| 00:02.03 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 00:17.38 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 00:34.35 | *** join/#asterisk wtf911 (~wtf911@unaffiliated/wtf911) |
| 00:35.08 | wtf911 | i see the difference between an obi110 and obi110 is a voip gateway. i'm guessing if i just want to use it for google voice or to connect to asterisk via SIP then i only need the obi100? |
| 00:39.16 | wtf911 | and assuming that's correct should i go for the obi100 or the grandstream h701? |
| 00:40.10 | mmlj4 | stay away from grandstream, as a general rule |
| 00:40.22 | WIMPy | Not that I know either of them, but if you want opinions you should tell us whet exactely you need first. |
| 00:40.23 | mmlj4 | what the other device is, I have no idea |
| 00:41.53 | wtf911 | im just looking to replace archaic verizon home phone with google voice pretty much |
| 00:42.38 | [TK]D-Fender | etfGoogle voice is about to die... |
| 00:42.43 | [TK]D-Fender | wtf911: Google voice is about to die... |
| 00:42.47 | wtf911 | :o |
| 00:42.58 | wtf911 | you don't think it will be renewed into 2014? |
| 00:43.34 | [TK]D-Fender | wtf911: GV is moving from XMPP to the new Google Hangouts |
| 00:43.52 | [TK]D-Fender | wtf911: The rug is being pulled out from under it and the protocol completely replaced |
| 00:44.04 | [TK]D-Fender | http://www.zdnet.com/google-moves-away-from-the-xmpp-open-messaging-standard-7000015918/ |
| 00:44.21 | [TK]D-Fender | "What independent XMPP server users and IM client users won't be able to do is use Google Hangout's additional services such as VoIP, video-conferencing, or have multiple participants in an IM session. If you want those, you'll need to use Google's Hangouts apps or the Google Web-based Hangouts client." |
| 00:44.29 | [TK]D-Fender | OBI = BAI BAI |
| 00:44.45 | [TK]D-Fender | Not too many people know it was short for "Obi-tuary" |
| 00:45.13 | wtf911 | :/ |
| 00:45.52 | wtf911 | well i have asterisk running with voip.ms as well, so just an adapter which works with that and or google voice for the time being |
| 00:46.05 | wtf911 | would you say the obi100 is a good choice? |
| 00:49.57 | wtf911 | i've been told stay away from grandstream so it's the obi or a cisco? |
| 00:50.22 | [TK]D-Fender | Obi is the only device that all by itself talks to GV. |
| 00:50.28 | [TK]D-Fender | That's if you care about that. |
| 00:50.46 | [TK]D-Fender | Now, ASTERIASK can already talk to GV for as long as that lasts, and all those ATA's speak SIP which * speaks |
| 00:51.28 | wtf911 | so assuming i don't care that obi is the only one to talk to GV itself, would there then be one which is best? |
| 00:51.32 | [TK]D-Fender | ~gs |
| 00:51.32 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 00:51.45 | wtf911 | :D |
| 00:51.57 | [TK]D-Fender | And this Obi stuff looks fly-by-night |
| 00:52.31 | [TK]D-Fender | I've heard they work OK though, but when the only thing that makes them special goes away all you're left with is what'll probably be a second rate SIP ATA |
| 00:53.02 | wtf911 | so cisco > obi? |
| 00:54.53 | [TK]D-Fender | Knowing GV is going away I don't see why I'd choose it. At which point, yes, Cisco.... |
| 00:59.52 | petris | I've never used or owned an Obi thing, but I am currently using asterisk with 4 Cisco phones: SPA 303 and 301 (rebranded Linksys phones), and Cisco 7940/7960 phones (proper Cisco phones). |
| 01:00.46 | petris | The proper cisco phones (7940/60) are nice when you can auto-configure them via TFTP, while the SPA phones are nice when you just want to configure them via http. |
| 01:23.34 | *** join/#asterisk m78pl (18609d57@gateway/web/freenode/ip.24.96.157.87) |
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| 01:23.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:23.58 | m78pl | is version 11 stable ? |
| 01:24.49 | [TK]D-Fender | yes |
| 01:26.30 | m78pl | but wasn't uneven numbers supposed to be experimental ??? like with the kernel/ |
