal phone |
| 16:28.47 | p3nguin | I was under the impression it was using an ITSP. |
| 16:28.50 | rethus | but ovver sip |
| 16:28.52 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 16:29.04 | WIMPy | Sure. No Ringing(), no ringing. |
| 16:29.22 | p3nguin | Here in the US, if I wait, the telco is ringing. |
| 16:29.24 | rethus | WIMPy: you mean for germany? |
| 16:29.36 | rethus | good to know |
| 16:29.54 | WIMPy | p3nguin: That's just because they have no idea if something is ringing or no, so they just assume it. |
| 16:29.58 | p3nguin | If I start out with Wait(14), the caller will hear three rings before my system does the next step. |
| 16:30.06 | WIMPy | not |
| 16:30.37 | p3nguin | So in Germany, you must indicate ringing to make the telco ring? |
| 16:31.11 | WIMPy | If you're using anytinh more than POTS. |
| 16:31.34 | p3nguin | Does he need to keep the Answer() before the Ringing()? Right now, I had him Answer() the line, then Ringing() and Wait(10) so it would play 10 seconds worth of ring cycles. |
| 16:31.52 | WIMPy | With ISDN or SIP the called party indicates that it is ringing. |
| 16:32.04 | WIMPy | No, no Answer. |
| 16:32.05 | p3nguin | If the telco accepts and understands 180 Ringing, maybe the Answer() isn't needed? |
| 16:32.21 | WIMPy | rethus: How do you get the call in to your Asterisk? |
| 16:32.31 | p3nguin | Like I said, I don't know how telcos outside the US do things. |
| 16:32.37 | rethus | sipgate |
| 16:32.52 | rethus | and the client call with normal phone |
| 16:33.59 | WIMPy | I'd expect that Ringing() should do it. But off course, with SIP you never know what the other end will accept or require. |
| 16:34.31 | p3nguin | The only disadvantage that I see with Answering first is that billing seconds start at that time. |
| 16:34.48 | WIMPy | yes |
| 16:35.00 | p3nguin | For some people, that's not good. |
| 16:35.10 | rethus | WIMPy: how would i do it... first Ringing() or ffirst Answer() than Ringing() followed ba Wait() ? |
| 16:35.25 | WIMPy | And you start RTP. |
| 16:35.30 | p3nguin | I know what I'd do: test it both ways (with and without Answer() first. |
| 16:35.42 | WIMPy | Ringing() then Wait(). |
| 16:36.01 | rethus | ah, ok, that prevent from billing seconds ;) |
| 16:36.07 | p3nguin | Anwering() first will bring the line "Up." RTP starts and then Ringing() plays the ringing sound while you Wait(). |
| 16:36.11 | WIMPy | You can also experimet with Progress() and Proceeding(). |
| 16:36.32 | rethus | WIMPY: how do you made it... |
| 16:36.58 | p3nguin | If you don't Answer() first, Ringing() might tell the telco to play ringing sounds. I'd test it. |
| 16:37.17 | *** join/#asterisk kannan (kann@123.238.238.126) |
| 16:37.21 | WIMPy | That's the idea. |
| 16:37.50 | rethus | k, seems to work. |
| 16:38.03 | rethus | now i have ringing(), wait(10) Answer() |
| 16:38.15 | p3nguin | If that works, that is the way I would leave it. |
| 16:38.57 | kannan | hello, I have a SIP brute force attack. I have checked the voip-info for fail2ban and also someother PERL script. The only thing is the server is on a public IP and is co-located . The CLI says sip registration from "100"@my-server-ip failed , tho it is from amazzon ecs Ip |
| 16:39.00 | rethus | yeeha, thats works |
| 16:39.11 | WIMPy | Instead of Wait(10); Answer() you can do Answer(10000), BTW. |
| 16:39.36 | kannan | even if i used uptables to ban the originating IP , it still continues |
| 16:39.48 | kannan | iptables , i meant |
| 16:40.17 | p3nguin | You did it wrong. |
| 16:40.27 | WIMPy | kannan: Then you must have used iptables incorrectly. |
| 16:40.40 | p3nguin | See also: you did it wrong. |
| 16:40.42 | p3nguin | :D |
| 16:41.18 | rethus | answer(1000) seems not to work |
| 16:41.21 | rethus | no ringtone anymore |
| 16:41.37 | p3nguin | What do you expect Answer(1000) to do? |
| 16:41.37 | WIMPy | 10000. It's ms. |
