| 00:03.11 | *** join/#asterisk mindCrime (~chatzilla@64.134.184.148) |
| 00:06.55 | WIMPy | Ouch. |
| 00:08.23 | WIMPy | With only two participants in a ConfBridge, and one of them puts the connection on hold, it propagates to the other participant. |
| 00:08.52 | WIMPy | If a third participant enters, the 2nd will continue to her MOH, but can be heard by others. |
| 00:21.10 | irroot | WIMPy you not sleepin ... |
| 00:22.21 | *** join/#asterisk Ean (ean@unaffiliated/ean) |
| 00:23.11 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:23.35 | WIMPy | Sleep? |
| 00:23.37 | WIMPy | What is that? |
| 00:25.20 | WIMPy | And what about yourself? |
| 00:26.20 | nny | Gotoif always messes with me |
| 00:26.29 | nny | exten => s,n,Gotoif(["${agentresponse}" = "2"]?s,1) |
| 00:26.35 | nny | what should happen if I press 1? |
| 00:26.42 | WIMPy | is pretty sure, it's the other way round. |
| 00:26.48 | nny | hahaha |
| 00:26.49 | nny | indeed |
| 00:27.02 | WIMPy | Remove the spaces. |
| 00:27.07 | nny | k |
| 00:27.39 | nny | NoOp("SIP/9900-0000099f", "1") in new stack -- Executing [s@dialer-agent-login:8] GotoIf("SIP/9900-0000099f", "[""="2"]?s,1") in new stack |
| 00:28.18 | nny | exten => s,n,Read(agentreponse,if-this-is-correct-press&1&otherwise-press&2,1) is what I use to get there |
| 00:29.14 | WIMPy | Is your DTMF working correctly? |
| 00:29.40 | nny | yeah the NoOP is from exten => s,n,NoOp(${agentreponse}) |
| 00:29.53 | nny | so it sees 1, but my expression syntax is broken |
| 00:29.58 | nny | reading up |
| 00:30.43 | *** join/#asterisk cookieliu (~cookie@27.115.97.6) |
| 00:32.46 | nny | i tried without quotes, but my variable isn't presented correctly |
| 00:32.54 | irroot | WIMPy in bit more pain than usual so online distraction |
| 00:33.08 | nny | ahh |
| 00:33.10 | nny | :< |
| 00:33.13 | WIMPy | nny: Quotes are ok like that. |
| 00:34.19 | irroot | nny <EXPR>?<true>:<false> |
| 00:34.21 | *** join/#asterisk lwizardl (~james@c-68-62-80-172.hsd1.mi.comcast.net) |
| 00:34.39 | nny | yeah my <EXPR> is off |
| 00:34.47 | Tim_Toady | nny: its atypo, u save it as agentreponse but u try to read the variable agentresponse |
| 00:34.56 | Tim_Toady | its a typo* |
| 00:35.00 | nny | omg |
| 00:35.03 | nny | thanks hahaha |
| 00:35.08 | nny | been at this all day, brain frying |
| 00:35.29 | nny | Tim_Toady: thanks |
| 00:35.35 | irroot | Tim_Toady good spot |
| 00:35.36 | nny | irroot: WIMPy thanks as well |
| 00:35.43 | nny | reponse.. :S |
| 00:36.13 | WIMPy | What does DVT and PQ mean in Jira? |
| 00:38.42 | pabelanger | WIMPy: URL? |
| 00:39.16 | WIMPy | When creating an issue. "found by" |
| 00:40.26 | pabelanger | WIMPy: PQ is a department in Digium |
| 00:40.28 | lwizardl | guys I was wondering a few things hopefully some here knows the answers :) 1) If I was to have an analog phone line connected to a asterisk box would that be a FXO card needed? 2) Is there a plugin to take analog faxes and save them as PDF files ? 3) What audio formats does asterisk support for hold tones ? |
| 00:41.05 | lwizardl | hold tones as in music for people on hold |
| 00:42.28 | irroot | lwizardl 1)Yes 2) Yes 3) MP3/Wav/Streaming/CD/Aux ie radio any 20ms 8000hz signed lin audio or transcodeable format |
| 00:44.05 | lwizardl | irroot, what would the plug in be called? also is there a way to be notified of the fax say like being emailed by the system. |
| 00:44.48 | pabelanger | WIMPy: created an issue to remove it, it does not providing any additional information for Asterisk issues |
| 00:45.02 | lwizardl | so for the hold music I can't just take say some mp3 files I ripped from my cds and transfer them to the box. |
| 00:45.35 | irroot | lwizardl either app_fax or res_fax using 1.8 recomend res_fax will store file as tiff you can convert it and email it using utilities on the system and the dialplan |
| 00:45.55 | WIMPy | pabelanger: Ok, I guess I'll use 'customer' then. Can't be wrong. |
| 00:45.58 | irroot | lwizardl yes you can |
| 00:46.56 | irroot | has been real busy of late with a project plan to get through some genral * stuff and go through jira soon |
| 00:47.10 | lwizardl | irroot, awesome any settings I need to match when I make the rips or just use the same thing I would use for my ipod |
| 00:48.51 | irroot | there is info online it will be less quality than for ipod unless you using HD but then id recomend asterisk 10 |
| 00:49.41 | irroot | otherwise sample rate is 8000 single chan |
| 00:50.46 | lwizardl | but using that sample rate will it sound like it should |
| 00:51.54 | Tim_Toady | irroot: so 10.0 supports sample rates beyond 8Khz? even inetrnally, eg using ast_streamfile() ? |
| 00:54.18 | irroot | Tim_Toady slin16 has been worked on but cant find all the details now |
| 00:54.58 | Tim_Toady | ok, i ll check the 10.0 source |
| 00:55.00 | irroot | lwizardl std phone line uses this so cant sound better over line ... |
| 00:55.28 | irroot | Tim_Toady use the source luke ... i dont have a copy near me ATM sorry |