| 01:26.46 | WIMPy | no |
| 01:27.47 | WIMPy | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 01:30.52 | m78pl | i see ok |
| 01:31.07 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.187) |
| 01:31.26 | m78pl | wonder if astersik will be around past 2020 like on that page lol |
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| 01:31.50 | WIMPy | Bets are still taken. |
| 01:32.03 | m78pl | oh yeah what's the payout ? |
| 01:32.46 | WIMPy | 1:0. Yours will be the first entry. |
| 01:33.21 | carrar | it will, but you won't be start it unless the module nsagov.so is loaded |
| 01:33.24 | m78pl | what's the best way nowadays to make some money with asterisk ? |
| 01:33.53 | WIMPy | Replace it with some commercial BS. |
| 01:42.30 | m78pl | is the channel more alive during business hours ? |
| 01:42.50 | WIMPy | yes |
| 01:43.33 | m78pl | haven't been here for years |
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| 02:53.16 | mmlj4 | it's kinda dead the last day or so |
| 02:53.19 | mmlj4 | rather strange |
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| 03:52.31 | malikmasr | Hey, im having problems with configuring a pots line...i can get incoming calls, but i cant get outgoing calls to work...i tested outgoing with my vitelity line and that works fine but pots line fails |
| 03:52.41 | malikmasr | i have an openvox A400P |
| 03:53.13 | malikmasr | does anyone know what may be the cause |
| 03:53.58 | mmlj4 | do you perhaps have the two wires reversed? |
| 03:54.08 | mmlj4 | that'll cause DTMF to fail |
| 03:58.03 | malikmasr | DTMF? Would the inbound work if that was the case? |
| 03:59.58 | mmlj4 | yes |
| 03:59.58 | malikmasr | i have an analog phone into port 1 and a phone line going into port 3 |
| 04:00.34 | malikmasr | which seems to be the proper wiring |
| 04:00.41 | mmlj4 | does your analog phone make calls on the line in question? |
| 04:00.57 | mmlj4 | or maybe I don't know what your setup is |
| 04:01.14 | mmlj4 | what's a port? let's start with that |
| 04:01.33 | malikmasr | ok so the A400P card has 4 ports, two fxs, two fxo |
| 04:01.37 | mmlj4 | ah. |
| 04:01.47 | mmlj4 | never mind then, the wiring isn't bad |
| 04:02.06 | mmlj4 | unless you made up your own cables |
| 04:02.15 | malikmasr | haha no |
| 04:02.21 | malikmasr | the cables seem fine |
| 04:02.29 | mmlj4 | but you could have mentioned a card... |
| 04:02.53 | malikmasr | sorry, that was my mistake |
| 04:03.08 | ChannelZ | You might need to add a delay to your dial |
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| 04:03.51 | ChannelZ | I mean all things being equal, that you've configured the hardware and channels correctly, which if you can get incoming calls seems legitimate. |
| 04:04.10 | malikmasr | I read about that, i tried putting w as the dial prefix for the zap trunk |
| 04:04.16 | malikmasr | but it doesnt seem to have fixed it |
| 04:04.17 | malikmasr | http://pastebin.com/BaBJWYg1 |
| 04:04.23 | malikmasr | thats the call log for outgoing |
| 04:04.41 | ChannelZ | must be FreePBX. |
| 04:04.48 | malikmasr | yes, freepbx |
| 04:05.50 | ChannelZ | well it looks normal, have you used dahdi_monitor to listen to the audio on the line and see what might be going on? |
| 04:06.23 | malikmasr | i have not, i can give that a try |
| 04:06.50 | malikmasr | to be honest, im not sure how that works |
| 04:08.39 | malikmasr | also im quite sure that the issue has to be with the outgoing pots line...i can make outgoing calls using my vitelity sip trunk fine |
| 04:10.08 | ChannelZ | you said you can get incoming on it.. |
| 04:10.25 | malikmasr | yes I can get incoming on the pots line |
| 04:10.27 | malikmasr | but not outgoing |
| 04:10.50 | ChannelZ | So it's reasonably to assume the line works, and perhaps it's something on your telco end. |
| 04:10.51 | malikmasr | ok so i ran dahdi_monitor 3 -v and i can see action on the tx |
| 04:10.53 | malikmasr | nothing on rx |
| 04:11.35 | ChannelZ | record it and listen to it |