| 16:41.44 | WIMPy | Did you leave the Ringing()? |
| 16:42.01 | kannan | ok thanks , i will check it |
| 16:42.02 | p3nguin | Answer(10000) will Answer() the line and then wait 10 seconds before going to the next step in the dial plan. |
| 16:42.05 | rethus | jes, i have choose 6 seconds =6000 |
| 16:42.08 | rethus | keep ringing |
| 16:42.18 | WIMPy | Oh, was it that way round? |
| 16:42.37 | WIMPy | Yes. You're right. That was wrong. |
| 16:42.39 | p3nguin | You have to use Wait() before an Answer() if you want to wait BEFORE answering. |
| 16:43.15 | p3nguin | Answer(10000) = Answer(), Wait(10) |
| 16:47.28 | p3nguin | Is 4G the largest SD card that isn't SDHC? |
| 16:47.37 | [sr] | howdy WIMPy |
| 16:48.32 | rethus | up on 8GB are all SDHC i think. |
| 16:49.19 | p3nguin | I don't think my Wii works with SDHC, so I need the largest regular SD I can find. I think the largest is 4G, but I'm not sure. |
| 16:50.04 | [sr] | i have a mini-sd of 8GB |
| 16:50.14 | p3nguin | It's not SDHC? |
| 16:50.18 | [sr] | on the phone |
| 16:50.43 | rethus | <PROTECTED> |
| 16:51.07 | [sr] | p3nguin: well no idea |
| 16:51.14 | p3nguin | :) |
| 16:51.49 | rethus | i have found a german news... they say: up to firmware-version 4.0 its no problem to use sdhc |
| 16:51.54 | rethus | http://www.computerbild.de/artikel/cbs-News-Demos-Patches-Nintendo-Wii-Firmware-4.0-SD-HC-Speicherkarten-Support-4201283.html |
| 16:52.07 | rethus | maybe translate it with google-translate to read it |
| 16:52.15 | [sr] | p3nguin: sdhc (mini) |
| 16:52.39 | rethus | this article is from march 2009 |
| 16:52.48 | rethus | so you have good chaces that it works for you |
| 16:53.09 | p3nguin | 4.0 and up, or up to 4.0? |
| 16:53.29 | rethus | 4.0 and up |
| 16:53.59 | p3nguin | Okay, I think mine is 4.3, so I'll try an SDHC in it. |
| 16:54.11 | p3nguin | Any idea what the maximum capacity would be? |
| 16:54.44 | rethus | iff sdhc is supportet, any capacity should work |
| 16:54.52 | rethus | same technology |
| 16:54.53 | p3nguin | Even 64GB? |
| 16:55.13 | rethus | i think so,,, but i have no wii to try it. |
| 16:55.29 | rethus | maybe try it with your 8GB microsd |
| 16:55.38 | p3nguin | That's [sr]. |
| 16:56.10 | p3nguin | I only have small capacity SD, but if SDHC will work, I'll get a big one. |
| 16:57.43 | WIMPy | The FS might have a limit. |
| 16:58.15 | p3nguin | It should use vfat/fat32. |
| 17:00.28 | WIMPy | So that makes Max 2TB. |
| 17:02.43 | [sr] | whats up WIMPy? |
| 17:04.21 | WIMPy | I think I finally found out how to make a working dialplan. |
| 17:04.29 | p3nguin | yay! |
| 17:07.59 | WIMPy | Ja, I guess I should write something about that... |
| 17:12.14 | [sr] | whould something like, have one port of the NT connect to the asterisk machine |
| 17:12.27 | [sr] | and the other port of the NT, connected to an ISDN phone, with MSN's configured |
| 17:12.38 | [sr] | to have a backup system in case the asterisk machine dies |
| 17:12.41 | p3nguin | BREAKING NEWS: WIMPy writes dial plan for Asterisk |
| 17:14.08 | p3nguin | I need lunch. |
| 17:17.45 | WIMPy | [sr]: If it's ptmp, sure. |
| 17:17.58 | WIMPy | p3nguin: "working" was the keyword. |
| 17:18.57 | [sr] | WIMPy: the ISDN phones are powered by the NT, are they? |
| 17:19.09 | WIMPy | Yes |
| 17:20.57 | WIMPy | Well, some require their own PSU, e.g. if they include some fancy stuff like a DECT base. |
| 17:21.06 | [sr] | have to think on that |
| 17:21.10 | [sr] | no DECT |
| 17:21.13 | [sr] | on my case |
| 17:21.25 | [sr] | what do people do for backup scenarios similar to this one? |
| 17:21.40 | WIMPy | Make sure you get one that can operate on emergency power supply. |
| 17:22.11 | [sr] | when the asterisk dies and its gonna take 12h to solve the problem (example) |
| 17:22.31 | WIMPy | Use the backup :-) |
| 17:22.59 | WIMPy | Or enjoy the silence :-) |
| 17:23.02 | [sr] | hehe |