| 00:55.39 | Tim_Toady | :> |
| 00:56.06 | lwizardl | yeah I just wasn't sure if say the songs would end up sounding like nes games versus the cd |
| 00:56.20 | lwizardl | thanks again irroot |
| 00:57.48 | nny | um gravity check |
| 00:57.49 | nny | http://pastebin.com/1ntvWyuz |
| 00:58.06 | nny | exten => s,n,Gotoif([${number}=9]?5) |
| 00:58.09 | nny | is wrong for some reason |
| 00:58.39 | WIMPy | '$[' missing |
| 00:58.44 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 00:58.54 | WIMPy | Err. only the $ |
| 00:59.40 | Janos | so my previous question again, does anyone knows if there is more than one reason for this notice to be in my log file "chan_sip.c: Peer 'xxxx' is now Lagged. (2016ms / 2000ms)" or can i conclude that at that moment a ping between the asterisk and the peer was taking more than 2000ms ? |
| 01:00.08 | WIMPy | Janos: Exactely that. |
| 01:00.31 | Tim_Toady | a sip notify response actually not an ICMP ping |
| 01:00.46 | WIMPy | yes |
| 01:02.31 | irroot | Tim_Toady yip if there is a delay in asterisk or a problem with the phone a ping can be near 0 but the poke can be high |
| 01:03.39 | irroot | ICMP is kernel level where the "poke" is at user level in a application specifically in one thread of a application in chan_sip |
| 01:03.45 | Janos | ok so this does not exactly means that the network is the cuase |
| 01:03.58 | Janos | s/cuase/cause |
| 01:04.01 | WIMPy | As Tim_Toady said, it's a SIP thing, so it's from application to application. |
| 01:05.02 | Janos | what would happen if in the middle of a call a peer becomes lagged ? would this be a undefined behavior ? |
| 01:05.07 | irroot | but clearly it uses the network so there is the possibility on network involvment |
| 01:05.27 | Janos | yeah, can't discard that quite yet |
| 01:05.40 | WIMPy | That would not matter. |
| 01:06.08 | WIMPy | You can even continue an existing call to a peer that becomes unreachable. |
| 01:06.21 | Janos | i'll just leave an mtr running through the night and see the results tomorrow about the network |
| 01:06.30 | irroot | Janos this is a registration thing so if in a call a phone is unregistered the call will flow as its active and this is typically in RTP if the rtp stops audio stops |
| 01:06.31 | WIMPy | But you can't place a new call there. |
| 01:06.39 | nny | WIMPy: thanks btw, that was it.. just a little slow this evening |
| 01:06.57 | nny | is there a way to plan announce sounds to only one side of a conference room? |
| 01:07.48 | Janos | ok if a call is already working it should keep working even if a peer becomes unreachable, but i would not be able to place another call, that's good to know |
| 01:08.00 | WIMPy | No. YOu could opnly take that party out of the conf, play something and put them back. |
| 01:08.16 | nny | hmm maybe I can announce something to the channel directly before they enter |
| 01:08.32 | nny | what you said heh |
| 01:08.54 | *** join/#asterisk MefhigosetH (~Mefhigose@190.179.34.122) |
| 01:09.03 | WIMPy | Janos: However, if ou have session-timers enabled, the call will be terminated if the related packets aren't processed, either. |
| 01:09.28 | Janos | googles asterisk session timer |
| 01:10.02 | WIMPy | It's a session related keep-alive at SIP level. |
| 01:10.20 | nny | WIMPy: i'll dig into it tomorrow. I wanna tell the first person in the conference another one joined, but not alert the second. :S |
| 01:10.29 | Janos | yeah i'm trying to find out how to check if i have them :P |
| 01:10.45 | nny | a second one joined* |
| 01:11.06 | nny | could use chanwhisper or something, but have to know the channel variable. May be able to do something with that |
| 01:11.20 | irroot | one thing to look out for is calls cutting after 32s if this happens the phones replies are not coming in and asterisk is cutting the call |
| 01:11.52 | WIMPy | nny: No, CanSpy doesn't work with things like Playback. But you could just put it into the conferences MOH. |
| 01:12.43 | WIMPy | Well, th last one leaving will get that as well, but maybe that's ok. |
| 01:14.18 | lwizardl | ok so let me see if I truly understand this or not. I'm guessing I am off base a tad bit. If I wanted to have 1 analog fax line I would need a fxo module, and if I wanted to have 2 separate VOIP phone number lines I would need 2 FXS modules |
| 01:14.33 | Janos | yeah well the issue i have right now is some calls are getting established but no sound in any direction, so far is completely random and it started around a week ago, asterisk process had around 40 weeks uptime, i restarted it yesterday but problem continues and the only weird thing i see on the logs are those Lagged notices |
| 01:15.04 | Tim_Toady | lwizardl: yes |
| 01:15.19 | WIMPy | nny: Define a hint for the conf and check the devicestate before entering. |
| 01:15.40 | lwizardl | Tim_Toady, thanks :) |
| 01:15.49 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-171-223.hsd1.va.comcast.net) |
