| 00:00.15 | ChannelZ | Ah. |
| 00:00.17 | elvisthedj | announcer presses 1.. audience hears 1 minute commerical.. announcer hears "rejoinging broadcast in 30 seconds.. 10 seconds.." etc |
| 00:00.31 | elvisthedj | demo of the system is tomorrow.. so i was sweating bullets |
| 00:00.32 | jkroon | sruffel is not going to like this one bit. the commit that broke it looks as innocent as anything. just confirming ... but I can't imagine that this can be correct. |
| 00:02.12 | elvisthedj | btw, i do voiceovers for lots of radio/tv stations. if you ever need any pro audio for your system, msg me and i'll help you out |
| 00:02.14 | ChannelZ | Well once you get past this you should look at Asterisk 10 and ConfBridge, it looks like it would make that sort of thing even a whole lot easier. (I don't remember how advanced ConfBridge was in 1.8, and I think it was as basic as it gets in 1.6 IIRC)) |
| 00:02.35 | ChannelZ | Do you have a VO demo? I work in video post |
| 00:03.30 | jkroon | haha, i do asterisk for 70 % of my living :p |
| 00:03.30 | elvisthedj | i do. mostly radio, so its going to have that.. radio sound. but i do lots of ebooks too |
| 00:04.02 | ChannelZ | That's fine. Non-union? |
| 00:04.27 | elvisthedj | ChannelZ: http://www.voices.com/talents/search?conditions%5Bkeyword%5D=kris+edwards |
| 00:04.32 | elvisthedj | non-union |
| 00:04.37 | elvisthedj | for now :) |
| 00:05.37 | WIMPy | It was very basic. |
| 00:06.01 | ChannelZ | cool will bookmark. Occasionally a client will ask my opinion :) |
| 00:07.39 | elvisthedj | ChannelZ: If it happens, send me a message and tell me that you're from this channel and I'll do my best to make you a zero dollar deal :) |
| 00:07.43 | elvisthedj | finally get to go home now |
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| 00:08.12 | ChannelZ | Yay! |
| 00:12.34 | *** join/#asterisk luisgrin (bec0c24a@gateway/web/freenode/ip.190.192.194.74) |
| 00:13.30 | luisgrin | hi, im trying to build a ivr using asterisk and google asr, my question is if it is possible to get the string that google asr "hears" |
| 00:20.48 | navaismo | luisgrin, yes ist possible in fact the example echo print that string |
| 00:21.27 | jkroon | does anyone happen to have an email address for sruffel? |
| 00:22.37 | navaismo | luisgrin, check the examples |
| 00:22.54 | luisgrin | navaismo: thank you, actually im a "client" im lookin for a ivr developement and i saw today a DEMO in which the pc echoed what i say in his own voice, so i gues it is possible to get the string writen |
| 00:23.14 | luisgrin | navaismo examples were? |
| 00:23.43 | navaismo | http://zaf.github.com/asterisk-speech-recog/ |
| 00:23.50 | luisgrin | ok, thanks a lot |
| 00:25.27 | navaismo | feels like he ruined a sale |
| 00:27.49 | luisgrin | dont feel so, i have too much work to do :) |
| 00:28.07 | luisgrin | but i want to know more in order to ask for more |
| 00:28.08 | navaismo | anyway Google ASR is not a recommended solution for production, 1st) google can send a bill for violating their terms 2) The ASR is not accurated |
| 00:28.33 | navaismo | 3) google in any momoent can put down that service |
| 00:28.35 | luisgrin | navaismo, then what asr do you recomend? |
| 00:29.30 | navaismo | If you want professional asr take a look on lumenvox or |
| 00:29.44 | navaismo | something else I think there a spanish company with asr for asterisk too |
| 00:30.01 | luisgrin | actually im it responsable and im trying to build an ivr for medical squedulling --first some prototype-- |
| 00:30.34 | luisgrin | so, i think google asr maybe enough for the demo |
| 00:30.40 | navaismo | Verbio |
| 00:31.35 | navaismo | My suggestion is: dont use volatile tools for production demos , use a professional tool for that |
| 00:31.45 | navaismo | http://www.voip-info.org/wiki/view/Voice+recognition |
| 00:31.47 | luisgrin | yes actually the person who show me the demo told me about this product and told me he think it is kind of 500u$s each line ... |
| 00:33.23 | navaismo | yep develop time and licensing make that cost |
| 00:34.12 | luisgrin | i know :) .... anyway i must say that asterisk and google asr api were very impresive for me today |
| 00:35.03 | jkroon | rofl, ok, so it can't figure out en-ZA in any reliable way ... getting some really, really funny crap back though. |
| 00:45.35 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 00:45.35 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 00:48.15 | EC_Joe | hi |
| 00:50.20 | sruffell | jkroon: darn it…that was the commit I feared. 2.6.1 already had another change to fix an error in that commit. |
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| 01:22.50 | jkroon | sruffell, don't think you got my last msg's, anything I can do to assist? |
| 01:24.21 | sruffell | Yes….I'll prepare a few patches and put them on a branch on the git repo. I bet it is related to always using the full-wave ring detect mode now, but I'm not sure in your case whether we aren't crossing the ring threshold..or if the driver isn't detecting enough polarity transitiions. |
| 01:24.56 | jkroon | ok, guess we'll tackle that tomorrow evening then. |
| 01:24.59 | sruffell | But I know that the ring threshold is changed when opermode is set to SOUTHAFRICA, so I'm going through those settings right now. Not sure how long it will take me to get a patch. |
| 01:25.07 | sruffell | ok…cool…thanks, and sorry about this :/ |
| 01:25.19 | jkroon | you've got my email address, so please let me know what/where. |
| 01:25.34 | jkroon | sruffell, bugs happen, it's a part of life. |
| 01:25.38 | sruffell | will do…I'll post on the JIRA issue you opened up. |
| 01:25.38 | jkroon | thanks for fixing :) |
| 01:25.46 | jkroon | perfect. |
| 01:25.56 | sruffell | got to love git bisect though, eh? |
| 01:26.00 | sruffell | I know I do…. |
| 01:26.00 | jkroon | anyway, i need some sleep now ... or I'm not going to make the day tomorrow. |
| 01:26.04 | jkroon | absolutely. |
| 01:26.52 | jkroon | particularly useful on larger projects, and in particular if you're not the only person committing code. |
| 01:27.46 | jkroon | anyway, thanks for your assistance, chat again. |
| 01:27.50 | sruffell | ok..ttyl |
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| 01:33.39 | limpc | anyone happen to know of a great company located in Austin, TX that provides SIP trunking? |
| 01:34.48 | kuruption | it has to be in austin? |
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| 02:03.17 | limpc | kuruption, yeah or nearby |
| 02:03.26 | limpc | im looking for something like voxlinenetworks.com |
| 02:03.59 | limpc | have metal with asterisk, so need colocation and prefer somewhere nearby that I can drive to if any problems with the system |
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| 02:23.32 | limpc | is a dual 4-core 2.4ghz Xeon (E5620) + 6gb ram enough to handle 240 concurrent SIP sessions? |
| 02:24.02 | limpc | dont think theres any transcoding |
| 02:31.04 | limpc | is the 30mhz per channel still an accurate assumption? |
| 02:34.50 | sruffell | I never even heard the 30mhz / channel assumption before. |
| 02:36.10 | sruffell | I honestly doubt you will have problems with 240 non-transcoded sip sessions that are just passing through (no recording, no inline dtmf, etc..) but nobody will ever say "yeah, that's fine…" since there are too many variables. |
| 02:36.42 | limpc | :) |
| 02:37.16 | limpc | well i dont remember where i saw the number but it was back in 2001 when I was doing fun stuff with asterisk. |
| 02:38.31 | sruffell | I'll assume you're not running your dial-plan via AGI that is loaded in off a tape drive? |
| 02:38.41 | sruffell | :) |
| 02:38.48 | limpc | -_- its not THAT long ago |
| 02:38.53 | sruffell | oh, better yet..a casette |
| 02:39.15 | limpc | naw we use punch cards, its the fastest |
| 02:39.24 | sruffell | nods knowingly |
| 02:40.20 | sruffell | Telephony and punch cards makes me think of this gem I first saw linked by leifmadsen: http://www.youtube.com/watch?v=uKfKtXYLG78 |
| 02:41.26 | limpc | wow. we acted like that back then? |
| 02:53.23 | leifmadsen | sruffell: lol nice |
| 02:53.46 | ChannelZ | Something has gone wrong. An error has occurred. |
| 02:55.51 | leifmadsen | sruffell: I love the little overscan bar at the bottom :) |
| 02:56.12 | leifmadsen | sruffell: asterisk should be using erlang. obviously. |
| 02:56.16 | sruffell | heh…that's one of the funnier tech videos. Especially since it wasn't made to be funny (or so I think....) |
| 02:56.30 | leifmadsen | ya I think it was serious :) |
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| 04:18.34 | eugene_myit | logout |
| 04:18.36 | eugene_myit | quit |
| 04:18.38 | eugene_myit | exit |
| 04:18.45 | ChannelZ | /quit |
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| 04:36.26 | dlynes_laptop | Was there a bug with double acks on SIP invites on 1.6.2 branch where you can make an unauthenticated call? |
| 04:38.42 | ChannelZ | ? |
| 04:42.11 | dlynes_laptop | I see a caller sending a SIP invite, then it gets sent two SIP unauthorizeds (no idea why there's two), then it sends two acks, then it sends another invite, and it's good |
| 04:42.50 | dlynes_laptop | This is for a caller that shouldn't be getting authenticated, because they shouldn't be able to guess a 25 character random SIP password |
| 04:43.09 | dlynes_laptop | especially not in one try |
| 04:51.36 | ChannelZ | post the sip debug |
| 04:54.55 | ChannelZ | bbl workout |
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| 06:23.29 | schmidts | good morning |
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| 06:25.51 | bulkorok | hi |
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| 06:52.48 | Maxus2 | Hi Asterisk people, i have a quick one: |
| 06:52.52 | Maxus2 | i have this: NOOP(exten=${EXTEN}&systemname=${SYSTEMNAME}) |
| 06:53.17 | Maxus2 | but i noticed everything after & is not displayed, is there a way to escape the & character? |
| 06:56.24 | kaldemar | Maxus2: \ |
| 06:56.49 | kaldemar | i guess you're using some old version of asterisk. am i right? |