| 17:23.13 | [sr] | on this case can't exists silence, it's a critital stuff |
| 17:23.24 | [sr] | tell me, the isdn phone it's a solutions, and others? |
| 17:24.05 | WIMPy | That's th good thing about ISDN phones. Just plug the internal line to the NT and continue. |
| 17:24.57 | [sr] | i havent tested... if I have both, asterisk and the ISDN phone, will they ring at the same time? |
| 17:25.26 | WIMPy | Some of the better small plastic PBXes have a relais to short circuit the (first) internal and external S0 if they aren't powered so your phone(s) still work if it without power. |
| 17:25.34 | WIMPy | Yes |
| 17:26.06 | [sr] | that nice |
| 17:26.19 | [sr] | so i thing i'll setup an isdn phone next to the SIP phone |
| 17:26.28 | [sr] | with the volume on 0% on the ISDN one :) |
| 17:28.03 | [sr] | maybe thats the best backup method |
| 17:28.51 | WIMPy | One of the reasons I prefer ISDN over SIP phones. |
| 17:29.27 | [sr] | WIMPy: but for a 50 phone instalation... its better SIP |
| 17:29.44 | [sr] | how are you going to to 50 ISDN extensions? gonna be complicated |
| 17:29.46 | WIMPy | Certainly easier. |
| 17:30.11 | WIMPy | 3 OctoBRI? |
| 17:30.48 | WIMPy | Don't know how much electricity costs in your place. But here it would make a huge difference on your bill. |
| 17:32.53 | [sr] | with all ISDN phones right? |
| 17:33.07 | WIMPy | yes |
| 17:33.22 | [sr] | SIP won't be cheaper also.. |
| 17:33.31 | [sr] | even with the new core i3 cpu's |
| 17:33.39 | [sr] | it's the server machine, the PoE switch |
| 17:33.59 | [sr] | but a traditional PBX also consumes power.. |
| 17:34.29 | WIMPy | It's not only about the server. It's the phones themselves. |
| 17:35.19 | WIMPy | But you should indeed put the switch ports in to the calculation as well. Switches take a lot of power as well. |
| 17:36.07 | WIMPy | I'd expect a SIP phone to cost around 1 EUR/month more than an ISDN phone. |
| 17:38.20 | [sr] | in your country... |
| 17:38.26 | [sr] | on here would be 2 for sure :p |
| 17:38.27 | WIMPy | Yes. |
| 17:38.37 | [sr] | i think we are the country that more power pay worldwide |
| 17:38.57 | WIMPy | Well, it's more like 1.30 if you use a decent switch and not just the small plastic one. |
| 17:39.12 | [sr] | if i tell you that in 1995 i used to pay +- 10 of power per month, and now pay +- 50, with same people, same devices |
| 17:40.05 | WIMPy | It has also doubled here in the last 10 years. |
| 17:40.15 | WIMPy | But factor 5 is really bad. |
| 17:40.28 | [sr] | it is. |
| 17:40.43 | [sr] | in here the minimum salary is 485/month |
| 17:41.05 | [sr] | and there for sure its 1000 and so |
| 17:41.19 | WIMPy | We don't have a minimum. |
| 17:41.52 | WIMPy | But we have those infamous 400-Jobs. |
| 17:42.02 | [sr] | didn't knew |
| 17:42.57 | WIMPy | Up to 400 you don't pay social insurances. |
| 17:43.11 | [sr] | how things are, i wanna go to mars! |
| 17:43.48 | WIMPy | ja |
| 17:44.03 | [sr] | here all jobs pay social security, BUT, doesn't help much, if you want to survive when something bad happens, you have to go to a particular doctor |
| 17:44.08 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:44.15 | [sr] | or have a particular health insurance |
| 17:45.03 | [sr] | if you have a simple cold, and go to the public, you enter at 9am, and get out at 11pm!! |
| 17:45.07 | *** join/#asterisk mutex7c (~mutex7c@HSI-KBW-095-208-202-191.hsi5.kabel-badenwuerttemberg.de) |
| 17:45.44 | WIMPy | :-( |
| 17:46.07 | [sr] | oh well, let talk about ISDN and SIP , its better than this!! |
| 17:46.52 | WIMPy | With a normal doc it's not that bad here. But if you need to see a specialist, chances are won't see him in your lifetile with the public insurance. |
| 17:47.22 | ChannelZ | Don't tell Obama |
| 17:48.14 | WIMPy | doesn't have his number. |
| 17:48.20 | [sr] | WIMPy: same here, if you need a specialist and can (), go to private or insurances |