| 04:11.59 | malikmasr | So basically i dial out, i can hear a "pickup" on the calling end, and then its dead, and speaking gives tx action nothing on rx |
| 04:12.00 | malikmasr | ok one sec |
| 04:12.03 | malikmasr | let me try recording |
| 04:12.25 | ChannelZ | On the device you're dialing from do you hear anything? Is it dead silence, or can you hear the analog noise floor? |
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| 04:15.54 | malikmasr | i hear a pickup noise, like someone picking up the line, then silence |
| 04:16.02 | malikmasr | and the destination never rings |
| 04:16.07 | malikmasr | theres no ringing at all actually |
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| 04:20.06 | malikmasr | i cant get the recording to play, this is slightly embarassing |
| 04:20.07 | malikmasr | sorry |
| 04:22.23 | ChannelZ | with what |
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| 04:22.55 | malikmasr | its recorded as a raw file |
| 04:23.03 | malikmasr | i tried converting it and playing it |
| 04:23.11 | malikmasr | using the sox command |
| 04:23.18 | malikmasr | and play command respectively |
| 04:24.21 | malikmasr | should i have configured zapta.conf? |
| 04:24.39 | malikmasr | since im using a zap trunk thats dahdi compatible? |
| 04:24.47 | ChannelZ | zap is dead, you're using dahdi now |
| 04:24.53 | malikmasr | oh, ok |
| 04:24.59 | malikmasr | so theres nothing for me to configure |
| 04:25.23 | ChannelZ | well no there still is, although maybe FreePBX did it for you. I don't know, I dont use that thing. |
| 04:25.31 | malikmasr | shoot, this is quite frustrating, i cant think of any reason it wont give outbound calls |
| 04:25.43 | malikmasr | well i configured dahdi manually |
| 04:25.57 | malikmasr | the dahdi_channels.conf file, the system.conf, and the chan_dahdi.conf |
| 04:26.09 | ChannelZ | NORMALLY you have /etc/dahdi/system.conf to configure the hardware, and /etc/asterisk/dahdi-channels.conf to configure the channels |
| 04:26.34 | malikmasr | yes, i have configured them, is there a chance i screwed up on that |
| 04:26.50 | ChannelZ | sorry /etc/asterisk/chan_dahdi.conf which may include dahdi-channels.conf if you used genconf |
| 04:28.23 | malikmasr | hmm, would it help if i posted their contents |
| 04:28.30 | malikmasr | i believe i have configured them properly |
| 04:28.32 | malikmasr | but perhaps not |
| 04:33.08 | ChannelZ | sorry distracted, hang on |
| 04:33.47 | malikmasr | no problem, take your time |
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| 04:34.31 | ChannelZ | run dahdi_monitor 3 -F /tmp/recording.raw and then attempt a call and let it sit there for 20 or 30 seconds at least and then send me that file |
| 04:35.08 | malikmasr | ok one sec |
| 04:39.24 | ChannelZ | W..T..F.. I'm yelling into my mic plugged into my audio interface's channel 1, but the signal light is flashing on channel 2... where there is nothing plugged in.... errr |
| 04:41.11 | malikmasr | how do u want me to send it? |
| 04:41.40 | malikmasr | haha thats bizarre |
| 04:41.42 | ChannelZ | can you ftp? |
| 04:41.50 | malikmasr | sure |
| 04:41.53 | ChannelZ | put it on clients.idolum.com in the incoming directory |
| 04:42.28 | ChannelZ | (power cycled the audio interface and now it's working.. freaking bizarre) |
| 04:42.33 | malikmasr | just uploaded it |
| 04:42.39 | ChannelZ | k sec |
| 04:44.13 | fling | Why when I uncomment `;exten => _+7…´ it has higher priority than `exten => _+73852XXXXXX…´ > http://dpaste.com/1218909/ |
| 04:44.20 | fling | ChannelZ: hello :] |
| 04:45.04 | ChannelZ | ok I don't hear a dialtone on your recording. Have you plugged an analog phone into that line (not your card, but into the jack from the wall)? Does it work? |
| 04:45.35 | malikmasr | yeah if i connect a phone directly to the wall line it works fine |
| 04:45.46 | malikmasr | i dont get a dial tone though when i call through asterisk |