| 01:16.41 | WIMPy | Janos. At least from Asterisks point of view they are not related. But they might have a common cause. |
| 01:17.36 | WIMPy | lwizardl: Apart from the fact that you don;t need to have a 1:1 relationship between numbers and lines, yes. |
| 01:19.03 | lwizardl | WIMPy, ok what I was looking to do was have the PBX system handle 2 businesses phone lines in the same business and then have each business with a few lines each more for like call waiting etc |
| 01:20.04 | lwizardl | WIMPy, trying to decide on the hardware that I need to buy now and then gather the media files I need while I wait for shipping to get here. |
| 01:22.06 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 01:22.08 | WIMPy | lwizardl: Well, one FXO port per POTS line you want to connect and one FXS port per steampunk phone you want to connect. |
| 01:23.24 | lwizardl | POTS as in traditional analog phone lines, and steampunk ? |
| 01:23.55 | WIMPy | Well, the technology of the 18th century :-) |
| 01:24.22 | lwizardl | lol yeah like Ma bell installed |
| 01:26.40 | lwizardl | do the phone line splitters work with asterisk voip ? the ones that let you place two phones on the same lines. Like one in the living room and one in the bedroom |
| 01:26.43 | lwizardl | etc |
| 01:28.41 | WIMPy | Why would you want do so if you have somethign where you can connect them individually? |
| 01:29.30 | lwizardl | well what I was wanting to do was have a phone in both the main floor of the business and also in the backroom |
| 01:29.50 | irroot | lwizardl use SIP phones |
| 01:30.08 | lwizardl | so I don't have to run to the floor if I am in the backroom sorting or doing a repair |
| 01:30.16 | lwizardl | irroot, I thought about that |
| 01:30.17 | irroot | if you use digium card get 8port with 4FXO / 4FXS and you sorted |
| 01:30.45 | *** join/#asterisk mindCrime (~chatzilla@cpe-076-182-089-009.nc.res.rr.com) |
| 01:31.05 | irroot | a simple solution is DECT phone have more than one phone on base station |
| 01:31.17 | irroot | can transfer it from phone to phone |
| 01:31.19 | lwizardl | irroot, not sure if this is an authorized seller or not but one like this http://www.8774e4voip.com/Digium_AEX800E_Asterisk_Trixbox_and_Switchvox_p/digium%20aex800e.htm |
| 01:31.25 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 01:31.53 | lwizardl | irroot, yeah was looking at that. I know I have a few of the uniden phones that have 5 phone cordless phones |
| 01:31.53 | nny | dammit lol |
| 01:32.02 | WIMPy | Yes, with DECT, you don't even need anything else to transfer calls. |
| 01:32.04 | nny | seems I can't run a command after a user exits via Meetme |
| 01:32.23 | irroot | lwizardl exactly that |
| 01:32.29 | nny | would a local channel work? |
| 01:33.01 | lwizardl | irroot, yeah just wasn't sure if it was a clone/pirate card seller or not |
| 01:33.46 | irroot | lwizardl cant comment on the provider im on the bottom end of the world |
| 01:34.35 | irroot | the green is FXS red FXO can see www.digium.com for the original site / docs |
| 01:34.35 | *** join/#asterisk BuenGenio (~BuenGenio@059148208218.ctinets.com) |
| 01:34.43 | dan__t | I don't quite like festival. |
| 01:34.45 | dan__t | At all, actually. |
| 01:34.46 | lwizardl | irroot, yeah, I think I understand you. I'm a very small business owner |
| 01:35.27 | dan__t | I purchased a Cepstral starter kit in '09 and I'm waiting to find out if I'm still entitled to that license heh |
| 01:36.34 | dan__t | Any other options out there for TTS? |
| 01:37.10 | Tim_Toady | epseak and flite as 3rd party plugins for asterisk |
| 01:38.12 | WIMPy | dan__t: Send the text via instant messanger to a call center in india. |
| 01:38.19 | WIMPy | scnr |
| 01:38.31 | dan__t | heh |
| 01:39.24 | dan__t | Weird. I call on my sip phone and I get what I expect, but I call from a land phone which goes through my SIP provider, the dialplan dies right before it's about to Festival() |
| 01:41.50 | ChannelZ | Dying before the festival is tragic |
| 01:42.13 | dan__t | heh |
| 01:42.40 | dan__t | Well. Shit. |
| 01:42.49 | ChannelZ | shits |
| 01:43.25 | dan__t | :/ |
| 01:44.14 | dan__t | Is it an issue of no audio from the called party - asterisk? could it be my outbound sip is fubar? |
| 01:44.50 | ChannelZ | ? |
| 01:44.52 | *** join/#asterisk gxdssoft (~gxdssoft@190.235.47.121) |
| 01:44.56 | ChannelZ | You say it's "dying", how so? |
| 01:45.17 | WIMPy | If it's fubar, maybe it's better off dying? |
| 01:45.34 | *** part/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 01:46.00 | dan__t | <PROTECTED> |
| 01:46.09 | dan__t | pri 4 is a line that Festival(says stuff) |
| 01:46.24 | ChannelZ | Do you Answer? |
| 01:46.37 | dan__t | Yes. This is far deep in to my dialplan. |
| 01:46.41 | ChannelZ | And is this the exact same context/exten that works called the other way? |
| 01:46.48 | dan__t | Yes. |
| 01:47.06 | dan__t | I call a 480.xxx.xxxx number from my soft/sip phone on my desktop just to make it easy |