| 17:48.57 | [sr] | in fact, is it wasn't the private, i may not be here today |
| 17:49.18 | [sr] | as they say, money talks, ......... walks |
| 17:49.47 | [sr] | the funny is that most of the privates work in public, but act != on the two places.. |
| 17:50.11 | [sr] | and have their own particular medical office... |
| 17:50.17 | [sr] | ironic |
| 17:50.48 | mutex7c | Hello everybody - I just wanted to introduce myself, since I am new to the channel. I am working intensely with asterisk and I am looking forward to share knowledge and experience here with you :) |
| 17:50.55 | WIMPy | Similar here. |
| 17:52.15 | cusco | similar here lol |
| 17:52.44 | [sr] | good, i don't feel alone!! |
| 17:52.45 | [sr] | hehe |
| 17:52.57 | ChannelZ | Welcome to the crazy! |
| 17:53.24 | mutex7c | I would rather say the chosen few - given how hard it is to get good staff with know-how around * |
| 17:53.27 | mutex7c | ;) |
| 17:54.42 | mutex7c | It is also nice - I can't remember when I stopped using IRC - that must have been more than 10 years ago. But I heard, this channel is the ultimate place to be :-O |
| 17:54.54 | [sr] | mutex7c: it is |
| 17:54.55 | ChannelZ | Kickin' it old-school |
| 17:55.20 | WIMPy | Yes, you find lots of us, who don;t have a clue, either :-) |
| 17:56.32 | mutex7c | That is great - you can't learn anything, if you think you know everything ;) |
| 17:56.59 | WIMPy | There are lots of things to lears with Asterisk. |
| 17:59.52 | [sr] | one funny thing i've been noticed the last month's |
| 17:59.58 | [sr] | lots of public services using asterisk |
| 18:00.04 | mutex7c | No doubt about that. I am curious, which problems might come up here occasionally ... I have been developing with asterisk for 2+ yrs now .... |
| 18:00.12 | [sr] | they don't even get the work to change the default MOH music |
| 18:00.37 | [sr] | was calling the airport to confirm one thing, and voilá, asterisk with default MOH music |
| 18:00.41 | ChannelZ | It's just that good. |
| 18:00.47 | mutex7c | Big security problems there too, I suppose ;) |
| 18:15.13 | ChannelZ | So it seems to be not entirely possible to create a GTalk extension in the form of (someone)@gmail.com where (someone) is a pattern... so that you could call via Google Talk from a softphone (where you could type it in) or even from redial on a SIP phone from a previous incoming GTalk call. |
| 18:23.37 | *** join/#asterisk knarfly (~vlad@c-65-34-181-97.hsd1.fl.comcast.net) |
| 18:33.13 | _Raptor_ | ;;btc |
| 18:33.18 | _Raptor_ | sry |
| 18:40.46 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
| 18:53.35 | rethus | i have this line: |
| 18:53.37 | rethus | exten => auth,n(countPin),GotoIf($[${PINENTRY} < 3]?auth,auth,pinRequest:auth,auth,hangup) |
| 18:53.59 | rethus | but the system always get to "hangup", cause PINENTRY is " " at startup |
| 18:54.20 | rethus | does empty not match the request "< 3" ? |
| 18:56.58 | p3nguin | no |
| 18:57.00 | mutex7c | try and test it by manually setting the variable to something less than 3 a line before. I think this won't work with an empty string, since the cast to integer will not work |
| 18:57.07 | p3nguin | null values are not less than anything. |
| 18:57.15 | mutex7c | nothing equals NULL |
| 18:58.36 | rethus | not for php. if you do i=0; i<5; i++ thats a normal construct in programming |
| 18:58.42 | rethus | so 0 is less than 5 |
| 18:58.49 | p3nguin | 0 is not null |
| 18:58.49 | rethus | and so on |
| 18:58.52 | p3nguin | null is NOTHING |
| 18:58.55 | p3nguin | 0 is a value. |
| 18:58.55 | mutex7c | yep |
| 18:58.58 | mutex7c | common mistake |
| 18:59.12 | mutex7c | a "0" string will be cast to 0 as an integer value |
| 18:59.15 | rethus | does asterisk differ between 0 an null? |
| 18:59.23 | p3nguin | null is null, 0 is a value of 0 |
| 18:59.33 | mutex7c | but something not a valid "number" won't be automatically cast to 0 |