| 04:45.51 | malikmasr | not sure why |
| 04:45.57 | malikmasr | but incoming works on that line |
| 04:46.11 | malikmasr | actually sorry, i get incoming on the line, but you cant hear |
| 04:46.13 | ChannelZ | ok your channel is probably configured with the wrong signalling, pastebin your /etc/dahdi/system.conf |
| 04:46.27 | malikmasr | http://pastebin.com/45LJWn0t thats all the config files |
| 04:46.30 | ChannelZ | wait..what do you mean you can't here? |
| 04:46.32 | ChannelZ | hear? |
| 04:46.53 | malikmasr | for incoming call, it will ring the analog phone, so redirect is fine...but neither side can hear anything |
| 04:47.01 | malikmasr | however, if i use the vitelity line, you can hear perfectly |
| 04:47.09 | malikmasr | and an sip internal call works fine as well |
| 04:47.17 | malikmasr | its just the pots line im struggling with |
| 04:48.12 | ChannelZ | well that's weird. Your config actually looks OK, are you in the US? |
| 04:48.24 | malikmasr | yes |
| 04:48.28 | malikmasr | pennsylvania |
| 04:48.51 | malikmasr | i even have the proper pots forwarded on my router |
| 04:49.02 | ChannelZ | well that doesn't matter for now |
| 04:49.06 | malikmasr | i have no idea what it could be |
| 04:49.22 | ChannelZ | So you have a 400P clone, with a 2-port FXS card and a 2-port FXO card? |
| 04:49.48 | ChannelZ | fling: I don't see what you mean in your config - those two extensions are in different contexts |
| 04:50.08 | ChannelZ | oh nm you included them all into each other |
| 04:50.19 | fling | ChannelZ: there is `include => hh-barnaul ´ in [music] context |
| 04:51.14 | malikmasr | yes |
| 04:51.32 | malikmasr | and it worked before, but my server was physically damaged, so i got a fresh install going |
| 04:51.37 | ChannelZ | anyway you have ambiguous extensions so I'm not surprised it's not doing what you want |
| 04:51.48 | fling | so `exten => _+73852XXXXXX´ only works with `exten => _+7.…´ commented :| |
| 04:51.54 | ChannelZ | Physically damaged how? Lightning? |
| 04:52.06 | ChannelZ | I'm wondering if your card, or at least the FXO card is somewhat wonk |
| 04:53.24 | malikmasr | The card should be fine, i took the drive out because i was doing some changes and a younger sibling got to it while i was away |
| 04:53.26 | fling | ChannelZ: how to fix it properly? I don't want to use stupid ${IF(…)} :P |
| 04:53.37 | malikmasr | kind of destroyed my drive haha |
| 04:54.36 | fling | ChannelZ: do I need to set priority somehow? hmm hmm |
| 04:55.29 | malikmasr | could my trunk config be the issue? |
| 04:56.23 | ChannelZ | malikmasr: well for whatever reason the telco doesn't seem to see the hook state change and give you a dialtone. But kewlstart should pretty much be the standard for US telcos unless you have a special setup for some reason |
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| 04:57.34 | malikmasr | hmm, its the same company i previously used |
| 04:57.48 | malikmasr | not sure what may have changed |
| 04:58.44 | ChannelZ | fling: read this to understand why asterisk matches the extension it does https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 04:58.54 | fling | ChannelZ: thanks |
| 04:58.58 | ChannelZ | malikmasr: who? |
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| 04:59.45 | malikmasr | centurylink i believe |
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| 05:01.23 | ChannelZ | wait a minute, your channels ARE configured backwards.... |
| 05:01.51 | malikmasr | how so? |
| 05:03.01 | ChannelZ | Which channels are which? Are 1 and 2 your FXS (like you'd plug a telephone into) and 3 and 4 your FXO (like you'd plug the line from the wall into)? |
| 05:03.10 | malikmasr | yes i believe so |
| 05:03.18 | malikmasr | my analog phone is in line 1 |
| 05:03.25 | malikmasr | my phone line is line 3 |
| 05:03.31 | malikmasr | i mean channel |
| 05:04.22 | fling | ChannelZ: there are sort rules :] |
| 05:05.35 | malikmasr | is it backwards somehow? |