| 01:47.08 | ChannelZ | codec? I don't know much about Festival |
| 01:47.30 | dan__t | Uh, all I guess? I'm not sure. I register with all, if that counts? |
| 01:47.39 | dan__t | It was working last night(TM) |
| 01:48.04 | ChannelZ | Well you can either smack it then, or reboot. Those usually fix things. |
| 01:48.35 | dan__t | Right, so, I want to do things the right way these days. |
| 01:49.24 | dan__t | * connects to the Festival server, then immediately disconnects. If * is speaking one codec from the SIP provider and Asterisk spits something out that's in a format the codec doesn't understand, that could cause an explosion, right? |
| 01:49.29 | ChannelZ | Ah. In that case, you smack it and then say "I'm sorry" |
| 01:50.22 | dan__t | k |
| 01:50.27 | ChannelZ | Yeah, ish. Does Festival just make files that get played back? |
| 01:50.48 | dan__t | I don't have a cache dir set up, so no |
| 01:50.53 | dan__t | I mean it can, but in this instance it doesn't/ |
| 01:51.07 | ChannelZ | Or does it hook in there a little deeper.. I thought it would just spit out essentially ulaw (or raw) and * would do the transcoding, I was just spitting out things coming to mind as to why two different paths might be different |
| 01:51.33 | *** join/#asterisk OldMonk (~raju@122.163.131.89) |
| 01:51.34 | dan__t | Understood |
| 01:51.39 | OldMonk | hi |
| 01:52.09 | ChannelZ | hoqdy |
| 01:52.12 | ChannelZ | howdy even |
| 01:52.24 | dan__t | hi. |
| 01:52.30 | OldMonk | what would be the best way to call an extension then call another extension and have the conference together in asterisk 1.6? can you call out through a meetme room? |
| 01:52.42 | OldMonk | s/have the/have them/ |
| 01:53.14 | Tim_Toady | you must configure festival to output 8kHz wavs |
| 01:53.15 | WIMPy | You don;t have to. Just use originate. |
| 01:53.36 | OldMonk | WIMPy: will that conference two or more lines? |
| 01:53.45 | WIMPy | Two. |
| 01:53.55 | dan__t | hrm |
| 01:53.58 | WIMPy | If you want more, you will indeed need meetme or confbridge. |
| 01:54.23 | dan__t | I'll look in to it, thank you Tim_Toady. |
| 01:54.28 | dan__t | Not sure what it does by default? |
| 01:54.42 | OldMonk | the second part of the question then: will meetme allow a user to call out? |
| 01:54.51 | lwizardl | so what would be the most cost effective (aka cheapest and fully functional) way to have 1 analog only fax line, 2 seperate voip phone lines, and atleast 2 active lines per number (call waiting) with hold button. would that be using the pbx cards and modules or just using the sip phones |
| 01:54.51 | WIMPy | no |
| 01:54.51 | Tim_Toady | dan__t: im not sure either, havent used it for ages |
| 01:55.04 | dan__t | hehe |
| 01:55.28 | lwizardl | just wondering becuase I see stuff like these for anywhere from $20-90 each http://cgi.ebay.com/GXP2120-6-line-Executive-HD-IP-Voip-SIP-Phone-Asterisk-/160629197583?pt=LH_DefaultDomain_0&hash=item25663f0b0f |
| 01:55.32 | Tim_Toady | switched to flite which is supposed to be a lightweight version of festival but its more like festival done right |
| 01:56.29 | WIMPy | "$100.00 economy shipping" I like that. |
| 01:56.58 | lwizardl | WIMPy, yeah standard ebay sellers. sell items cheap and over charge on shipping |
| 01:57.13 | OldMonk | next, when i transfer a call, is there any way of figuring out in the new dialplan that this is a transfer? |
| 01:57.25 | *** join/#asterisk nix8n82 (~nate@24.143.28.16) |
| 01:57.35 | dan__t | Waiting to see if my Cepstral license is still valid :/ |
| 01:57.42 | WIMPy | OldMonk: Depends on how you do the transfer. |
| 01:57.51 | *** join/#asterisk drynish (~sabayonus@modemcable039.7-200-24.mc.videotron.ca) |
| 01:57.54 | drynish | Hello |
| 01:58.21 | OldMonk | WIMPy: to tell the truth, i don't really know -- the callers press the transfer button in Qutecom, dial the new number, and voila! it's transferred |
| 01:58.58 | WIMPy | That's probably just a new call then. |
| 01:59.03 | drynish | I have some echo with oslec |
| 01:59.16 | drynish | and checking on the internet I'm supposed to not have any |
| 01:59.21 | drynish | I don't know where to check ;) |
| 01:59.54 | OldMonk | WIMPy: is there another way? |
| 02:00.09 | OldMonk | oh, you mean have the dialplan effect the transfer? |
| 02:00.10 | WIMPy | Yes, DTMF transfers. |
| 02:00.29 | OldMonk | WIMPy: could you elaborate, how would i do that? |
| 02:00.43 | WIMPy | see features.conf |
| 02:00.46 | dan__t | Hm, definitely a festival thing. Asterisk knows when the festival server isn't around, as indicated by * complaining when festival server is not running. |
| 02:01.23 | dan__t | And a Playback() works. |
| 02:01.41 | dan__t | So I'm back to the same conclusion: Festival hates me. Past, present, and future |
| 02:03.05 | OldMonk | WIMPy: so i can use a blindxfer or atxfer... with that too, how do you figure out in the new extension that you reached through a transfer and not a direct dial? |