| 18:59.39 | weinerk | Please help: from inside an AGI script - I need to play music while executing some activity. |
| 18:59.51 | mutex7c | but will be NULL, which cannot be an operand for this type of operation |
| 19:01.06 | p3nguin | If PINENTRY can ever be null, you could set PINENTRY to a value before doing the comparison. |
| 19:01.21 | p3nguin | exactly like you did in the php line above. |
| 19:01.25 | rethus | yes maybe initiate with 0 |
| 19:01.37 | p3nguin | i=0 ... this sets i to 0 so that it it not null. |
| 19:01.51 | p3nguin | If you do not set i=0 there, it's null and the rest of the command will fail. |
| 19:04.41 | mutex7c | looks strange anyway - what do you intend to do exactly? If you try to enforce at least a pin of 4 digits length, this won't work that way anyways ... you'll need to get the length of the variable and not its value |
| 19:07.52 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 19:10.25 | p3nguin | I thought he was trying to see if the value was less than 3. 2, 1, or 0 |
| 19:11.02 | rethus | i readout the var... it is inkcrement by each wrong pin-entry |
| 19:11.15 | rethus | if 3 wrong invalid pins => hangup |
| 19:11.27 | p3nguin | You could use Read() for that. |
| 19:11.33 | p3nguin | if you wanted. |
| 19:11.36 | mutex7c | ah ok - then never mind |
| 19:11.42 | rethus | i use read |
| 19:11.47 | rethus | for reading the pin. |
| 19:12.12 | rethus | but i increment the var PINENTRY in my agi-script, if wrong passwort was enterd |
| 19:12.24 | p3nguin | Then start out by setting it to 0. |
| 19:12.28 | mutex7c | initializing the var is the way to go anyways |
| 19:13.21 | mutex7c | that is the problem with scripting - "lazy casting" of string vars to numeric vars is often a source for problems |
| 19:20.28 | rethus | have i do SET(PINENTRY=0) inside of the context or inside of general ? |
| 19:22.08 | *** join/#asterisk Sertys (~cwalker@hotel-palas.com) |
| 19:23.07 | mutex7c | neither - inside the extension block |
| 19:23.46 | rethus | means inside of [auth] for me? |
| 19:24.15 | mutex7c | in this case you should check, if the var is NULL and then set it to 0 accordingly to initialize |
| 19:25.15 | p3nguin | I use ExecIf() for that a lot of times. |
| 19:26.48 | p3nguin | ExecIf($[${ISNULL(${myVar})}],Set,myVar=0) |
| 19:26.54 | p3nguin | something like that, anyway. |
| 19:26.59 | p3nguin | (not checked for syntax) |
| 19:27.28 | rethus | k, i have a lock |
| 19:27.44 | p3nguin | Newer Asterisk has changed the ExecIf syntax slightly. |
| 19:27.46 | mutex7c | then - fire at will *chuckles* |
| 19:29.42 | mutex7c | Europe is dark by now, so I will be calling it a day ... cu |
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| 19:57.03 | MRH2 | hi can i check that it is still not recommended to run asterisk in a vm (if using call recording, conferencing etc...) |
| 19:58.07 | p3nguin | confirmed |
| 19:58.12 | MRH2 | thanks |
| 19:58.31 | p3nguin | Although, many people do it. |
| 19:59.05 | MRH2 | many people also think g729 is good enough though |
| 19:59.06 | MRH2 | ;) |
| 20:02.00 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 20:02.02 | MRH2 | are there any notes on people using this in a production environment .. viirtualisation is always improving |
| 20:02.11 | *** part/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 20:04.25 | MRH2 | I could see it ok if ur just issuing sip reinvites |
| 20:14.24 | rethus | does the switch-back from agi-script to extension.conf need many performance? |
| 20:18.08 | p3nguin | I have no idea what a switch back is. extensions.conf is where the dial plan executes stuff. |
| 20:18.49 | *** part/#asterisk sidh (~tinom@intellitec2.net) |
| 20:19.06 | rethus | i start my agi-script with AGI(auth.php) |
| 20:19.13 | p3nguin | right |
| 20:19.42 | p3nguin | When the script is done, dial plan execution continues. |