| 06:57.28 | Maxus2 | 1.8 |
| 06:57.34 | Maxus2 | slash doesn't seem to work |
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| 06:59.31 | Maxus2 | its wierd i end up getting: NoOp("SIP/R-1000005-0006-0000001f", "exten=0410488007") in new stack |
| 07:00.07 | Maxus2 | when i do: NOOP(exten=${EXTEN}&systemname=${SYSTEMNAME}) |
| 07:00.40 | Maxus2 | oh never mind i think i know why |
| 07:00.48 | Maxus2 | it because of curl realtime |
| 07:00.51 | Maxus2 | its |
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| 07:05.02 | Maxus2 | thanks for the help :) |
| 07:17.18 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 07:17.22 | v0lZy | hello |
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| 07:26.24 | diab | Matin v0lZy :) |
| 07:26.53 | v0lZy | Matin? |
| 07:28.12 | diab | Morning :) |
| 07:29.35 | v0lZy | in what language is that? |
| 07:29.55 | diab | Fr |
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| 07:56.46 | bartbb | someone here who can help me getting elastix with a2billing working? :/ |
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| 08:11.51 | wdoekes | ~elastix |
| 08:11.51 | infobot | extra, extra, read all about it, elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
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| 08:14.48 | bartbb | wdoekes thanks |
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| 08:15.35 | youjelly | Hey guys,would anyone recommend voxbone over teliax or voicepulse? |
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| 08:58.17 | v0lZy | anyone here using asterisk 10 and higher? |
| 08:58.28 | v0lZy | (or have compelling reasons not to use it) |
| 09:03.08 | v0lZy | I'm having some issues transfering... |
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| 09:57.02 | WIMPy | v0lZy: higher |
| 09:57.32 | v0lZy | WIMPy <3 |
| 09:58.31 | v0lZy | WIMPy: I'm using 10.5.1 |
| 09:59.15 | v0lZy | and im having someissues ... when i transfer calls, the transferee doesnt hear ringing (attended or unattended transfer) of the phone hes being transfered to |
| 10:00.10 | WIMPy | didn't notice such a thing so far. |
| 10:01.52 | v0lZy | WIMPy: i think it might be cause i havent configured almost anything else but sip.conf and extensions.conf |
| 10:02.19 | v0lZy | that... or i should be using answer() or something somewhere |
| 10:03.07 | WIMPy | Answer on a transfer? |
| 10:03.12 | v0lZy | this is my dialplan |
| 10:03.14 | v0lZy | http://bpaste.net/show/G4J624I71gi8dACZS0l3/ |
| 10:04.18 | v0lZy | its pretty simple... sip.conf stuff points to [SIPXX] contexts, those include [DialPatterns] which gosub to [AppInternalCalls].... that one checks for call forwarding stuff, and then at the end of it, it does a Dial... |
| 10:04.30 | v0lZy | i get music on hold |
| 10:05.02 | v0lZy | but as soon as the person that is transfering me puts down the phone in attended transfer... i get silence until the person im transfered to picks up. Audio then works both ways |
| 10:05.32 | v0lZy | i tried checking channels too... and when the person im transfered to hangsup... i still have an active line... no busy or call terminated or anything |
| 10:05.41 | v0lZy | so im either missing something in the dialplan... or other configuration files |
| 10:06.19 | WIMPy | Ok, let me check with a sip phone... |
| 10:08.18 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 10:09.00 | WIMPy | Mope I get ringing. |
| 10:09.28 | v0lZy | so my dialplan is ok. |
| 10:09.37 | v0lZy | what else would be controling this? |
| 10:09.42 | v0lZy | music on hold? |
| 10:09.47 | v0lZy | features.conf? |
| 10:10.05 | WIMPy | Chan_sip or even your phone. |
| 10:10.29 | WIMPy | But I don't think it just tells the phone to play sounds. |
| 10:10.39 | WIMPy | So it must be in the sip channel. |
| 10:13.32 | kaldemar | indications.conf should also have valid configs. |
| 10:15.48 | v0lZy | sip channel... lets see |
| 10:17.35 | WIMPy | Fully working transfers are the one biggest isse that keeps me from using Asterik in offices. |
| 10:17.52 | v0lZy | sim channel... |
| 10:17.54 | v0lZy | i dont have that |
| 10:18.02 | v0lZy | sip.conf or sip_notify.conf |
| 10:20.55 | v0lZy | ah |
| 10:21.02 | v0lZy | indications.conf fixes this ringing thing |
| 10:21.04 | v0lZy | thanks |
| 10:22.08 | v0lZy | another thing though |
| 10:22.11 | v0lZy | when the user hangsup |
| 10:22.18 | v0lZy | the person i've been transfered to |
| 10:22.22 | v0lZy | the original caller still has a line |
| 10:22.30 | v0lZy | ... |
| 10:22.36 | v0lZy | thats rather strange, aint it? |
| 10:23.25 | WIMPy | What does "has a line" mean? |
| 10:24.31 | v0lZy | its still active |
| 10:24.35 | v0lZy | channel is up |
| 10:24.37 | v0lZy | silence though |
| 10:25.02 | WIMPy | Which channel? |
| 10:25.29 | WIMPy | I see 3 channels in a transfer. 2 must stay up and the other one is hung up by the user. |
| 10:27.06 | v0lZy | yeah but when the person that was transfered to hangs up |
| 10:27.16 | kaldemar | v0lZy: pastebin CLI output of a call. guessing is futile. |
| 10:28.13 | WIMPy | Not sure I get it right. After the transfer is completed and the (2nd) conversation is over, and person C hangs up, A stays active? |
| 10:28.18 | v0lZy | SHINOBI*CLI> core show channels |
| 10:28.20 | v0lZy | Channel Location State Application(Data) |
| 10:28.21 | v0lZy | SIP/16-00000000 call@AppInternalCall Up Dial(SIP/50,,Tt) |
| 10:28.23 | v0lZy | SIP/50-00000003 (None) Up AppDial((Outgoing Line)) |
| 10:28.30 | v0lZy | even though 50 hung up |
| 10:28.34 | youjelly | Would anyone recommend voxbone over teliax or voicepulse? |
| 10:28.47 | youjelly | anyone tried voxbone |
| 10:28.49 | v0lZy | WIMPy: exactly |
| 10:29.16 | v0lZy | A calls B, B transfers to C, B hangsup, A and C have a conversation, C hangsup, A's channel is still online. |
| 10:29.18 | WIMPy | If you still see Dial running, that doesn't look like the cllee hung up. |
| 10:31.02 | v0lZy | maybe its just this softphone |
| 10:31.10 | v0lZy | its disconnecting after 8 seconds |
| 10:31.17 | v0lZy | for no apparent reason. |
| 10:31.23 | v0lZy | nothing in its debug |
| 10:31.35 | v0lZy | noting in verbose 9 with asterisk |
| 10:31.37 | v0lZy | maybe debug.. |
| 10:32.17 | v0lZy | nothing in debug 9 either |
| 10:32.39 | WIMPy | Look at sip debug. |
| 10:34.47 | *** join/#asterisk gurra (~gurra__@unaffiliated/gurra) |
| 10:36.24 | v0lZy | erm.. whats the command? |
| 10:36.29 | v0lZy | set sip debug doesnt do anything |
| 10:37.04 | v0lZy | never mind |
| 10:37.05 | WIMPy | sip set debug on |
| 10:37.06 | v0lZy | got it |
| 10:40.04 | v0lZy | http://bpaste.net/ |
| 10:40.22 | v0lZy | erm |
| 10:41.12 | v0lZy | http://pastebin.com/eFjWgsxn |
| 10:41.30 | v0lZy | i turned it off right after it dropped |
| 10:42.13 | v0lZy | REGISTER sip:192.168.1.6:50 SIP/2.0 |
| 10:42.17 | v0lZy | this is weird though... |
| 10:42.48 | v0lZy | but not part of this call i think |
| 10:42.54 | v0lZy | not weird.. have it configured, forgot |
| 10:43.37 | WIMPy | The paste doesn;t mean much without the timing. |
| 10:43.53 | WIMPy | Compare when you hang up and when you see a BYE in the debug. |
| 10:44.48 | v0lZy | X-Asterisk-HangupCause: Protocol error, unspecified |
| 10:44.50 | v0lZy | X-Asterisk-HangupCauseCode: 111 |
| 10:45.04 | v0lZy | sip/50 thingy... |
| 10:45.34 | WIMPy | Looks like something went wrong at SIP level. |
| 10:46.25 | v0lZy | i only get this with this softphone |
| 10:47.35 | v0lZy | Scheduling destruction of SIP dialog '450a6e090b7b46b57e1b688337dd34ac@192.168.2.249:5060' in 6400 ms |
| 10:47.39 | v0lZy | what abot this? |
| 10:47.48 | v0lZy | 7 seconds is what it showed |
| 10:48.00 | v0lZy | sounds pretty close to 6400ms |
| 10:48.09 | v0lZy | why does it schedule a destruction? |
| 10:48.42 | v0lZy | ah |
| 10:48.44 | v0lZy | NAT maybe.. |
| 10:49.31 | v0lZy | phones are on 192.168.1.* |
| 10:49.37 | v0lZy | softphone is on 192.168.2.* |
| 10:49.49 | v0lZy | * is on 192.168.2.249 |
| 10:52.05 | v0lZy | i think ill deal with that NAT thing later... |
| 10:52.35 | v0lZy | but as its on topic |
| 10:52.39 | v0lZy | nat=never and nat=yes |
| 10:52.55 | v0lZy | this is to say theres NAT between that device and asteriskpbx, right? |
| 10:56.17 | v0lZy | hm yeah nat |
| 10:56.25 | v0lZy | oneway audia and the whole shebang. |
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| 11:01.16 | schmidts | gangbang what? |
| 11:01.33 | schmidts | ah sorry i should read correctly ;) |
| 11:01.54 | *** join/#asterisk eicto (~eicto@144-71.dsl.aichyna.com) |
| 11:03.19 | v0lZy | :D |
| 11:03.37 | v0lZy | !# :D |
| 11:03.50 | v0lZy | #! even.. |
| 11:04.18 | v0lZy | nat gangbang.. ehehe :D |
| 11:04.41 | v0lZy | probably not far from the truth if u take into account how much NATting routers there could be between two phones |
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| 11:25.33 | v0lZy | WIMPy, schmidts: i want to implement hints |
| 11:25.36 | v0lZy | but im using gosub |
| 11:26.05 | WIMPy | Where is the relation? |
| 11:26.11 | v0lZy | i was under the impression that i have to put exten => hint stuff before priority 1 |
| 11:26.21 | v0lZy | WIMPy: well its not working :D |
| 11:26.41 | WIMPy | Yes. |
| 11:26.56 | WIMPy | But it doesn't matter where you put them. |
| 11:27.35 | v0lZy | as in.. whatever context? |
| 11:27.39 | v0lZy | i can have a separate context? |
| 11:27.47 | v0lZy | like |
| 11:27.50 | v0lZy | [BLF] |
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| 11:28.00 | v0lZy | exten => 44,hint,SIP/44 ? |
| 11:28.34 | v0lZy | or does this have to be in the start context? |
| 11:28.53 | WIMPy | Yes, you can have them in an extra context. |
| 11:29.05 | WIMPy | You need to specify that in sip.conf anyway. |
| 11:30.14 | v0lZy | I see |
| 11:30.24 | v0lZy | whats the best way to do it? |
| 11:30.37 | v0lZy | i mean... do i have to specify each one individually |
| 11:30.42 | v0lZy | or can i do something with extensions |
| 11:30.54 | v0lZy | like s,hint,SIP/${EXTEN} or something |
| 11:31.03 | WIMPy | Depends on your securiy needs. |