| 02:04.37 | OldMonk | oh, ${BLINDTRANSFER} -- will that work for attended transfers too? |
| 02:04.46 | WIMPy | There should be something in a variable, or in the channel function. |
| 02:05.08 | WIMPy | I haven't tried. |
| 02:05.51 | OldMonk | cool, i can play with this and check |
| 02:06.17 | OldMonk | WIMPy: incidentally, will originate conference a caller with two more channels? |
| 02:06.51 | WIMPy | No, just two channels. |
| 02:07.08 | OldMonk | WIMPy: making three in all? caller + 2 channels? |
| 02:07.24 | WIMPy | No. Two channels in total. |
| 02:07.37 | OldMonk | oh, not a conference at all then |
| 02:08.10 | WIMPy | no |
| 02:08.31 | WIMPy | >2 channels always requires an application to do so. |
| 02:08.54 | ChannelZ | some actual SIP phones have their own 3-way conf built-in |
| 02:09.06 | OldMonk | ChannelZ: yeah, qutecom has that |
| 02:09.51 | WIMPy | Yes, I had to use that, because of a bug I discovered that left the 2nd participant deaf. |
| 02:09.53 | OldMonk | but even that has limitations: you have to first select conf then make the calls; you cannot make a call, then select conf and conf it with another call |
| 02:10.12 | WIMPy | Err, what? |
| 02:10.23 | ChannelZ | Hmm. My Linksys SPAs let you shove someone on hold, dial someone else, and then hit 'conf' to bring them together. |
| 02:10.39 | ChannelZ | them/all of oyu |
| 02:10.44 | ChannelZ | s/oyu/you/ |
| 02:11.16 | OldMonk | WIMPy: you cannot conf on the fly in qutecom: you have to first select conference, then dial the two extensions that you want to conference together |
| 02:11.20 | WIMPy | recursive search&replace. Nice :-) |
| 02:11.38 | OldMonk | hehheh |
| 02:11.44 | WIMPy | Ah, ok. so that was not about Asterisk. |
| 02:12.41 | OldMonk | so i still need a way in asterisk to dial and conference |
| 02:12.46 | ChannelZ | well make a MeetMe room; blind-transfer the first person in, then call up the other guy and do the same |
| 02:13.10 | OldMonk | ChannelZ: oh, right! |
| 02:13.26 | OldMonk | i forgot you can xfer someone into a meetme |
| 02:14.35 | OldMonk | maybe a stupid question, but would it be ok to have, say, 1000+ meetme rooms in an asterisk installation? |
| 02:15.13 | ChannelZ | It might not work that well |
| 02:16.07 | ChannelZ | Depends on the server, codecs involved, etc. Asterisk is doing a lot of audio mixing in conferences |
| 02:16.39 | ChannelZ | Essentially a separate one for each participant |
| 02:16.40 | OldMonk | just a handful will be active at any given time -- 1000+ would be defined, though |
| 02:16.58 | ChannelZ | You can make dynamic rooms |
| 02:17.03 | OldMonk | oh? |
| 02:17.45 | ChannelZ | or maybe I'm thinking of queues. |
| 02:18.08 | OldMonk | oh wait, i have to use confbridge anyway -- not all participants will be on DAHDI channels |
| 02:18.12 | WIMPy | You can SELECT an empty meetme. |
| 02:18.22 | ChannelZ | hmm nope the 'd' flag of MeetMe. |
| 02:18.27 | WIMPy | ConfBridge is dynamic. |
| 02:18.28 | ChannelZ | That doesn't matter OldMonk |
| 02:18.46 | ChannelZ | but ConfBridge is indeed newer and better if it currently has all the features you need |
| 02:19.04 | ChannelZ | (it's a little stripped compared to MeetMe but apparently a lot better in Asterisk 10) |
| 02:19.06 | WIMPy | But don't put a confbridge on hold. |
| 02:21.18 | ChannelZ | nobody puts baby in a corner! |
| 02:21.37 | OldMonk | WIMPy: er, why not? |
| 02:23.29 | WIMPy | Because that will have evil side effects with two participants in the confbridge. |
| 02:24.05 | OldMonk | because the way i see it, the caller will have to do something like this: join a conference C, put it on hold, make call A, transfer A to C, make call B, transfer B to C, then go back to C |
| 02:24.22 | OldMonk | sounds horrifically complicated :( |
| 02:24.49 | WIMPy | That was what I did, but it failed. |
| 02:25.08 | ChannelZ | yeah. Instead, A calls B and transfers him into the conference. A calls C and transfers him into the conference. A joins the conference. |
| 02:26.23 | OldMonk | ChannelZ: yeah, that makes more sense |
| 02:26.39 | WIMPy | Actually... When thinking about it, it was a bit special, because it happened after one of the other two got lost and I had to call him again. |
| 02:26.56 | WIMPy | That's why I put it on hold. |
| 02:27.17 | OldMonk | WIMPy: that's always a possibility in my scenario too |
| 02:27.26 | OldMonk | WIMPy: what exactly did putting the conf on hold do? |
| 02:27.35 | OldMonk | played music to everyone? |
| 02:28.22 | WIMPy | No, only to the one who remained in the conf. But it stayed that way, even when we where 3 articipants again. |
| 02:28.29 | OldMonk | ouch! |
| 02:28.46 | WIMPy | ie. th 2nd one continued to get MOH, but we could hear him. |
| 02:28.47 | OldMonk | WIMPy: how about if you mute yourself before making the new call? |
| 02:28.50 | ChannelZ | Why do you need to put anyone on hold for this? |
| 02:29.10 | WIMPy | Mute is a local thing. |