| 05:05.43 | ChannelZ | ok nevermind, they are right then... assuming you do have it actually right. |
| 05:06.07 | fling | ChannelZ: this is why > http://dpaste.com/1218966/ ; _+7. goes first |
| 05:06.10 | malikmasr | ah, this is frustrating haha, asterisk is confusing |
| 05:07.04 | ChannelZ | well it's confusing.. because an FXO port uses FXS signalling, and an FXS port uses FXO signalling from Asterisk's point of view (because it is acting as the opposite end of the connection in either case) |
| 05:07.37 | ChannelZ | So your FXO ports (lines) are configured as fxsks and your FXS ports (phones) are configured as fxoks |
| 05:07.48 | ChannelZ | what does dahdi_scan say |
| 05:10.34 | malikmasr | http://pastebin.com/AJthU3x0 |
| 05:10.39 | malikmasr | thats the result |
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| 05:13.43 | fling | ChannelZ: I have replaced `_+7.´ with `_+7[0-24-9][0-79][0-46-9][0-13-9].´ ; I don't like it but it works. |
| 05:14.02 | fling | I'll continue reading :P |
| 05:15.58 | ChannelZ | malikmasr: hmm. I'm not sure what's going on. Looking at your audio file again I actually hear a fragment of a dialtone when you hung up. Somehow your telco is missing the signalling or your card has gone wonk. Have you tried channel 4 for the hell of it? (plug the line in there, reconfig to dial out channel 4 instead of 3) |
| 05:16.11 | fling | I'm receiving a lot of junk sip guest calls but they are not saying anything, why? |
| 05:16.25 | malikmasr | ok sweet, ill give that a try |
| 05:16.28 | ChannelZ | What do you mean 'not saying anything' |
| 05:16.38 | malikmasr | thanks so much for your help |
| 05:16.49 | fling | ChannelZ: I say hello and I hear nothing. |
| 05:16.59 | malikmasr | i will come back and let you know how it goes, i have to unplug my computer to get access, the wires are a mess back here haha |
| 05:17.34 | ChannelZ | malikmasr: I'm also troubled by you saying incoming calls work but with no audio. |
| 05:17.40 | fling | they are trying to call 810972595669713, 00972595669713, 810972599166964… |
| 05:17.51 | malikmasr | yeah i have no idea what could cause that |
| 05:17.56 | ChannelZ | fling: and why do you accept those from unauthenticated peers? |
| 05:17.57 | malikmasr | i thought the issues would be linked |
| 05:18.13 | fling | ChannelZ: I want to know who are they :P |
| 05:18.19 | ChannelZ | fling: they are bots anyway, fishing to see if your system in unsecured and if they can make calls through you |
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| 05:18.53 | ChannelZ | malikmasr: I'm still not convinced your ports are actually right.. like maybe you have the physical ports on the card confused with each other |
| 05:19.00 | fling | ChannelZ: ok, I will let bots do some calls… |
| 05:19.12 | malikmasr | i will take out the cards and double check |
| 05:20.34 | malikmasr | brb |
| 05:20.45 | ChannelZ | I've never tried reversing my cables (and I'm not where my hardware is right now) to know how it behaves to say one way or the other, if it half-works like you're getting, |
| 05:20.48 | ChannelZ | oops |
| 05:20.51 | ChannelZ | too slow |
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| 06:04.15 | fling | ChannelZ: it is all now working :P |
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| 07:48.07 | ChannelZ | glad to hear it |
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| 08:55.21 | fling | ChannelZ: == Spawn extension (fling, 22306, 1) exited non-zero on 'SIP/000-000001d8' |
| 08:55.43 | fling | I had GoGub instead of GoSub :D |
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| 11:27.13 | fling | ChannelZ: pattern sorting rules are not working with included contexts :| |
| 11:27.25 | fling | Included pattern is always sorted after hmm hmm |
| 11:45.12 | WIMPy | That's the idea. |
| 11:45.22 | fling | WIMPy: umm? |
| 11:45.39 | WIMPy | It's the only way YOU can sort them. |
| 11:46.11 | fling | WIMPy: I sort them putting into included concexts :P |