| 11:31.37 | WIMPy | Hmm. There was some (limited?) support for patterns, but I don't really know. |
| 11:32.14 | dlynes_laptop | Are hints working properly now? The last time I tried them, they were failing miserably |
| 11:32.31 | dlynes_laptop | I haven't tried using them since v1.4.something, though |
| 11:32.41 | WIMPy | dlynes_laptop: How manyu decades ago was that? |
| 11:32.43 | dlynes_laptop | And much older Aastra firmware versions |
| 11:33.04 | dlynes_laptop | WIMPy, decades? No...more like about 5 or 6 years ago |
| 11:33.05 | WIMPy | I'm pretty sure they used to work on 1.4. |
| 11:33.19 | dlynes_laptop | WIMPy, they worked on 1.4...just not very well |
| 11:33.44 | dlynes_laptop | WIMPy, sometimes a reload would throw them into chaos |
| 11:33.51 | WIMPy | Either they work, or they don't. |
| 11:33.59 | dlynes_laptop | WIMPy, often reboots of the phone would cause them to stop working |
| 11:34.25 | dlynes_laptop | WIMPy, it would often be dependent on which phone came up first as to whether they worked or not |
| 11:34.34 | WIMPy | Are you sure that's an Asterisk issue and not a phone issue? |
| 11:34.48 | v0lZy | sounds like a phone issue |
| 11:34.51 | v0lZy | had that stuff too |
| 11:34.51 | dlynes_laptop | WIMPy, no idea...but hints were in their infancy at that point |
| 11:34.51 | WIMPy | Sounds strange. |
| 11:34.55 | v0lZy | updated firmware fixed it |
| 11:34.59 | v0lZy | wimpy |
| 11:35.01 | v0lZy | if i put |
| 11:35.06 | v0lZy | subscribecontext=BLF |
| 11:35.26 | v0lZy | and then in my extensions.conf i do [BLF] and in it exten => 16,hint,SIP/16 etc |
| 11:35.30 | v0lZy | that should do the trick? |
| 11:35.36 | v0lZy | or must i include this in each phones context? |
| 11:35.41 | WIMPy | yes |
| 11:35.47 | v0lZy | (i send each phone to its own context) |
| 11:35.52 | v0lZy | include = yes/no? |
| 11:36.03 | WIMPy | Only in the subscribecontext. |
| 11:36.17 | v0lZy | k |
| 11:36.39 | dlynes_laptop | v0lZy, you can also specify subscribecontext=BLF in a template, and then inherit that template for each peer |
| 11:37.01 | v0lZy | in sip.conf i added subscirbecontext=BLF |
| 11:37.06 | v0lZy | took a minute or two but now i got the lights :D |
| 11:37.30 | dlynes_laptop | v0lZy, yeah...it has to get updates from the phones |
| 11:37.49 | v0lZy | ok i think i got a basic setup working now |
| 11:38.15 | v0lZy | call forwarding set/unset, check |
| 11:38.40 | v0lZy | voicemail (for internal use only), leave message + check messages, check |
| 11:38.48 | v0lZy | BLF, check |
| 11:39.03 | v0lZy | music on hold, check |
| 11:39.08 | v0lZy | transfer stuff... hm... |
| 11:39.10 | v0lZy | test. |
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| 11:40.39 | v0lZy | ok attended transfer doesnt update the phones callerid |
| 11:40.50 | v0lZy | blind transfer does however. |
| 11:41.02 | v0lZy | same issue as a while ago... this time on a fresh install... humm. |
| 11:42.05 | WIMPy | The whole callerid handling needs some tidying up. That includes documentation. I never know what's a bug and what's a design failure. |
| 11:43.07 | v0lZy | btw |
| 11:43.30 | v0lZy | can sip phones dial themsleves directly through ip's without a pbx in between? |
| 11:43.46 | WIMPy | Sure |
| 11:43.56 | WIMPy | SIP is a P2P thing. |
| 11:44.17 | v0lZy | i see |
| 11:44.21 | v0lZy | a pbx is pure convenience then |
| 11:44.29 | v0lZy | for entering numbers i suppose |
| 11:44.35 | v0lZy | my phone cant dial ip i think |
| 11:44.46 | v0lZy | i mean.. i dont know of a way to dial an ip from it :D |
| 11:44.46 | WIMPy | And to locate the phones. |
| 11:45.29 | v0lZy | well if u have an IP |
| 11:45.33 | v0lZy | whats there to locate |
| 11:45.40 | v0lZy | say u have a public IP |
| 11:45.46 | WIMPy | On some you can just enter 12*34*56*78. If all else fails, you should be able to use the phone book to dial URLs. |
| 11:45.57 | v0lZy | ah |
| 11:45.59 | v0lZy | star |
| 11:46.02 | v0lZy | lets see |
| 11:46.05 | WIMPy | IF |
| 11:46.22 | v0lZy | holly shit it works |
| 11:46.32 | v0lZy | didnt know my phone can do that :D |
| 11:47.07 | v0lZy | what about blf stuff |
| 11:47.25 | WIMPy | Depends on the phone. |
| 11:47.28 | v0lZy | do they pass it around without a server (though i imagine unreliable) |
| 11:47.34 | WIMPy | But might be possible. |
| 11:48.08 | WIMPy | If it works, you subscribe to the phone in question directely. |
| 11:48.44 | v0lZy | via IP... |
| 11:49.02 | v0lZy | i imagine latency etc issues would make it uselesss |
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| 11:56.38 | v0lZy | i guess all i need now is to hookup to an external provider |
| 11:56.41 | v0lZy | see if it all works |
| 11:56.56 | v0lZy | and configure intercom, yeah, forgot about that one |
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| 12:08.59 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:10.35 | kresp0 | welcome [TK]D-Fender |
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| 12:41.48 | v0lZy | question about hangup |
| 12:42.11 | v0lZy | should i place it after Dial? |
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| 12:42.50 | v0lZy | as I mean... if a dial fails for example.. non existing extension etc |
| 12:42.59 | v0lZy | a hangup probably makes sense after that, right? |
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| 12:46.24 | *** join/#asterisk danfromuk (~IceChat77@2.27.27.15) |
| 12:46.40 | danfromuk | Hi. In GotoIf, how do you do an OR statement. |
| 12:46.47 | danfromuk | Is it ||? |
| 12:46.56 | danfromuk | Or a single | maybe? |
| 12:47.20 | WIMPy | v0lZy: Yes. |
| 12:47.46 | [TK]D-Fender | danfromuk, || |
| 12:47.55 | [TK]D-Fender | danfromuk, and that is in an expressions, not "gotoif" |
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| 12:52.42 | danfromuk | great, thanks. |
| 12:53.24 | danfromuk | Also, in the h extension once a call is ended, how do you stop execution? Will Hangup still stop execution even though the call is already hungup? |
| 12:54.12 | [TK]D-Fender | yes |
| 12:54.25 | danfromuk | Thanks again. |
| 12:55.51 | kaldemar | one pipe is enough for or. |
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| 13:01.38 | v0lZy | btw |
| 13:01.49 | v0lZy | is there a way to grab who im transfering? |
| 13:01.52 | v0lZy | from sip channels or something? |
| 13:03.06 | [TK]D-Fender | If someone dials a non-existant extension the call will never have gotten accepted in the first place |
| 13:03.48 | [TK]D-Fender | And to grab someone yuo've tansfered you chould use an AMI redirect |
| 13:03.58 | [TK]D-Fender | and you'll have to be able to isolate the target channel... |
| 13:04.02 | WIMPy | If it was dialled en-block. |
| 13:05.14 | v0lZy | im thinking if its at all possible |
| 13:05.28 | v0lZy | to have asterisk read back a number that was transfered through attended transfer |
| 13:05.58 | *** join/#asterisk Forge__ (~forge@94.100.30.139) |
| 13:06.25 | v0lZy | or something to supplement the behaviour where my attended transfers dont update the screen |
| 13:07.41 | [TK]D-Fender | Stop doing attended transfers |
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| 13:10.25 | v0lZy | [TK]D-Fender: yeah, i told my users that they can put the original call on hold, call the person they are transfering to, have that person hangup on them, then blindtransfer... |
| 13:10.37 | v0lZy | but everyone's a george jetson these days |
| 13:10.58 | v0lZy | pressing more than one button is herecy to them |
| 13:11.20 | [TK]D-Fender | These are the rules. Don't like what happens when you don't follow them? Too bad. |
| 13:11.44 | v0lZy | wish people here would go with that |
| 13:11.56 | WIMPy | Hooray for key system emulation mode! |
| 13:12.03 | v0lZy | this is the balkans |
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| 13:12.38 | v0lZy | its such a cliche... 'thore are the rules' results in 'rules are ment to be broken' smartass nirvana quotint or whoever that was. |
| 13:13.14 | WIMPy | That's te spirit. |
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| 13:15.12 | [TK]D-Fender | v0lZy, Tell them to jsut off a cliff and tell gravity that it has to stop as well... |
| 13:15.17 | [TK]D-Fender | jump* |
| 13:16.00 | v0lZy | i wish they'd try and prove me wrong on that one :D |
| 13:16.13 | v0lZy | would sovle 90% of the problems in this shithole |
| 13:23.07 | v0lZy | i actually had someone ask me if they also need electricity or is just internet ok |
| 13:23.34 | v0lZy | thought it was stupidity back then |
| 13:23.52 | v0lZy | turns out its much more deeply rooted in the people here not wanting to learn anything |
| 13:24.07 | v0lZy | and accept their surroundings the way they are |
| 13:24.47 | v0lZy | a bullshitting environment where the only measure is how much u can spin it :D |
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| 13:26.05 | v0lZy | hence, no need to understand or accept anything if u can always squeeze yourself out with your tongue. |
| 13:28.41 | danfromuk | Is CDR(duration) supposed to show the length of a call including the ring time? |
| 13:28.52 | danfromuk | Currently it seems to store the start time of a call. |
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| 13:31.40 | [TK]D-Fender | duration - billsec = ringing |
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| 13:38.08 | v0lZy | gotta run |
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| 13:38.12 | v0lZy | talk tomorrow guys, bye |
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| 13:45.06 | leifmadsen | danfromuk: yes, duration is total call length. billsecs is non-ring time. |
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| 14:14.40 | kresp0 | I heard that asterisk 1.8 has problems with SIP TLS. Anyone knows if that true for the last asterisk versions? |
| 14:16.54 | bchia | I remember there being a bug initially with TLS, but I thought it got sorted out. At the moment I can't track down the annoucement |
| 14:19.29 | [TK]D-Fender | kresp0, But happen. They also tend to get fixed. So saying "I heard something" usually doesn't mean much. I suggest you actually try for yourself and if you actually end up having trouble, then show us. |