| 02:29.26 | OldMonk | ChannelZ: scenario wimpy outlined -- one of the participants drops off and you have to call her back into the conf |
| 02:29.41 | WIMPy | >> ... it happened after one of the other two got lost and I had to call him again. That's why i put my connection to the conf on hold. |
| 02:29.47 | ChannelZ | Who cares? If you put the conference on hold, you can't hear what's going on anyway so what's the point.. you hang up (leave the conf), call the person back, shove them back in, and rejoin |
| 02:29.56 | OldMonk | ah, right |
| 02:30.15 | WIMPy | Yes, you can do so. |
| 02:30.27 | ChannelZ | It's a little extra work I suppose but not by much |
| 02:30.49 | WIMPy | But not what you'd usually do. |
| 02:31.03 | WIMPy | Especially not if the one who got lost calls you back. |
| 02:31.05 | *** join/#asterisk nny1 (~Scott@64.203.238.198) |
| 02:31.54 | WIMPy | On some phones you have to think about, how to drop the current call instead of rejecting the new one. |
| 02:31.56 | nny1 | does anyone have a clever way to run something after someone leaves a meetme ? |
| 02:32.41 | WIMPy | nny1: Listen on AMI for the leave event? |
| 02:32.57 | OldMonk | or maybe just disable MOH when someone joins a meetme/confbridge |
| 02:33.27 | WIMPy | I don't think that would make a difference. |
| 02:33.44 | WIMPy | The channel goes in to hold state. |
| 02:34.03 | WIMPy | You might even get MOH from elsewhere. |
| 02:34.38 | nny1 | yeah no moh on channel atm, trying to execute a TrySystem on channel leave, I have one user accessing via context 1, the second via context 2. When 2 leaves, want to jump to next in context |
| 02:34.47 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 02:35.40 | OldMonk | nny1: sorry, the MOH was in another context (NOT asterisk context!) |
| 02:36.23 | WIMPy | Would make more sense if the channel was on hold while you're the only conference participant. Might even be that meetme did it like that. Too long ago. |
| 02:38.23 | nny1 | WIMPy: would that jump to next in context on one context? |
| 02:38.37 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 02:39.55 | nny1 | too bad commands like meetme and dial con |
| 02:40.06 | nny1 | don't have the option to jump out of context etc when their done |
| 02:40.09 | nny1 | they're* |
| 02:40.55 | WIMPy | If you want to jump to another context, use Goto. |
| 02:41.44 | WIMPy | Ok. Bedtime. |
| 02:41.46 | nny1 | WIMPy: but can't call it if a user leaves the channel |
| 02:41.47 | nny1 | ok |
| 02:41.48 | OldMonk | thanks WIMPy |
| 02:41.50 | nny1 | i'll keep lookin |
| 02:42.18 | WIMPy | If the user is gone the h extension is the only option. |
| 02:42.49 | nny1 | ooh |
| 02:42.52 | nny1 | i'll look into it |
| 02:43.07 | nny1 | that works! WIMPy thanks |
| 02:43.16 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 02:43.18 | nny1 | i owe you a coke |
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| 03:48.04 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 03:50.15 | wasanzy | I have a problem compiling chan_ss7 |
| 03:50.48 | wasanzy | am getting this error: make: *** [l4isup.o] Error 1 |
| 03:50.53 | wasanzy | any one to help? |
| 03:50.56 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 03:51.02 | wasanzy | I searched but no solution |
| 03:52.20 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
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| 04:29.59 | ChannelZ | hrmm. \< isn't a special sequence in regex is it? |
| 04:31.46 | wasanzy | me? |
| 04:35.08 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 04:40.04 | ChannelZ | well if you know the answer... |
| 04:42.07 | *** join/#asterisk jefimenko (~jordanl@unaffiliated/jordanl) |
| 04:42.31 | ChannelZ | Ahh. \< and \> are word boundaries. Never knew that. |
| 04:43.04 | jefimenko | how can i find out why asterisk doesn't start? the logs don't show anything relevant before it quits. i even tried running asterisk with -f -vvvvvvvvv |
| 04:43.37 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 04:43.38 | ChannelZ | run it with -d (and add more d's if it's not helping) |
| 04:44.35 | jefimenko | <PROTECTED> |
| 04:44.35 | jefimenko | <PROTECTED> |
| 04:44.35 | jefimenko | <PROTECTED> |
| 04:44.35 | jefimenko | <PROTECTED> |
| 04:44.35 | jefimenko | $ |
| 04:45.05 | jefimenko | this is with "asterisk -f -vvvvvvvvvvvvvvv -ddddddddddddddd" |
| 04:46.26 | ChannelZ | something ain't right because the output should look much different |
| 04:47.03 | ChannelZ | asterisk shouldn't be running at all... then asterisk -f -d |
| 04:47.16 | jefimenko | it's not running |
| 04:47.39 | jefimenko | ps aux | grep -i ast |
| 04:47.40 | jefimenko | nothing |
| 04:49.01 | X-Rob | Hah |
| 04:49.06 | X-Rob | Well I just fixed my problem. |
| 04:49.31 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 04:49.42 | jefimenko | i think i dealt with some stupid problem like this a long time ago having to do with module loading |
| 04:50.03 | jefimenko | where asterisk doesn't show any warnings/errors on startup |