| 11:46.56 | WIMPy | exactely |
| 12:13.57 | fling | WIMPy: should not I use skypiax? |
| 12:14.17 | WIMPy | What is that? |
| 12:18.04 | fling | module chan_skypiax.so for skype |
| 12:18.44 | WIMPy | But only for Freeswitch. |
| 12:27.47 | fling | Do I need siptosis? |
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| 14:56.55 | onixx | hello all. I am using 390991 and experience a severe problem. issuing "reload" on CLI will make asterisk exit 90% of the time |
| 14:57.42 | onixx | I am wondering how I can determine what is causing this. looking in log files does not reveal much |
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| 14:58.54 | WIMPy | ~collectdebug |
| 14:58.55 | infobot | collectdebug is, like, a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 14:59.14 | WIMPy | First enable core dumps. |
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| 15:00.40 | onixx | thanks WIMPy; also , any clues what this mean... it just started since I updated from svn : xmldoc_update_config_option: XML Documentation for option '^.*$' in modules 'features' not found! |
| 15:01.06 | onixx | I get about 20 of these when loading |
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| 15:05.57 | igcewieling1 | onixx: Have you considered using a released version of Asterisk? |
| 15:06.29 | onixx | igcewieling1: yes but there's some changes I participated in jira that I need ;-) |
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| 15:10.03 | WIMPy | onixx: Make sure you don't have anyhting parking related in your features.conf. |
| 15:10.23 | WIMPy | If in doubt, use a blank one. |
| 15:10.46 | onixx | WIMPy: yes, removed those... was not starting at all before |
| 15:11.09 | onixx | WIMPy: I guess the documentation warnings are normal for now ? |
| 15:12.06 | onixx | all I have in features.conf is featuremap section with automon |
| 15:12.08 | WIMPy | I dind't see them. |
| 15:13.19 | onixx | this is the exact error: config_options.c:982 xmldoc_update_config_option: XML Documentation for option 'pickupsound' in modules 'features' not found! |
| 15:13.41 | WIMPy | Right. I see them. All missing documentation for features keywords. |
| 15:14.16 | onixx | ha !! I fell better now |
| 15:15.02 | onixx | just need to figure the crash on reload. |
| 15:15.36 | onixx | I did it on a june 1st trunk too so I suspect its something with my configs |
| 15:15.38 | WIMPy | never uses reload, but it seems to work for me. |
| 15:16.53 | onixx | I do it by habit... not even sure what it exactly reloads ! |
| 15:17.12 | onixx | (must be reloading pretty much everything) |
| 15:17.24 | WIMPy | At least that's the idea. |
| 15:19.04 | onixx | funny, it works as much as I want it to work when starting ! I can reload 20 times in a row if I want |
| 15:19.34 | onixx | it seems to crash if I let it run for a 10-15 minutes then reload |
| 15:20.05 | WIMPy | How fast? |
| 15:20.27 | WIMPy | IIRC it won't reload if the previour (re)load hasn't completed, yet. |
| 15:20.34 | WIMPy | previous |
| 15:20.43 | onixx | no... sorry, that's not what I meant... |
| 15:21.12 | onixx | If i start asterisk and reload... I can do it all I want and it does not make asterisk exit |
| 15:21.26 | WIMPy | But it DOES reload? |
| 15:21.36 | onixx | yes it reloadas |
| 15:21.43 | WIMPy | Sounds rather evil. |
| 15:22.10 | fling | WIMPy: is it bad to use google voice? |
| 15:22.30 | onixx | if I do nothing for 10-15 minutes... then type reload on cli. it may crash and exit immediately |
| 15:23.01 | WIMPy | fling: Depends on what you think is bad. You know that google is not evil. Or so they say. |
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| 15:23.10 | onixx | I have debug on now... letting it sit for a few minutes now |
| 15:24.02 | onixx | WIMPy: is [featuremap] staying in features.conf ? |
| 15:24.02 | fling | WIMPy: I want to call _+1. for free :P |
| 15:24.55 | WIMPy | onixx: I have no idea, what's going on there, but I think it's only the parking functions that are being revamped. |