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| 14:21.56 | bchia | That's good advice - if you search https://issues.asterisk.org/jira/ for "SIP TLS" you can actually see a lot of bugs have been fixed. |
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| 14:23.03 | Katty | gooood morning lovables. |
| 14:23.25 | n3hxs | Ahh, little Miss Sunshine! Morning Katty |
| 14:23.36 | Katty | how're you dear |
| 14:23.38 | Katty | hugs n3hxs |
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| 14:23.52 | n3hxs | So far so good. |
| 14:24.03 | bchia | Hey that's nice "loveables" :) |
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| 14:40.23 | kresp0 | thank you [TK]D-Fender and bchia |
| 14:43.49 | *** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 14:48.01 | mcf3782 | For several days now, I've seen messages like this showing up in my Asterisk log file: |
| 14:48.09 | mcf3782 | [Aug 14 03:09:37] NOTICE[2483] chan_sip.c: Call from '' to extension '00972595339204' rejected because extension not found in context 'Internal'. |
| 14:48.28 | [TK]D-Fender | Means just what it says |
| 14:48.40 | mcf3782 | I'm concearned that I have a security issue. |
| 14:48.51 | mcf3782 | Nobody is in the building at that time. |
| 14:48.59 | [TK]D-Fender | are you allowing unauthed calls? |
| 14:49.14 | [TK]D-Fender | Did you point [general] in sip.conf to somewhere USEFUL? |
| 14:49.35 | pabelanger | mcf3782: *CLI> dialplan show 00972595339204@Internal |
| 14:49.43 | pabelanger | Oh |
| 14:49.49 | pabelanger | ya, security |
| 14:50.19 | mcf3782 | I'm not *supposed* to be allowing unauthed calls. |
| 14:50.52 | *** join/#asterisk mjordan (~mjordan@c-50-130-8-246.hsd1.al.comcast.net) |
| 14:50.52 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:51.21 | kaldemar | guesses mcf3782 does not have allowguest=no and has context=Internal in [general]. |
| 14:51.50 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:52.26 | [TK]D-Fender | mcf3782, now try answering those questions... after actually looking. |
| 14:52.43 | kresp0 | mcf3782: a better name for that context would be "unauthorized" instead of "Internal" |
| 14:52.53 | mcf3782 | I am looking at the files again now. :) |
| 14:53.29 | pabelanger | anybody using a blackfin with asterisk? |
| 14:53.31 | kresp0 | or "not-authenticated" |
| 14:53.33 | *** part/#asterisk mjordan (~mjordan@c-50-130-8-246.hsd1.al.comcast.net) |
| 14:56.06 | mcf3782 | There is no "allowguest=" in my sip.conf. Does it default to 'yes' if it's not explicitly set to 'no'? |
| 14:57.26 | mcf3782 | I've had a tcpdump capture running on the mirror port on the switch since these messages started showing up in the Asterisk log file. What's troublesome is that I can't match any captured network traffic timestamps to the timestamps on the Asterisk log messages. |
| 14:58.20 | bchia | according the sample config "allowguest" is "default enabled" http://svn.asterisk.org/svn/asterisk/branches/1.8/configs/sip.conf.sample |
| 14:59.10 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:59.16 | [TK]D-Fender | mcf3782, First step, where is [general] pointing? |
| 14:59.45 | mcf3782 | Ahh.. I wonder if that changed somewhere previous to 1.8, and the old default was 'default disabled'? |
| 15:00.20 | mcf3782 | TK [general] points to 'context=Internal'. At least I think that's the answer to the question you're asking. |
| 15:01.20 | [TK]D-Fender | it is |
| 15:01.53 | [TK]D-Fender | If that is meant to have something useful in it then you should almost certinaly NOT be sending [general] tehre |
| 15:02.08 | [TK]D-Fender | Real peers should go somewhere else |
| 15:03.03 | mcf3782 | OK. I'll add that to the list of things that need to be changed on this box. |
| 15:06.51 | *** join/#asterisk navaismo (~navaismo@189.191.94.251) |
| 15:22.01 | mcf3782 | I pulled a copy of sip.conf from a backup a month ago, to see if anything had changed. When that backup was made, [general] pointed to context=default. Looks like I may have more than one kind of security issue. |
| 15:22.16 | mcf3782 | *sigh* |
| 15:22.16 | mcf3782 | Would someone please shoot me. |
| 15:22.29 | lorsungcu | that is illegal, unfortunately. |
| 15:22.47 | mcf3782 | details, details. I promise I won't tell anyone. |
| 15:23.06 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:25.30 | *** join/#asterisk retentiveboy (~retentive@74-95-28-34-Atlanta.hfc.comcastbusiness.net) |
| 15:25.34 | _Corey_ | lorsungcu: Only if you're aiming to kill... maiming could be legal |
| 15:26.12 | mcf3782 | _Corey_ - Anything other than a kill shot, is wasted ammo. ;) |
| 15:26.27 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:26.28 | kresp0 | mcf3782: check your machine logs |
| 15:26.36 | kresp0 | /var/log/auth.log |
| 15:26.38 | kresp0 | etc |
| 15:27.32 | kresp0 | if someone has changed that sip.conf, it has to be root |
| 15:27.35 | kresp0 | or the asterisk user |
| 15:28.30 | kresp0 | check if you have new users on that machine |
| 15:28.34 | mcf3782 | I've got a lot of log files to look through. I'm going to take the box down this evening and do a dd image of the system drive so I have something for forensics if needed. |
| 15:29.09 | kresp0 | good |
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| 15:49.48 | mcf3782 | I also need to sort through the rats nest of cables on the network switch. It seems very strange that the pbx appears to be seeing traffic (the traffic that's causing the log message I posted); that the mirror port isn't sending to the machine where my tcpdump is capturing. |
| 15:51.22 | mcf3782 | All the while, trying not to look too suspicious.. If part of my security problem is internal, I don't want to tip off the "problem" that I'm onto him/her. |
| 15:51.41 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-193.mobile.uci.edu) |
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| 15:55.28 | lorsungcu | you're right. it could be anyone. |
| 15:55.52 | lorsungcu | they might even be on this channel |
| 15:55.53 | lorsungcu | right now |
| 15:57.01 | mcf3782 | Could be. :) |
| 15:59.00 | *** join/#asterisk danfromuk (~IceChat77@2.27.27.15) |
| 16:00.08 | mcf3782 | I wonder how much paperwork and useless hoop-jumping is required to get the lock on the network room changed? |
| 16:01.33 | danfromuk | Has anyone set up a SIP endpoint on Virgin Media? |
| 16:01.50 | danfromuk | Specifically, what do you need to do with the router to get the SIP packets through? |
| 16:08.09 | kresp0 | mcf3782: did you know who are the persons with the need to have the keys to the network room? |
| 16:08.47 | kresp0 | if so, change the lock now and give a copy + explanation to each person when possible |
| 16:11.13 | lorsungcu | i would hope that you have access to your equipment from outside a locked closet, though... |
| 16:11.30 | mcf3782 | Unfortunately, that's a big list. |
| 16:11.37 | mcf3782 | I do. :) |
| 16:12.14 | lorsungcu | then i would be more concerned with the 8 billion other people that fall into the "outside of your network closet" list |
| 16:12.28 | mcf3782 | Good grief! The building people want $500 to change the lock, and $20/key!!! |
| 16:12.46 | lorsungcu | cheaper than the potential phone bill.. |
| 16:14.16 | *** part/#asterisk jsjc (~Adium@237.157.78.188.dynamic.jazztel.es) |
| 16:14.59 | mcf3782 | lorsungcu - the point of the network closet access, is to figure out if someone has added some device somewhere, that is making it possible for the PBX to see traffic that my sniffer port isn't seeing. |
| 16:16.37 | lorsungcu | i can't really comment on how your switch works |
| 16:16.57 | lorsungcu | but if you arent seeing any traffic |
| 16:17.03 | lorsungcu | and calls are being attempted |
| 16:17.16 | lorsungcu | then sounds like someone's got call files made up |
| 16:17.35 | lorsungcu | and is attempting that way |
| 16:17.59 | lorsungcu | check the /etc/passwd file |
| 16:18.06 | lorsungcu | see if theres anything new there |
| 16:18.44 | mcf3782 | nothing new in /etc/passwd or /etc/shadow |
| 16:23.13 | lorsungcu | checked /var/log/secure? |
| 16:23.55 | mcf3782 | yep. Nothing odd looking in there either. |
| 16:24.10 | lorsungcu | how much logging do you have |
| 16:24.26 | mcf3782 | When I come back from lunch, I'm going to install tripwire on the pbx |
| 16:30.22 | *** part/#asterisk whito (~whit@bas3-windsor12-1128740895.dsl.bell.ca) |
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| 16:47.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 17:05.16 | *** join/#asterisk lkcl (~lkcl@host81-154-42-230.range81-154.btcentralplus.com) |
| 17:06.15 | lkcl | hi, i'm getting the app_conference module up-and-running, with asterisk 1.8 - it's not compiling because cid is now missing from ast_channel |
| 17:07.20 | WIMPy | It has been renamed/restructured, but in Asterisk 11 it is completely gone. |
| 17:07.27 | lkcl | anyone got any clues as to how to obtain the... ok... |
| 17:07.43 | lkcl | joy! :) |
| 17:07.44 | mjordan | WIMPy: well, not completely. Just hidden a bit :-) |
| 17:07.46 | leifmadsen | ya, use app_confbridge |
| 17:07.47 | WIMPy | So it might be worth to skip thae versions in between. |
| 17:07.54 | Qwell | leifmadsen: 1.8 |
| 17:07.57 | leifmadsen | ahhh |
| 17:08.01 | WIMPy | Well, gone for 3rd parties. |
| 17:08.01 | leifmadsen | don't use that crap |
| 17:08.04 | Qwell | heh |
| 17:08.09 | leifmadsen | asterisk 11-beta1 is the new hotness! |
| 17:08.18 | mjordan | WIMPy: what do you mean? |
| 17:08.20 | lkcl | leifmadsen: does app_confbridge... ok, actually it doesn't matter what i use as long as i can get videoconferencing properly working. |
| 17:08.21 | leifmadsen | turns down the troll volume |
| 17:08.22 | Qwell | leifmadsen: You're still on beta1? You're so far behind. |
| 17:08.40 | leifmadsen | lkcl: video conferencing? use asterisk 10 with app_confbridge |
| 17:08.43 | Qwell | lkcl: Use Asterisk 10, so you get app_confbridge. |
| 17:08.44 | leifmadsen | it does video conf |
| 17:08.45 | lkcl | the goal is: "get a working, stable videoconferencing system up and running" |