| 04:50.04 | X-Rob | jefimenko, 'asterisk -vvvg' .. the last few lines will be the error |
| 04:50.05 | jefimenko | it just dies |
| 04:50.46 | jefimenko | X-Rob: thanks, but still the same thing :( |
| 04:50.58 | X-Rob | jefimenko, pastebin the entire startup |
| 04:51.27 | wasanzy | I am getting this Error when compiling chan_s77: http://pastebin.com/mY8jHfuv please help |
| 04:51.30 | wasanzy | any idea |
| 04:51.47 | ChannelZ | either you've got a junk module or are preloading something bad (it should say 'parsing modules.conf fairly early on) or a massive syntax error in whatever conf files it is reading that is killing it. |
| 04:53.04 | ChannelZ | I've never seen debug output nothing though so I don't know what the hell is happening |
| 04:53.08 | X-Rob | wasanzy, at a guess, you may need to compile and install dahdi. |
| 04:53.13 | X-Rob | and libpri |
| 04:53.37 | X-Rob | but ast_dsp_set_digitmode is in asterisk/dsp.h |
| 04:53.44 | X-Rob | oh |
| 04:53.50 | X-Rob | I bet you don't have asterisk installed. |
| 04:54.44 | wasanzy | X-Rob, I did compiled and install dahdi already |
| 04:54.56 | wasanzy | even asterisk is installed |
| 04:55.18 | X-Rob | wasanzy, sorry then. |
| 04:55.20 | ChannelZ | is the source in the right dir structures? Your output seems to be looking in a relative but specific path |
| 04:55.55 | wasanzy | X-Rob, that is ok, you are helping |
| 04:56.08 | ChannelZ | though the actual error is a little strange, I'd expect to see more barf if it couldn't find headers and things |
| 04:56.32 | wasanzy | ChannelZ: I did intall evey thing to the default dir |
| 04:56.34 | X-Rob | possibly incompatible versions, wasanzy / ChannelZ |
| 04:56.42 | ChannelZ | yeah |
| 04:56.43 | ChannelZ | shrugs |
| 04:57.22 | wasanzy | I am following this: http://www.asterisk-peru.com/files/procedimiento_sangoma.txt |
| 04:57.36 | wasanzy | though am using different versions |
| 04:57.51 | wasanzy | let me get the versions according to the doc and see |
| 04:57.55 | X-Rob | uh |
| 04:58.00 | X-Rob | chan_ss7 is at version 2.0 |
| 04:58.07 | X-Rob | http://www.netfors.com/chan_ss7_free |
| 04:58.09 | X-Rob | you're using 1.3 |
| 04:58.16 | X-Rob | there's probably your first mistake |
| 04:59.07 | wasanzy | when I use version 2.0, I get the same error |
| 04:59.13 | *** part/#asterisk jefimenko (~jordanl@unaffiliated/jordanl) |
| 04:59.20 | irroot | is back after some shut eye |
| 04:59.39 | X-Rob | and you're using asterisk 1.6? |
| 05:00.20 | wasanzy | I tried both 1.4 and now 1.6, but the same error |
| 05:00.36 | wasanzy | 1.8 gave me a different plenty error |
| 05:01.45 | X-Rob | I dunno. |
| 05:01.46 | X-Rob | http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+channels |
| 05:01.49 | X-Rob | I'd start there. |
| 05:03.37 | wasanzy | ok |
| 05:04.17 | wasanzy | some thing I kept wondering is, do I need two separate asterisk box to be able to setup chan_ss7? |
| 05:04.30 | X-Rob | Um |
| 05:04.34 | X-Rob | Do you know WHY you want SS7? |
| 05:05.58 | ChannelZ | he is haxx0r, make bluebox |
| 05:07.15 | wasanzy | because am going to setup dialers to be able to interface with asterisk so calls can be made from application |
| 05:07.37 | X-Rob | Are you sure you don't want to use PRI signalling? |
| 05:07.43 | X-Rob | because that's normally what hardware diallers use. |
| 05:09.22 | wasanzy | no I want to explore ss7 |
| 05:11.05 | X-Rob | well, may I suggest you explore PRI before you go into the very very complex and difficult world of SS7? |
| 05:11.43 | wasanzy | that is not a bad idea |
| 05:12.26 | wasanzy | the card I have is an analog card so I don't know if it will help in the ss7 thing I want to do |
| 05:12.54 | X-Rob | No. |
| 05:12.56 | X-Rob | No it won't. |
| 05:13.50 | wasanzy | which of the cards, can I go for? I can search from google but, just to make it faster, can you tell me? |
| 05:14.02 | X-Rob | Get a digium card. |
| 05:14.40 | wasanzy | the libss7 driver, which card is it for? |
| 05:14.57 | X-Rob | it says it's tested with digium cards |
| 05:15.01 | X-Rob | but that's nothing to do with compiling. |
| 05:15.43 | wasanzy | yes, but just want to make sure I have the right hardware before going feater |
| 05:15.55 | X-Rob | If you want to use SS7 I would recommend getting the support package. |
| 05:16.01 | X-Rob | Costs E1750 |
| 05:16.13 | X-Rob | http://www.netfors.com/chan_ss7_premium |
| 05:16.48 | X-Rob | If you don't want to pay that much, they may offer one-off support - ask them. http://www.netfors.com/about_us |
| 05:16.57 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:17.16 | X-Rob | I'm being extremely serious here, SS7 is not for someone who is new to Asterisk. |
| 05:18.28 | irroot | SS7 is what causes nightmares in children |
| 05:18.44 | ChannelZ | and explosive diarrhea |
| 05:19.07 | X-Rob | Crap. I just noticed that 1.8.5 is out. |
| 05:19.14 | X-Rob | has to rebuild all his RPMs |
| 05:19.30 | X-Rob | meh. Will do it later. |