| 20:20.11 | rethus | in auth.php a do some stuff... if the pin and TLN not like needed, i switch back to extension to exten=>auth(enterPin) to request the pin again |
| 20:20.29 | rethus | so it switch between the extension and the script |
| 20:21.34 | p3nguin | It never leaves the extension. |
| 20:24.57 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
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| 20:38.53 | *** join/#asterisk chrissbx (~chrissbx@bas1-montreal07-1176421849.dsl.bell.ca) |
| 20:39.57 | chrissbx | Hi. I'm on a wifi connection with rather high packet losses (around 3% or maybe sometimes more), and people tell me it's audible when I'm using SIP. |
| 20:41.19 | chrissbx | The SIP client I'm using (Twinkle) is not able to send each packet twice in such cases, as sipdroid does. |
| 20:41.37 | chrissbx | This makes me consider again going IAX2. Is there a way to make asterisk or another IAX2 client to do this? |
| 20:42.52 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 20:49.36 | ChannelZ | Hmm I don't think any client retransmits. |
| 20:51.16 | p3nguin | Any idea how to bridge speaker output to an input using alsa? I don't have a male-to-male cable to connect between the jacks, so it needs to be done in software. |
| 20:55.59 | chrissbx | ChannelZ: just in case you're interested, here's how sipdroid does it: http://code.google.com/p/sipdroid/wiki/NewImprovedAudio |
| 20:57.14 | Tim_Toady | great way to deal with packet loss, send everything twice :P |
| 20:58.35 | chrissbx | I'm not sure whether you're being sarcastic; I think it's probably really the best one can do, and at least for the end user bandwidth in the range needed for voip isn't usually a bigger concern than quality. |
| 21:00.44 | ChannelZ | Well I know of no other implementations that do this, IAX or otherwise |
| 21:01.00 | chrissbx | And it seems so simple that I'm kinda tempted to just hack some program to do it; although just *always* sending twice the traffic may not please the service provider; more to the point though it would only fix one direction. |
| 21:04.24 | chrissbx | files it under another issue with his voip setup that's unsolved |
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| 21:34.47 | *** join/#asterisk radic (~radic@tmo-097-108.customers.d1-online.com) |
| 21:47.29 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 21:56.30 | ChannelZ | Falling Skies finale tonight.. woot |
| 22:08.24 | *** join/#asterisk cyborg-one (1000@188-115-162-27.broadband.tenet.odessa.ua) |
| 22:22.56 | ChannelZ | Anyone want on this Google+ mess? |
| 22:23.13 | p3nguin | adds channelz to his circle |
| 22:26.31 | ChannelZ | hmm interesting |
| 22:27.04 | ChannelZ | "aji_client_info_handler: User xxx@gmail.com/TalkGadgetXXXXX does not support discovery." This spit out on my home and work consoles at the same time. |
| 22:28.07 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 22:28.20 | ChannelZ | Ahh. If I hit the 'chat' thing on Google+ it probes me |
| 22:33.16 | *** join/#asterisk datarecall (~data@loxely.illusivecreations.com) |
| 22:47.03 | leifmadsen | jackmcbarn: as stated previously, "<leifmadsen> Additionally, no DTMF would ever be listened for because you're only using Playback(), which does not listen for DTMF -- you need to use Background()" |
| 22:48.52 | ChannelZ | answering questions through a wormhole? :) |
| 22:58.33 | *** join/#asterisk Precognist (~yeshualoo@adsl-75-15-226-185.dsl.bkfd14.sbcglobal.net) |
| 22:59.33 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 23:00.40 | Precognist | Hello Room. |
| 23:01.01 | WIMPy | Hello member. |
| 23:01.24 | Precognist | wow, that NickServ process is something else. I have a problem i was hoping i could get help with. |
| 23:01.43 | p3nguin | ~ask |
| 23:01.43 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 23:02.00 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 23:02.07 | Precognist | nice |