| 17:08.51 | leifmadsen | ya use asterisk 10 |
| 17:09.04 | lkcl | ahhh, ok - thanks. ok, so what features does it have? H263? |
| 17:09.12 | mjordan | lkcl: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
| 17:09.14 | leifmadsen | H.263 and H.264 |
| 17:09.16 | lkcl | i'll ... ahh goodie. |
| 17:09.18 | WIMPy | mjordan: That we no longer see things like caller id structures on the channel. |
| 17:09.18 | leifmadsen | mjordan: well played sir |
| 17:09.21 | lkcl | mjordan: good stuff |
| 17:09.35 | mjordan | WIMPy: right, but they aren't gone. You can still access the information using the channel API |
| 17:09.39 | lkcl | goes off to read up.... |
| 17:09.40 | Qwell | WIMPy: It's just opaque now. |
| 17:10.04 | WIMPy | It requires changes again. |
| 17:10.10 | mjordan | WIMPy: that is true |
| 17:10.10 | Qwell | Things change. |
| 17:10.11 | WIMPy | That was my point. |
| 17:10.19 | mjordan | okay :-) I can definitely agree with that |
| 17:10.47 | lkcl | HA! app_confbridge looks f****g well-cool. HA |
| 17:10.54 | WIMPy | But indeed it might not be worth anyway. |
| 17:11.55 | mjordan | lkcl: just as an FYI on video conferencing, Asterisk 11 introduced media format negotiation. Asterisk 10 is much closer conceptually to Asterisk 11 then Asterisk 1.8 is to Asterisk 10 - so if you have disparate endpoints that require media negotiation and design your system around Asterisk 10, when Asterisk 11 is released it'll be an easier upgrade path |
| 17:11.56 | lkcl | video_mode - ok, it doesn't do transcription, and everyone must do the same CODEC, but that's ok for a first try. |
| 17:13.05 | Qwell | lkcl: video transcoding is super expensive |
| 17:13.06 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:13.10 | lkcl | mjordan: i'm setting something up for a friend (paid-for though!), they want eaaaasy. no-one need do "logins", you just put in your name and join. |
| 17:13.24 | lkcl | Qwell: that would explain why openmcu was a dog, even when it worked |
| 17:13.43 | lkcl | rrrright! i have to try this. |
| 17:13.45 | mjordan | lkcl: ConfBridge doesn't require pins and whatnot if you don't want it to |
| 17:13.50 | lkcl | goooooood |
| 17:13.58 | lkcl | ok right now to find asterisk 10 source code... |
| 17:14.56 | Qwell | source code? |
| 17:14.59 | lkcl | mjordan: by endpoints, you mean the clients? (ekiga, netmeeting) |
| 17:15.04 | lkcl | Qwell: ha ha very funny |
| 17:15.17 | mjordan | lkcl: yes |
| 17:15.26 | leifmadsen | lkcl: svn co http://svn.asterisk.org/svn/asterisk/branches/10 |
| 17:16.04 | WIMPy | If you use svn, you can get your own 11 beta. |
| 17:16.36 | lkcl | mjordan: ok so i'll be testing ekiga (w32, linux) and xmeeting (macosx), see how they get on. ekiga's not available for macosx. this is real cross-platform stuff! real big mix of people using this stuff. |
| 17:16.42 | lkcl | leifmadsen: taaaaa |
| 17:16.59 | leifmadsen | I like jitsi |
| 17:17.08 | leifmadsen | heavily tested with video conf on ConfBridge() |
| 17:17.48 | mjordan | I've used jitsi, ekiga, and blink on Mac myself |
| 17:18.24 | lkcl | leifmadsen: yeah? ah ha! ok, i remember using jitsi: it forced me to use a username and registration. i believe i managed to put it into "advanced" mode to stop it from stopping me making calls |
| 17:18.30 | lkcl | if that makes any sense :) |
| 17:18.33 | *** join/#asterisk learath_ (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 17:18.55 | lkcl | but i can't tell people "oh yes, download jitsi, then go into advanced mode, *then* you can call" - it's too complex for them. |
| 17:19.09 | lkcl | mjordan: ekiga is available for mac?? |
| 17:19.44 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:20.28 | lkcl | ok.... although i'm doing an svn co of ast10, what's the chances of app_confbridge working in v11? |
| 17:20.44 | leifmadsen | the chances? |
| 17:20.47 | leifmadsen | I suspect quite high |
| 17:20.50 | WIMPy | 0 |
| 17:20.55 | lkcl | leifmadsen: he he |
| 17:21.01 | lkcl | WIMPy: none? ah well.... |
| 17:21.02 | WIMPy | Which one? |
| 17:21.10 | drmessano | Not sure I would use a beta 1 in production, but I don't expect it to be broken |
| 17:21.13 | leifmadsen | why would app_confbridge work in one version but not another? |
| 17:21.24 | Qwell | drmessano: pfft, Asterisk betas are always perfect |
| 17:21.45 | *** join/#asterisk jkroon (~jkroon@dsl-244-38-32.telkomadsl.co.za) |
| 17:21.48 | drmessano | Is that why the releases are buggy as hell? |
| 17:21.52 | lkcl | drmessano: i liiike working on cutting-edge stuff. |
| 17:21.56 | drmessano | Bugs added on in RC stage |
| 17:22.25 | WIMPy | Actually, I think I had less issues with SVN versions than with releases. |
| 17:22.37 | drmessano | lkcl: It's not cutting-edge, it's "testing". Difference between using the newest, and using something in beta. |
| 17:22.51 | drmessano | 11.0.0 would be cutting-edge |
| 17:23.40 | Katty | nomnomnom |
| 17:23.48 | lkcl | okok :) |
| 17:24.03 | lkcl | rright. hmmm.... configuration options.... choices, choices... |
| 17:24.08 | leifmadsen | I'd consider a checkout from subversion branch to be cutting edge :) |
| 17:24.08 | WIMPy | Apart from the ICE stuff, TRUNK has been working very well for several weeks for me. |
| 17:24.34 | Katty | nothing works on ice! |
| 17:24.38 | lkcl | ok. openh323 stuff... (deep joy) |
| 17:24.51 | leifmadsen | ya don't use it |
| 17:24.53 | lkcl | i've just been playing with openh323 *shudder*. |
| 17:25.56 | lkcl | i actually managed to get h323plus compiled! stunning achievement! openmcu actually worked, did some video (wow)... but it worked once and only once, and segfaulted on a 2nd user connection |
| 17:26.13 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:26.45 | lkcl | anyone got any recommendations for video codecs.... oh wait, app_confbridge doesn't do any transcription so it doesn't matter, does it |
| 17:26.52 | leifmadsen | nope |
| 17:26.55 | leifmadsen | use h.264 |
| 17:27.04 | leifmadsen | everywhere |
| 17:27.07 | lkcl | leifmadsen: ack. |
| 17:27.29 | jkroon | confbridge can deal with video? |
| 17:28.02 | lkcl | jkroon: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 - it can pass it through |
| 17:28.05 | lkcl | unaltered. |
| 17:28.13 | [TK]D-Fender | "Follow the speaker" |
| 17:28.19 | [TK]D-Fender | no MUX at all |
| 17:28.21 | lkcl | so each caller has to have the same CODECs |
| 17:28.47 | jkroon | ok, so how does it decide who the "speaker" is? or does it listen in somehow? |
| 17:28.49 | [TK]D-Fender | which is people start a circle-jeck of talking it could get really dizzy watching the feeds flip ;) |
| 17:28.50 | lkcl | btw, openmcu actually has some code which puts up to 4 speakers as picture-in-picture |
| 17:28.57 | lkcl | jkroon: read the wiki page - i did |
| 17:29.00 | [TK]D-Fender | jkroon, speaker volume |
| 17:29.16 | jkroon | ok, that's pretty cool! |
| 17:29.37 | lkcl | pbx mxml-index.c:35:20: fatal error: config.h: No such file or directory |
| 17:29.38 | lkcl | beh?? |
| 17:29.43 | Qwell | [TK]D-Fender: I think there's something there to prevent that |
| 17:29.58 | lkcl | ahh, make -j3 bolloxed things up... |
| 17:30.12 | Qwell | [TK]D-Fender: like a minimum "stop" duration from the previous speaker, before the next is allowed...or something? |
| 17:30.23 | Qwell | I know there was discussion of something like that, anyways. |
| 17:30.28 | [TK]D-Fender | That would be nice if it isn't there already |
| 17:30.41 | [TK]D-Fender | Also nice maybe if there was a prioritization level... |
| 17:31.50 | [TK]D-Fender | Also great if there was an option to "lock" onto specific callers so someone "directing" a conference can ensure a certain flow. |
| 17:32.24 | lkcl | bloody hell that was a quick build. |
| 17:32.41 | jkroon | sruffell, ok, just merged and ran your debug build ... how bad is it if there is _nothing_ in the dmesg output? |
| 17:32.44 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 17:34.45 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:34.57 | sruffell | echo 1 > /sys/modules/wctdm24xxp/parameters/debug to see anything |
| 17:35.22 | luckman212 | anyone have a moment to do a DTMF test for me? |
| 17:35.49 | luckman212 | i am having trouble dialing into one particular IVR (it doesn't recognize my DTMF) and I'm trying to see if the issue lies on my end or theirs |
| 17:36.11 | luckman212 | out of 3 SIP trunks I have, only 1 of them seems to be able to reliably pass thru DTMF to this IVR |
| 17:37.56 | jkroon | sruffell, already did, only output I got was when loading chan_dahdi.so into asterisk about it setting VMWI on channels 20 and 21, and then SLIC_LF OK lines for the same channels. |
| 17:39.07 | luckman212 | er... pretty please? |
| 17:39.54 | sruffell | jkroon: ok…let me make another patch….looks like it's not even crossing the threshold. |
| 17:40.46 | lkcl | ok - help request! i need an absolute minimum basic config setup (from scratch) that will allow me to get confbridge up-and-running. |
| 17:40.47 | jkroon | waits patiently, thanks for your efforts. |
| 17:41.29 | lkcl | it's pretty much dedicated-purpose for this server: there's not going to be any authentication, no users, no radius, mysql, ldap - nothing. |
| 17:41.49 | lkcl | you dial, you're joined - that's it. |
| 17:43.00 | lkcl | i set up and maintained asterisk once back in 2003-2005 (including with a digium 2-channel card, woo-hoo!) so roughly know what i'm doing |
| 17:44.11 | jkroon | lkcl, it changed a LOT ... mostly for the better. |
| 17:44.47 | lkcl | :) |
| 17:45.17 | lkcl | jkroon: top of my wish-list is the entire connectivity of asterisk being controllable via python. |
| 17:45.32 | jkroon | look into the AMI |
| 17:45.39 | lkcl | but anyway.... |
| 17:45.58 | jkroon | or possibly (E)AGI - haven't ever used this though. |
| 17:46.11 | lkcl | ok. so. i have a blank /etc/asterisk subdirectory - what's the fastest way to get up-and-running with confbridge? |
| 17:46.38 | WIMPy | 'make samples' |
| 17:46.48 | lkcl | WIMPy: ack. ta |
| 17:47.08 | lkcl | ... in the correct directory of course ha ha |