| 15:25.31 | WIMPy | fling: There's no such thing as a free lunch. But I heard it's a possibility to not pay per minute. |
| 15:27.05 | fling | WIMPy: so is it good for free calls? Or do I need another service? |
| 15:27.50 | WIMPy | You know that it probably won't last for long. |
| 15:28.17 | fling | hmm hmm |
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| 15:31.51 | fling | Are not you using it? |
| 15:31.57 | WIMPy | no |
| 15:32.03 | igcewieling1 | If (your Asterisk actually crashes and you are running the latest version in your branch) then { file a bug report on jira }; |
| 15:32.42 | igcewieling1 | no sane person uses free SIP calling services. |
| 15:38.44 | WIMPy | No sane person uses SIP calling services. |
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| 15:41.18 | newtonr | drmessano: I see you! |
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| 16:01.51 | onixx | WIMPy: I believe it was my features.conf automon that caused problems ! |
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| 18:13.21 | finalius | hi, i have a (i think) simple question about dialplans with a voip provider |
| 18:16.17 | ChannelZ | ..and then.. |
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| 18:19.39 | finalius | ok, basically i have a couple of extensions, all in one context, all sip phones, and a VOIP provider. what i'd like to do is that any number which does not match a defined extension should be routed to the voip provider and an incomming call from the provider should ring on all extensions |
| 18:20.36 | finalius | i've registered with the provider via register => in the sip.conf, and i can route an incoming call to one extension |
| 18:20.40 | finalius | but outgoing calls don't work |
| 18:21.08 | [TK]D-Fender | ~pb |
| 18:21.08 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude. |
| 18:21.10 | [TK]D-Fender | ^^ |
| 18:21.12 | ChannelZ | Well you should separate your 'incoming' calls into their own context and one for your 'outgoing' - otherwise you've created a security hole whereby people calling in could potentially call back out. |
| 18:21.17 | [TK]D-Fender | Show use what you've made in the dialplan |
| 18:21.19 | [TK]D-Fender | us* |
| 18:21.31 | ChannelZ | So what you're doing is a bit of a bad idea to start with. |
| 18:21.33 | [TK]D-Fender | And incoming and outgoing are 2 separate things tolook at |
| 18:24.33 | [TK]D-Fender | finalius: And show us the failed call with "core set verbose 10", "sip set debug on" |
| 18:25.14 | finalius | ok, give me a second, there's a lot of passwords and phone numbers in those files :) |
| 18:25.39 | [TK]D-Fender | We're anly talking about extensions.conf .... and you shouldn't have any in there at all |
| 18:25.53 | [TK]D-Fender | If you do you're probably doing things wrong... |
| 18:27.20 | finalius | ah ok |
| 18:27.31 | finalius | well, this is sip.conf as well, it's all rather small |
| 18:27.34 | finalius | http://pastebin.com/JU3GwDz1 |
| 18:27.47 | finalius | i've left the phone number and voip-provider in it, feel free to call me :p (it's a test number) |
| 18:28.10 | igcewieling1 | putting your extensions in the same context as your outbound and inbound calling extensions is what cause people to get fired because their PBX was broken into. |
| 18:28.23 | igcewieling1 | You might want to polish up your resume |
| 18:28.23 | [TK]D-Fender | finalius: You are missing a LOT of SIP settings for Asterisk to work from behind NAT |
| 18:28.28 | [TK]D-Fender | This is jsut for starters |
| 18:29.14 | igcewieling1 | [TK]D-Fender: maybe he didn't read the Asterisk Book? |
| 18:29.22 | finalius | i read about that |
| 18:29.29 | [TK]D-Fender | finalius: And please show us the failed call..... |
| 18:30.00 | ChannelZ | Dial(SIP/guest-voip/${EXTEN:1}) because won't SIP/123@guest-voip consider 'guest-voip' a domain name (and fail)? |
| 18:30.37 | [TK]D-Fender | ChannelZ: It should use the peer.... |