| 23:03.48 | ChannelZ | Are you getting enough oxygen? |
| 23:08.48 | ChannelZ | I'm sensing this is going to be really good. |
| 23:11.33 | p3nguin | haha |
| 23:21.09 | p3nguin | And here I was thinking I was going to get to help someone with a serious problem. Silly me. |
| 23:21.49 | Precognist | not done yet |
| 23:22.05 | WIMPy | Maybe we should leaveUSo you think, we can stop waiting and leave? |
| 23:22.10 | WIMPy | oops |
| 23:23.58 | Precognist | sure. it'll be here when you get back, |
| 23:24.56 | singler | Precognist: if you are writing long line with problem then better pastebin or split, IRC has line limit, so you text may cut |
| 23:25.34 | Precognist | thank you. will need that. |
| 23:26.06 | WIMPy | Or get it published as a book. |
| 23:27.55 | p3nguin | Maybe Colloquy will split the message appropriately, if we're lucky. |
| 23:30.39 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:33.03 | wasanzy | do I need ss7 protocol to be able to build an IVR system? |
| 23:33.21 | WIMPy | Uh-oh |
| 23:33.56 | ChannelZ | Yes! except it's all underneath and you don't need to worry about a single bit of it. |
| 23:34.12 | ChannelZ | So stop |
| 23:34.18 | ChannelZ | Jut be happy |
| 23:35.13 | ChannelZ | s/Jut/Just/ |
| 23:35.40 | wasanzy | I asked before but am back because I am confused |
| 23:35.53 | ChannelZ | The feeling is mutual |
| 23:36.41 | WIMPy | So you think you might be able to get rid of the SS7 obsession now? |
| 23:37.26 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 23:38.38 | ChannelZ | NEVAR! |
| 23:48.07 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 23:50.07 | singler | creating new issue for asterisk svn branch 1.8 what version should I select in Jira? Unreleased 1.8.5.1? |
| 23:51.18 | WIMPy | 1.8 SVN - There's a text filed for the exaclt version below. |
| 23:53.32 | Precognist | might not need it. lots of editing. |
| 23:53.40 | Precognist | ok, where to start... _-SETUP-_ My DSL comes from the wall into a wifi router/gateway, this is then connected to a netgear wifi router(1) (base-station) set to 'repeater'. a second wifi router(2) receives. Both wireless and wired user machines connect to router(2). My asterisk machine and a SPA3000 connects to router(1). _-Machine Info-_ asterisk machine is an old dell xpsm140 (inspiron)w/ Ubuntu 10.10 server, |
| 23:53.40 | Precognist | naked w/ asterisk 1.6. user machines are Macbookpro osx10, win7ultPc, & iphones. each with different softphones/sip installed. _-What i have-_ I have done this ((http://jacolyte.posterous.com/tutorial-how-to-get-asterisk-set-up-and-makin)) and got it working one way. i can call from my mac (softphone) and pickup on iphone (softphone)good, but calls from my iphone(softphone) to my mac(softphone) have no audio on ma |
| 23:53.40 | Precognist | win to mac good audio, mac to win good. _-Question-_ i have some time. i really want to learn this. i don't mind starting from scratch or doing something difficult. starting simple, how do i connect the SPA and asterisk? |
| 23:54.07 | singler | there is SVN, but not 1.8 SVN. I guess SVN is a trunk? |
| 23:54.24 | wasanzy | no one to answer me? |
| 23:54.54 | singler | wasanzy: you do not need ss7 to build ivr |
| 23:55.13 | WIMPy | singler: There are several SVN versions. |
| 23:55.32 | WIMPy | wasanzy: You heard it all before. |
| 23:55.46 | WIMPy | I told you what it's good for. |
| 23:56.17 | p3nguin | precognist: Put it on the same LAN as Asterisk, with an Ethernet cable. Create a SIP entry in asterisk sip.conf for the device. Configure it in the web interface. |
| 23:57.28 | wasanzy | singler: thank u. |
| 23:58.34 | singler | WIMPy: I know that, but on Jira "Affects Version/s" does not have "1.8 SVN", it has only SVN |
| 23:59.32 | WIMPy | singler: What's your issue with that? You put the exact version in the version field. |