| 17:50.05 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:50.48 | lkcl | ok - right, following the wiki, i now have a conference called "fancybridge" - how do i make it callable? |
| 17:52.12 | WIMPy | Make an extension. |
| 17:52.19 | lkcl | playing the "dumb admin here" - sorry folks - there's nothing in the wiki which says how to actually connect the bridges (whatever those are)... |
| 17:52.26 | lkcl | ah... i was expecting that answer :) |
| 17:52.55 | WIMPy | accidentally read "dump admin here" |
| 17:52.59 | lkcl | ha ha |
| 17:53.42 | lkcl | ok so it's just.... |
| 17:53.53 | lkcl | exten => 500,n,Dial(fancybridge) |
| 17:53.57 | lkcl | for example |
| 17:54.02 | lkcl | would that do the trick? |
| 17:54.03 | [TK]D-Fender | No |
| 17:54.16 | [TK]D-Fender | this is not a "dial" |
| 17:54.21 | lkcl | [TK]D-Fender: apologies, i haven't used this stuff for a long time |
| 17:54.31 | [TK]D-Fender | it is its own application |
| 17:54.34 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 17:54.54 | [TK]D-Fender | "core show application confbridge" |
| 17:54.56 | [TK]D-Fender | ^ |
| 17:55.03 | lkcl | [TK]D-Fender: sorry, that still doesn't give me a clue |
| 17:55.08 | lkcl | [TK]D-Fender: is that a console command? |
| 17:55.11 | [TK]D-Fender | yes |
| 17:55.18 | [TK]D-Fender | read the apps instructions |
| 17:55.23 | lkcl | yep got it. |
| 17:55.28 | sruffell | jkroon: Can you set the debug flag before running dahdi_cfg and tell me if you see lines like BATTERY on 2/1 (-)! in the dmesg output? |
| 17:56.41 | leifmadsen | sruffell: !! |
| 17:56.45 | leifmadsen | sruffell: ohai |
| 17:56.52 | lkcl | ; Or a conference room (you'll need to edit meetme.conf to enable this room) |
| 17:56.53 | lkcl | ; |
| 17:56.53 | lkcl | ;exten => 8600,1,Meetme(1234) |
| 17:56.57 | lkcl | ah ha! |
| 17:57.01 | sruffell | waves at leifmadsen |
| 17:57.01 | lkcl | is beginning to get it |
| 17:57.08 | leifmadsen | sruffell: did you happen to get my memo? |
| 17:57.17 | sruffell | ??? |
| 17:57.21 | jkroon | The AOC stuff also looks very interesting. |
| 17:57.24 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:57.25 | leifmadsen | sruffell: /msg memoserv read 1 |
| 17:57.32 | [TK]D-Fender | lkcl, Not MeetMe. that is ANOTHER conferencing app |
| 17:57.33 | leifmadsen | I think that's the format :) |
| 17:57.35 | jkroon | sruffell, sure, will quickly do so |
| 17:58.09 | jkroon | sruffell, setting debug=1 during insmod should be sufficient? |
| 17:58.18 | *** part/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 17:58.24 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 17:58.25 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:58.26 | sruffell | yes…will just run some more verbose tests during start…but won't hurt anything |
| 18:00.36 | jkroon | dahdi: Warning: Span WCTDM/0 didn't specify a spantype. Please fix driver! |
| 18:01.06 | lkcl | bloodyhellfire, it's connected |
| 18:01.19 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 18:01.20 | sruffell | yeah..don't worry about that…. |
| 18:01.26 | lkcl | rushes upstairs to try connecting the 2nd computer.... |
| 18:01.30 | jkroon | sruffel pastebin.com/5g1qhKcR |
| 18:01.54 | sruffell | and then when you run dahdi_cfg? |
| 18:02.07 | jkroon | not seeing any battery messages. |
| 18:02.20 | jkroon | jip |
| 18:02.35 | jkroon | and then re-ran it ... |
| 18:03.11 | jkroon | got BATTERY on lines 1,2,3,4,5,7,8, NO BATTERY on 6 - which is correct (line is currently dead in the water) |
| 18:03.36 | jkroon | must've run dmesg too quickly after dahdi_cfg ... |
| 18:03.39 | sruffell | and now if you call in…still no ring state transitions? |
| 18:03.43 | lkcl | holyyy shit, i have video working! |
| 18:04.12 | jkroon | sruffell, i'm being a dumb nut. dialed the wrong number earlier... |
| 18:04.25 | lkcl | hmmmm.... |
| 18:04.28 | sruffell | crosses fingers for good info |
| 18:04.43 | *** join/#asterisk l2trace99 (~l2trace@rrcs-71-43-104-238.se.biz.rr.com) |
| 18:04.45 | jkroon | you got it. |
| 18:05.23 | jkroon | pastebin.com/iEawn7T0 |
| 18:06.25 | jkroon | that look pretty sane IMHO. |
| 18:06.39 | sruffell | ahh…so not enough transitions. |
| 18:06.43 | sruffell | progress! |
| 18:07.06 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 18:07.27 | sruffell | I'll push something in a min here for you to fetch and try |
| 18:08.04 | jkroon | yea, seems to mostly transit four times before returning to ringoff, and every now and again it transits between positive/negative and ringoff in single cycles. |
| 18:10.01 | sruffell | jkroon: ok..dahlin-289-1 branch updated..you should just be able to git pull and try again |
| 18:11.30 | sruffell | dahlin-298-1 :/ |
| 18:12.26 | jkroon | that simple? Why not two? Or even 1? |
| 18:13.11 | sruffell | two might be good……I'm not sure if one would interfere with NEON MWI |
| 18:16.26 | jkroon | sruffell, from my perspective (given that I don't understand the ins and outs of DAHDI) it's quite simple - the quicker we can detect that a line starts ringing the better, and the same for "stops ringing", because if I need to explain to another receptionist why she sometimes gets "dead air" ... |
| 18:16.30 | jkroon | ok, that was pretty quick into the dialplan already ... |
| 18:17.12 | jkroon | and, as a more important message mr sruffell - you're a star! |
| 18:17.16 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 18:17.27 | jkroon | that works, thank you very much. |
| 18:18.35 | jkroon | sruffell, need me to comment on the bug? |
| 18:19.17 | teloniusz | hi. I have a strange problem with E1 communication - asterisk working as a SIP->E1 gate. |
| 18:19.29 | lkcl | [TK]D-Fender: i was using that MeetMe thing as a way to understand about how to connect to an application, via extensions.conf. i've got it now (yay!) - i'm just recompiling on my server because ekiga wants port 5060 and asterisk wants port 5060.. oops |
| 18:20.23 | WIMPy | Maybe it would be easier to look at the cofig file. |
| 18:20.26 | teloniusz | When a call comes from SIP, sequence of events is such that my asterisk sends SETUP, receives PROCEEDING, then CONNECT, then sends CONNECT ACK and for no apparent reason after that sends DISCONNECT the very second. |
| 18:21.05 | WIMPy | teloniusz: There must be a reason. Show us the debug. |
| 18:21.08 | teloniusz | So the connection as a whole ends with "Normal call clearing" (code 16) |
| 18:21.57 | teloniusz | WIMPy: a moment... |
| 18:22.45 | sruffell | jkroon: cool! thanks for working with me to track it down. I'll check with those who know MWI better to see what the best setting is. But yeah, if you wanted to comment on the issue that would be great. |
| 18:22.51 | lkcl | jkroon: about the "python asterisk" stuff, i meant like this: http://code.google.com/p/py-asterisk/ |
| 18:23.45 | Qwell | lkcl: we (the Asterisk developers) use starpy for a lot of stuff |
| 18:24.03 | jkroon | sruffell, perfect, will do. |
| 18:24.04 | lkcl | jkroon: as in, entirely bypassing the entire /etc/asterisk/*.conf system - all of it - and literally write say a 15-line program with a loop that routes and deals with everything |
| 18:24.05 | Qwell | Our fork, specifically. https://github.com/asterisk-org/starpy/ |
| 18:24.11 | lkcl | Qwell: ooo! exciting! |
| 18:24.19 | teloniusz | WIMPy: http://pastebin.ca/2180111 |
| 18:24.20 | lkcl | let me take a look at that.... |
| 18:24.43 | lkcl | Qwell: can you tell i might be an avid python programmer, eh? :) |
| 18:24.45 | WIMPy | lkcl: Like we did before Asterisk? |
| 18:25.39 | teloniusz | WIMPy: debug 2 set on the PRI and verbosity set to 5, but no debug on SIP |
| 18:26.26 | lkcl | WIMPy: :) |
| 18:27.36 | jkroon | sruffell, done, thanks a million. |
| 18:29.09 | WIMPy | teloniusz: I see one Connect acknowledge too much. What are you connected to? |
| 18:30.25 | teloniusz | WIMPy: do you mean SIP, or PRI? |
| 18:30.42 | WIMPy | The reaction to that is pretty bad anyway. But the real issue is at the other end. |
| 18:30.47 | WIMPy | PRI |
| 18:31.15 | *** join/#asterisk drknus (user@2600:3c03::f03c:91ff:feae:8a58) |
| 18:31.56 | WIMPy | Certainly it shouldn't use cause 16. |
| 18:31.59 | teloniusz | WIMPy: it's Sangoma A102 connected to a 2n GSM gate. But isn't it Asterisk who sends CONNECT ACK twice? |
| 18:32.57 | WIMPy | Oh, sorry. You're right. Looks like I looked at the wrong lines. |
| 18:33.42 | WIMPy | Or is it twice? Can you do it again with debug 2? |
| 18:34.38 | teloniusz | WIMPy: I set it up with 'pri set debug 2 span 1'... |
| 18:35.30 | teloniusz | WIMPy: still, it for sure looks like it's twice, but still - it's the same second... |
| 18:35.59 | WIMPy | Damn. I need some glasses today. |
| 18:36.20 | teloniusz | WIMPy: ...and I see only one connect.ack in the gate log |
| 18:36.26 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:37.21 | WIMPy | It might be only one, but rather confusing debug output. |
| 18:38.13 | WIMPy | It is only one. |
| 18:38.49 | *** join/#asterisk mvanbaak (~michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 18:38.55 | Qwell | yay |
| 18:38.56 | Qwell | mvanbaak: thanks |
| 18:39.02 | *** part/#asterisk mvanbaak (~michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 18:39.26 | lkcl | ahh, right. ok. got a bit of a NAT issue with ekiga and asterisk (now running on my server). |
| 18:39.34 | *** join/#asterisk navaismo (~navaismo@189.191.94.251) |
| 18:39.54 | lkcl | i'm connected via openvpn, so when i dial the demo number sip:1@10.8.0.1 it all works |
| 18:40.04 | lkcl | (and the confbridge as well, yay!) |
| 18:40.21 | lkcl | however if i dial the public address sip:1@217.147.94.29 .... no dice |
| 18:40.30 | WIMPy | teloniusz: There are these 3 lines about hangup in between that are for a similar callref. |
| 18:40.51 | lkcl | any suggestions, anyone, on how to solve this? |
| 18:41.47 | WIMPy | Or more likely it is the same callref, but displayed in another way. |
| 18:42.07 | teloniusz | WIMPy: which lines? |
| 18:42.11 | lkcl | hmmm... would "directmedia=no" help? |
| 18:42.30 | WIMPy | teloniusz: q931_hangup |
| 18:42.39 | teloniusz | 179-181, I see |