| 18:30.58 | ChannelZ | ok. I rarely use that syntax |
| 18:31.45 | ChannelZ | But yes, stabbing in the dark without seeing the actual call failure |
| 18:32.27 | ChannelZ | wanders off to get a haircut |
| 18:33.04 | finalius | http://pastebin.com/PJZxKrr3 |
| 18:33.15 | finalius | this is the log when trying to call my cellphone |
| 18:34.08 | igcewieling1 | finalius: try it with a real softphone instead of the weird webrtc thingy which nobody can get working anyway |
| 18:34.41 | ChannelZ | Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 18:34.52 | finalius | i'm using x-lite |
| 18:34.52 | igcewieling1 | then pastebin the new failed call masking NOTHING except passwords in sip.conf. |
| 18:35.03 | igcewieling1 | finalius: xlite supports to Opus codec? |
| 18:35.12 | igcewieling1 | set your codec to ulaw and try it again |
| 18:35.15 | ChannelZ | your guest-voip has no hostname or IP. |
| 18:35.30 | ChannelZ | it has no idea how to contact them |
| 18:35.41 | igcewieling1 | finalius: it is starting to look like if you do the opposite of everything you have done it will start working. 8-| |
| 18:36.21 | finalius | okay okay okay, one at a time :) |
| 18:36.34 | igcewieling1 | a=rtpmap:125 opus/48000/2 NOT SUPPORTED. |
| 18:36.40 | finalius | my guest-voip has fromdomain=guest1.voipgateway.org |
| 18:36.41 | igcewieling1 | a=rtpmap:100 speex/16000 only if you built support for it in Asterisk |
| 18:36.43 | [TK]D-Fender | finalius: [guest-voip] has no HOST field <------------------- |
| 18:36.45 | [TK]D-Fender | ^^^^^^^^^^^ |
| 18:36.56 | ChannelZ | fromdomain is something different. |
| 18:36.56 | igcewieling1 | finalius: put the correct host= in the peer in sip.conf |
| 18:37.42 | [TK]D-Fender | finalius: And change this : exten => _9.,1,Dial(SIP/${EXTEN:1}@guest-voip) |
| 18:37.55 | [TK]D-Fender | finalius: To this : exten => _9.,1,Dial(SIP/guest-voip/${EXTEN:1}) |
| 18:38.08 | ChannelZ | dejavu |
| 18:38.10 | ChannelZ | :P |
| 18:38.52 | ChannelZ | ok I'm really leaving now. Have fun gents. |
| 18:38.57 | finalius | thanks ! |
| 18:39.05 | finalius | ok with the host= stuff it works |
| 18:39.12 | finalius | and i'll use the other syntax |
| 18:39.22 | finalius | now, final question (for now :) ) |
| 18:39.25 | [TK]D-Fender | finalius: It is required. FROMDOMAIN is rarely required. |
| 18:39.51 | finalius | [TK]D-Fender: I copied this from a tutorial :( |
| 18:40.05 | finalius | anyhow, last question: can I use a "catch all if no extension is matched" instead of _9. ? |
| 18:40.20 | finalius | so basically any number that is not 500 (with current config) should go to guest-voip ? |
| 18:42.01 | igcewieling1 | finalius: you might want to read the book again |
| 18:43.00 | finalius | igcewieling1: i'd love to - is there a link? |
| 18:43.07 | igcewieling1 | ~book |
| 18:43.08 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:43.08 | finalius | because, seriously, the wiki-page just sucks |
| 18:43.51 | finalius | the wiki might be helpful for people who know their way around, but the "beginner's configuration" is about as helpful as a blank page |
| 18:43.53 | igcewieling1 | the voip-info is good if you want a mismash of docs for asterisk 1.2, 1.4, 1,6, 1.8, and Asterisk 11 all mixed together. |
| 18:44.12 | igcewieling1 | finalius: telecom and PBXs are complicated. |
| 18:45.52 | finalius | igcewieling1: agreed if you have a bit setup |
| 18:46.02 | finalius | all i'm trying to do is making a simple phone call :) |
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| 18:46.25 | igcewieling1 | finalius: There is no such thing as a "quick setup of a PBX to make a simple phone call" |
| 18:46.43 | igcewieling1 | Makes about as much sense as thinking there can be a "simple amputation of a limb" |
| 18:53.33 | igcewieling1 | If you want a different kind of complicated, you could try FreePBX (supported on #FreePBX, NOT #asterisk) |
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