| 18:43.06 | teloniusz | it's all call ref 49261 |
| 18:43.59 | WIMPy | It's 0x406D vs 0xC06D, so the top bit which shouldn't really exist. |
| 18:45.25 | teloniusz | WIMPy: I'll look if I accidentally haven't included another call into log... |
| 18:45.40 | WIMPy | I don't think so. |
| 18:45.59 | WIMPy | But BTW: You are aware that you called nowhere? |
| 18:47.21 | WIMPy | wonders why there's a connect at all. |
| 18:47.23 | teloniusz | WIMPy: why? Is the number format wrong? |
| 18:47.40 | WIMPy | It doesn't exist at all. |
| 18:47.56 | *** join/#asterisk b0ot (~tmccurdy@147.177.41.53) |
| 18:48.04 | teloniusz | ee... |
| 18:48.18 | WIMPy | Err, no I see it. Looks like domething is going wrong here. |
| 18:48.20 | b0ot | asterisk fully supports H.323 correct? |
| 18:48.40 | WIMPy | Maybe a wrapped line? |
| 18:48.50 | Qwell | b0ot: depends on your definition of "fully" |
| 18:48.59 | WIMPy | Yes, that's it. |
| 18:49.59 | WIMPy | Dangers of copy&paste. |
| 18:50.25 | teloniusz | yeah |
| 18:53.09 | *** join/#asterisk Leddy (leddy@krypton.evosurge.com) |
| 18:55.56 | teloniusz | WIMPy: any idea how to extend these debug settings to better know what's going on? |
| 18:56.15 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-ldxeacrgmncoetmi) |
| 18:56.15 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 18:56.33 | WIMPy | YOu had bot debug and verbose turned up? |
| 18:57.04 | teloniusz | hmm |
| 18:57.11 | teloniusz | debug, not neccessarily |
| 18:57.24 | teloniusz | i thought that debug is an alias for verbose |
| 18:57.35 | WIMPy | Nope |
| 18:57.38 | teloniusz | core set debug on? |
| 18:57.40 | WIMPy | Two different things. |
| 18:58.02 | WIMPy | core set debug 9 |
| 18:58.14 | teloniusz | ok, I've set up sip debug and core debug to 5 |
| 18:58.32 | teloniusz | ok, let it be 9 |
| 18:58.39 | teloniusz | now I have to wait :> |
| 18:59.11 | WIMPy | Probably makes no difference. |
| 19:06.21 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 19:07.17 | *** join/#asterisk voxter (~hardcore@70-36-63-61.dyn.novuscom.net) |
| 19:07.44 | voxter | anyone know, is there a way to make the qualify option actually check the username in question, as opposed to merely treating any reply to an OPTIONS packet as OK? |
| 19:08.19 | voxter | having an issue where sip phones have their reg changed, but retain the same source port, and asterisk keeps checking the old extension, and deems it to be OK/online when that extension is no longer in use. |
| 19:08.54 | WIMPy | That's just the way it is. |
| 19:09.43 | teloniusz | WIMPy: ok, thanks, I've got to go for now; I'll probably be back for more questions when my logs fill up :> |
| 19:12.27 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:12.36 | *** join/#asterisk Graiden (~Graiden@rrcs-97-76-17-221.se.biz.rr.com) |
| 19:12.57 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:13.08 | jkroon | hi guys, i'm considering using XMPP PubSub, is it possible to somehow partition what device states is shared via each client? |
| 19:17.28 | leifmadsen | jkroon: there is not |
| 19:18.32 | jkroon | leifmadsen, ok, that's unfortunate. |
| 19:21.40 | anonymouz666 | jkroon: everything will be shared, including dahdi channels, local channels and others. |
| 19:22.07 | anonymouz666 | I mean, the device state of these channels. |
| 19:22.39 | jkroon | anonymouz666, useful for same-client machines, not so nice for a single machine with multiple clients |
| 19:27.13 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 19:45.47 | *** join/#asterisk navaismo (~navaismo@189.191.94.251) |
| 19:54.30 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 19:56.02 | *** join/#asterisk aossama (~ao@41.232.120.128) |
| 19:56.36 | TheCompWiz | I've run into a strange situation... anyone know why asterisk logs would jump backwards 6-7 minutes? i.e. logs show events from 05:59:43, 05:59:51, 05:59:54, then 05:52:51. |
| 19:57.00 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 19:57.21 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-oaljklrgqgaauzhk) |
| 19:57.36 | ChannelZ | I haven't looked at v11 yet though I saw gtalk/jabber has been refreshed, is there a means to change status of your accounts from the dialplan (Available, Away..) |
| 19:57.46 | TheCompWiz | there is no NTP client on this box... (it can't even access any NTP servers that might exist... due to firewall rules) nor was the system time changed. |
| 19:59.04 | SerajewelKS | what is the default configuration for GSM when used with Record()? I have some output sound files, but there is no header to convey bitrate etc. information that i can use to decode the file. |
| 20:00.17 | WIMPy | GSM 06.10 |
| 20:00.28 | WIMPy | All others require a licence. |
| 20:01.22 | SerajewelKS | VLC detects that the files are GSM, but it plays back as garbage |
| 20:03.43 | Graiden | Backjumping logs... that's a new one. |
| 20:04.39 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 20:06.17 | ChannelZ | What logs? The CDR logs, or general, or... |
| 20:06.39 | JasonL | I know this is off topic, but has anyone ever noticed an issue with Polycom IP phones, not passing through network to the PC port? We connected 5 phones at a customer site and none of the PC's work behind the phones, they don't obtain an IP from the DHCP server. Connect the PC direct and it works fine. Their switch is unmanaged therefore we don't think it's a voice vlan issue. |
| 20:08.08 | navaismo | Sphinx is a eal solution to use with asterisk? |
| 20:09.07 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 20:09.09 | wcselby | o/ |
| 20:09.25 | Graiden | JasonL: I've never seen that in bulk. Bad switch ports of course, but never like that. Do they have their own DHCP forwarder maybe? |
| 20:09.36 | wcselby | i've got an issue with a client where their MOH is restarting every 8 seconds |
| 20:10.19 | wcselby | everytime it restarts, I get the whole "Remote UNIX connection / disconnected" thing |
| 20:10.24 | wcselby | i'm not sure where to troubleshoot |
| 20:10.50 | lorsungcu | that's all that displays in the clu? |
| 20:10.52 | lorsungcu | u |
| 20:10.53 | lorsungcu | i |
| 20:11.03 | wcselby | this client is not using any kind of GUI |
| 20:11.14 | lorsungcu | the command line |
| 20:11.19 | Chainsaw | wcselby: That normally suggests an asterisk -rx call. How are you generating the MOH? |
| 20:11.24 | JasonL | Graiden: I'm not sure about DHCP forwarder. I know that those phones PC port work fine with another dhcp server |
| 20:11.43 | wcselby | the only thing that happens in the CLI when the music restarts is the remote unix connection / remote unix connection disconnected call. |
| 20:11.48 | wcselby | moh is called using mpg123 |
| 20:11.54 | wcselby | it's an mp3 file |
| 20:12.22 | wcselby | "application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 -s --mono" |
| 20:12.50 | Graiden | JasonL: Wonky, truly. I've run into DHCP passing issues before when using a DD-WRT WDS platform, but nothing on a simple network like that. I'd say try a full reset on one of the devices, if that fixes it, you know it was in your config somewhere. |
| 20:13.55 | lorsungcu | JasonL is there a link at all; can you manuall set IP and pass traffic, or does the PC not even see that a cable is plugged in? |
| 20:27.03 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-18.nwrknj.fios.verizon.net) |
| 20:29.57 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 20:30.45 | JasonL | Graiden: Thanks.. Will try that. |
| 20:30.57 | JasonL | lorsungcu: I will try that first see what happens, thanks for the help |
| 20:31.09 | Graiden | I was just gonna say... check lorsungcu's solution first lol |
| 20:31.12 | Graiden | Good luck! |
| 20:31.15 | JasonL | hehe |
| 20:31.17 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 20:33.59 | *** part/#asterisk trumee (~parul@188-222-165-248.zone13.bethere.co.uk) |
| 20:36.13 | *** join/#asterisk darkdrgn2k (~darkdrgn@209.90.253.66) |
| 20:36.31 | darkdrgn2k | Can asterisk be used as a Back to Back proxy, and if so any infstructions out there on how to do it? |
| 20:37.02 | WIMPy | ~b2bua |
| 20:37.02 | infobot | it has been said that b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 20:37.34 | WIMPy | That's what Asterisk is. |
| 20:39.43 | *** join/#asterisk jsjc (~Adium@237.157.78.188.dynamic.jazztel.es) |
| 20:42.43 | leifmadsen | there is basically B2BUA and proxies -- separate entities |
| 20:43.08 | *** part/#asterisk jsjc (~Adium@237.157.78.188.dynamic.jazztel.es) |
| 20:43.50 | lkcl | ok - apologies but i'm having difficulty getting ekiga to talk UDP packets to my server, so i'm going to try adding in SIP registration. can someone advise how i can add a single user+password in, somewhere into asterisk config files, to turn asterisk into a SIP "registrar"? |
| 20:44.27 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 20:44.47 | jpsharp | register devices to Asterisk? Put the entry in /etc/asterisk/sip.conf |
| 20:45.31 | lkcl | jpsharp: thanks for responding. ok, that's probably enough of a hint - thanks |
| 20:46.11 | lkcl | jpsharp: ok - type=user and all that, i think i get it |
| 20:47.16 | teloniusz | WIMPy: OK, I've got something. It seems that the connection is being dropped by SIP peer... |
| 20:47.43 | leifmadsen | well, type=user wouldn't allow a registration |
| 20:47.55 | leifmadsen | you use type=peer or friend to allow registration |
| 20:48.00 | WIMPy | teloniusz: At least it must be fast at doing so :-) |
| 20:48.00 | teloniusz | WIMPy: ...but there's strange thing: I tried to set debug on this peer and there it goes: |
| 20:48.03 | leifmadsen | lkcl: www.asteriskdocs.org |
| 20:48.23 | teloniusz | WIMPy: channel.c: Didn't get a frame from channel: SIP/7090264402-0000e9f3 |
| 20:48.42 | teloniusz | WIMPy: and a soft changup right after that. |
| 20:48.53 | teloniusz | hangup even |
| 20:49.21 | lkcl | leifmadsen: taaaaa |
| 20:50.14 | WIMPy | Sounds like a strange time. |
| 21:00.52 | TheCompWiz | Graiden & ChannelZ: I agree its strange... but I'm seeing it on all asterisk logs. the system's syslog shows no such discrepancies and no errors at or around the time this happens. I'm seeing it on about a dozen different servers (3 different sets of hardware... but 12 servers total) (sorry for the delayed response) |
| 21:01.01 | *** join/#asterisk jsjc (~Adium@237.157.78.188.dynamic.jazztel.es) |
| 21:01.23 | TheCompWiz | (asterisk 1.6 fwiw) |
| 21:01.56 | *** part/#asterisk jsjc (~Adium@237.157.78.188.dynamic.jazztel.es) |
| 21:03.01 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 21:04.49 | jaytee | anyone here used Cisco 7960 phones with sip firmware? |
| 21:06.22 | pabelanger | Likely not, I don't think Cisco allows you to distribute it |
| 21:06.25 | Nugget | jaytee: Just Say No. |
| 21:06.27 | pabelanger | unless they changed it |
| 21:06.39 | ChannelZ | Hmm. I never look at my logs that often to be honest |
| 21:06.45 | _Corey_ | uh, yeah... i've used the sip firmware for a long time |
| 21:06.52 | ChannelZ | But it's not something I've seen when I have |
| 21:06.55 | leifmadsen | I have used it a long time ago |
| 21:07.03 | Nugget | I've still got a few 40s and 60s laying around, but I don't use them any more. It's just too much of a pain to use them alongside the more modern x5 and x1 phones |
| 21:07.07 | leifmadsen | 7960 is certainly easier to configure than the 7961 |
| 21:07.15 | _Corey_ | It sucks and Cisco stopped supporting it, but it works |
| 21:07.16 | Nugget | the 40/60 phones are just way too different |
| 21:07.23 | leifmadsen | Nugget: that :) |
| 21:07.33 | leifmadsen | I love the tftp only feature |
| 21:07.41 | leifmadsen | why think when you have tftp provisioning! |
| 21:07.57 | _Corey_ | I have a customer who curses me with them... every time I think I'm going to be done with them they buy some company and end up with another 200+ of the stupid things |
| 21:07.59 | TheCompWiz | ChannelZ: I know the feeling. Unfortunately, I'm not sure what the *real* problem is... but asterisk seems to fork off into a bunch of processes & become non-responsive... but nothing in the logs is giving me any help... except for that one discrepancy. |
| 21:08.12 | jaytee | I have 3 of these and they already have the SIP firmware. I'm just trying to find more info than what is on voip-info.org |
| 21:08.15 | Nugget | using any cisco phones with asterisk is an exercise in masochism, though |
| 21:08.33 | Nugget | if they didn't look so stunningly awesome on everyone's desks I'd replace them with digium phoness :) |
| 21:08.37 | jaytee | rather than type each .cnf file out by hand from scratch. |
| 21:09.27 | _Corey_ | Nugget: stunningly awesome? lol... I used to think so. I guess it wore off |
| 21:09.43 | Nugget | well we do have a lot of the color screen ones :) |
| 21:13.44 | leifmadsen | Nugget: so you have a working .cnf file for the x1 phones? |
| 21:14.04 | *** join/#asterisk GameGamer43 (users.5533@id-5533.hampstead.irccloud.com) |
| 21:14.07 | leifmadsen | I've never seen someone actually get an x1 phone working with asterisk :) |
| 21:14.12 | _Corey_ | leifmadsen: We have them |
| 21:14.19 | _Corey_ | too f'ing many of them |
| 21:14.21 | Nugget | yeah, I run a mix of 41s and 61s and 45s and 65s |
| 21:15.01 | _Corey_ | the xml file is a lot more convoluted but it works ok |
| 21:17.03 | leifmadsen | ya for sure -- I never could get it working, and there was zero documentation when I tried to get my 7970 working. |
| 21:17.08 | kikohnl | I have a 7960 and 7941 both with current SIP versions running, minor issues to get them up |
| 21:17.10 | leifmadsen | would be cool if someone posted a reasonable example somewhere |
| 21:17.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:17.33 | _Corey_ | leifmadsen: I can PB one if you want to see it |
| 21:18.07 | leifmadsen | sure would be interested. I could blog about it I guess |
| 21:18.41 | jpsharp | I ran several hundred 7940s at my previous employer. |
| 21:19.55 | jaytee | is there anywhere I can download a sample .cnf file for the 7960? searched Cisco but so far found nothing for config downloads. |
| 21:22.15 | darkdrgn2k | any on ehave any experiances with an 1120e on asterisk |
| 21:24.20 | _Corey_ | leifmadsen: enjoy... http://pastebin.com/k50qMWjf |
| 21:25.12 | *** join/#asterisk aossama (~ao@41.232.127.75) |
| 21:25.15 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 21:25.43 | kikohnl | 7941 config http://pastebin.com/1UkDbp7f |
| 21:26.19 | puzzled | I had my fair share of problems with 79x1s too. nuggets like an unpublished 12(?) char limit on certain fields were "fun". just avoid them unless you use CCM in which case they work very well |
| 21:27.41 | _Corey_ | jaytee: here you go... 7940/7960 sample file : http://pastebin.com/JZ9Hy9mc |
| 21:27.52 | _Corey_ | you also need a SIPDefault.cnf for that one |
| 21:27.52 | jaytee | thanks, Corey!!! |
| 21:28.10 | _Corey_ | let me know if you need that too while I have this open |
| 21:30.01 | jaytee | sure, it would definitely help |
| 21:31.50 | _Corey_ | jaytee: np, ... here you go: http://pastebin.com/nhrYVZzg |
| 21:32.01 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 21:32.49 | kikohnl | 7960 sipdefault.cnf http://pastebin.com/Q3bKB8Lb |
| 21:34.10 | kikohnl | 7960 SIPmacaddress.cnf http://pastebin.com/eb2bqXzY |
| 21:37.34 | jaytee | kikohnl, thank you. between your files and Corey's I should be able to figure this out and get one of these working. |
| 21:39.14 | sruffell | <PROTECTED> |
| 21:47.22 | *** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:8c4c:ca87:2fc8:2a46) |
| 21:47.41 | *** join/#asterisk tompaw (~tompaw@tompaw.xxx) |
| 21:47.44 | tompaw | Hi. |
| 21:48.14 | tompaw | I might be missing something obvious here, but it's been a really long day and I would appreciate your help. |
| 21:48.36 | jpsharp | Well, what's the issue? |
| 21:48.51 | tompaw | on 1.8.15, got two G711 peers, A and B. When they both do an echotest separately, it works great. |
| 21:49.20 | tompaw | But when I do A<>*<>B call, then the audio from A->B kicks in after 30 seconds! |
| 21:49.26 | tompaw | No sooner, no later, 30 seconds. |
| 21:49.46 | tompaw | B->A works right away. |
| 21:49.58 | tompaw | Traces look normal, no errors in logs, etc. |
| 21:51.10 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 21:57.57 | luckman212_ | tompaw: rtp set debug on |
| 21:58.38 | tompaw | ok |
| 21:59.56 | tompaw | luckman212_: got 5x "Sent RTP P2P packet" to A and 4x "Sent RTP P2P packet" to B the moment call is answered |
| 22:00.15 | tompaw | And then nothing, when the A->B audio starts working, nothing new appears in the debug log. |
| 22:01.29 | luckman212_ | ok well then rtp is not flowing thru your asterisk box |
| 22:01.42 | tompaw | are you saying I'm not proxying the call? |
| 22:01.44 | luckman212_ | are these 2 devices both on the same network? |
| 22:01.44 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 22:01.56 | luckman212_ | yes your asterisk has removed itself (or tried to) from the media path |
| 22:02.05 | tompaw | no, they're billion kms aways from each other and behind thousand NATs |
| 22:02.08 | tompaw | that would explain a lot |
| 22:02.11 | luckman212_ | ah yes! |
| 22:02.21 | luckman212_ | you need canreinvite=no |
| 22:02.24 | tompaw | any idea how to force my dear * to stay in the middle? |
| 22:02.30 | tompaw | DAMN! |
| 22:02.37 | tompaw | I'm gonna shoot myself with a caffeine shot. |
| 22:03.15 | luckman212_ | if you're running 1.8+ you can use directmedia=no I believe that is the preferred method |
| 22:04.12 | luckman212_ | if you've got specific peers that are behind nat, and others that aren't you can define subnets to permit/deny via directmediadeny=w.x.y.z and directmediapermit=a.b.c.d etc etc |
| 22:04.31 | tompaw | I always want it to stay in the middle! |
| 22:04.59 | tompaw | I totally forgot about that option, my mind was poisoned with wholesale switches and freeswitch and 20 hours of no sleep! |
| 22:05.00 | luckman212_ | tompaw: why? |
| 22:05.01 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 22:06.05 | tompaw | luckman212_: cause it's gonna be transcoding G711<>G729 and the agents will be using 3g-based, voip-blocked, triple-nat'd internets probably. |
| 22:06.27 | *** join/#asterisk hoho` (~hoho@unaffiliated/hoho/x-4898770) |
| 22:07.14 | hoho` | hello |
| 22:07.30 | hoho` | is there any way to configure a mgcp gateway as a trunk? |
| 22:07.44 | luckman212_ | tompaw: ok, also asterisk would always keep itself in the media path if it needs to transcode ... so you may have other issues going on. it's also quite strange that you hear audio after 30 seconds... I would expect it to either work right away or not at all. unless the NAT device at one or both ends is doing something really 'ugly' |
| 22:08.00 | tompaw | luckman212_: in this one particular test there was no transcoding |
| 22:08.07 | tompaw | that explains everything |
| 22:08.10 | luckman212_ | yep |
| 22:08.12 | tompaw | with that option provlem is solved |
| 22:08.31 | tompaw | I actually thought that new Wireshark is somehow supersmart and hides them RTP packets |
| 22:08.38 | tompaw | now I know it wasn't lol |
| 22:09.43 | WIMPy | It does what you tell it to do. |
| 22:10.58 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 22:12.20 | tompaw | thanks for your help mate, I can now sleep in peace. |
| 22:12.44 | WIMPy | Better than rest in peace, I guess. |
| 22:13.03 | tompaw | better than rest in pieces. |
| 22:13.05 | hoho` | lol |
| 22:13.27 | WIMPy | right |
| 22:27.17 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 22:58.20 | *** join/#asterisk lkcl (~lkcl@host217-43-149-4.range217-43.btcentralplus.com) |
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| 23:09.05 | lkcl | ok, with 2 clients (ekiga and xmeeting) i have audio working perfectly with app_confbridge: how would i go about getting video up-and-running, and/or testing to debug why it's not? |
| 23:11.55 | *** part/#asterisk newtonr (~newtonr@nat/digium/x-ldxeacrgmncoetmi) |
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| 23:38.56 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 23:57.07 | lkcl | got it! sip.conf - videosupport=yes. yay! |