| 05:19.51 | wasanzy | the chan_ss7 I guess can work with Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card? because that was what I found out |
| 05:20.11 | X-Rob | OMG. |
| 05:20.15 | ChannelZ | still doesn't understand what you're trying to do |
| 05:20.17 | X-Rob | I remember why I stopped hanging out here. |
| 05:20.24 | X-Rob | ChannelZ, he's trying to run a SS7 link across a FXO port. |
| 05:20.31 | ChannelZ | yeah to what end |
| 05:20.36 | X-Rob | And then he's going to be back wondering why it doesn't work. |
| 05:20.59 | X-Rob | wasanzy, serously, give up on SS7. SS7 is the procol used for interconnection telcos. |
| 05:21.03 | wasanzy | oh yes, I realized that, but I like to take risk |
| 05:21.29 | X-Rob | interconnecting |
| 05:21.43 | X-Rob | ..and protocol. Sigh. |
| 05:21.55 | wasanzy | with little intro, I will be able to achieve what what I want to |
| 05:22.03 | X-Rob | not with analog hardware. |
| 05:22.18 | X-Rob | 'I have a duck. I would like to plug it into my PCI bus' |
| 05:22.26 | X-Rob | Ducks and PCI buses don't interconnect. |
| 05:22.34 | X-Rob | you might be able to squash it in there, but it won't do anything. |
| 05:22.34 | ChannelZ | You just didn't push it hard enough. |
| 05:23.31 | wasanzy | no problem, I will go for the digium card as you recommended |
| 05:23.45 | X-Rob | and what are you going to plug that digium card into? |
| 05:23.52 | ChannelZ | 's eye twitches |
| 05:23.56 | X-Rob | obviously, a PCI or PCIe slot |
| 05:24.04 | X-Rob | but what's the RJ45 connection goign to be plugged into? |
| 05:24.09 | ChannelZ | Oh, I know this one! The duck! |
| 05:24.13 | X-Rob | Hahahah |
| 05:24.15 | X-Rob | I was waiting for that 8) |
| 05:24.28 | wasanzy | giving up on project is not my style |
| 05:24.44 | irroot | almost no one uses SS7 for customer connex seen it in madegascar but they have E1 option too the cell company's do it seen tekelec SS7 kit its sweet at couple M$ |
| 05:25.01 | X-Rob | wasanzy, your project doesn't make sense. |
| 05:25.05 | X-Rob | duck. pci slot. |
| 05:25.09 | irroot | when in a interconnect site like telehouse ldn you will connect with SS7 |
| 05:25.22 | *** join/#asterisk turtlefence (~turtlefen@2406:9800:1999:1:2882:7847:58c5:93ac) |
| 05:25.22 | ChannelZ | It sounded to me like "I want to make telephone calls" earlier but.. |
| 05:26.54 | wasanzy | no not just phone calls |
| 05:27.19 | ChannelZ | dirty phone calls |
| 05:27.26 | wasanzy | lol |
| 05:27.36 | turtlefence | lol |
| 05:27.45 | irroot | scams from +230/234 numbers |
| 05:27.55 | wasanzy | ChannelZ are you talking to me? |
| 05:28.18 | ChannelZ | I talk to anyone who will listen. I have no friends |
| 05:29.07 | wasanzy | X-Rob, I know the project doesn't make sense but I want to get into it, so what I want to do now is to get all the detials I need and the requirement |
| 05:29.32 | X-Rob | Right. So what are you going to plug into the RJ45 connector on the digium card? |
| 05:30.16 | wasanzy | am going to have a re-looked at what I really want to achieve, then that will tell me what I need to do |
| 05:30.31 | X-Rob | I'm telling you now, what you want to achieve does NOT include SS7 |
| 05:30.46 | kaldemar | deja vu |
| 05:30.55 | X-Rob | If it does, then you'll need to be paying lots of money to people who know what they're doing. |
| 05:30.55 | ChannelZ | no doubt |
| 05:31.14 | wasanzy | this is what I call improper planning of a project, and I accept that my self |
| 05:31.38 | X-Rob | look at your project. Find anywhere you've got 'SS7' written and cross it out. Write 'PRI' there instead. |
| 05:31.43 | ChannelZ | Earlier I was going to suggest you could work for the government, but now that you just admitted that, nevermind |
| 05:31.47 | X-Rob | Then go 'Do I need PRI?' |
| 05:32.18 | wasanzy | sure |
| 05:34.03 | wasanzy | but one question, if Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card doesn't work with ss7, what does it do then? |
| 05:34.23 | X-Rob | it does FXO and FXS |
| 05:34.30 | ChannelZ | makes phone calls |
| 05:34.42 | X-Rob | *agree* ChannelZ |
| 05:35.33 | wasanzy | ok |
| 05:36.43 | ChannelZ | scratches his head |
| 05:37.09 | X-Rob | wasanzy, go find out what the difference is between FXO, FXS, PRI_CPE and PRI_NET |
| 05:37.11 | wasanzy | so what you are saying is that, with ss7, I need some kind of connection with some telecom coampy for it to work right? |
| 05:37.21 | X-Rob | when you understand that, you'll be in a lot better position to actually figure out what you want. |
| 05:37.31 | wasanzy | ok |
| 05:37.47 | ChannelZ | I truly am fascinated by what you are trying to build, I wish you would just explain it a little. |
| 05:38.13 | ChannelZ | You're a tease and I want more! |
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| 05:38.33 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:39.54 | ChannelZ | *sigh* |
| 05:40.10 | X-Rob | I think it's time to try a complete install. |
| 05:40.44 | ChannelZ | of |
| 05:40.46 | irroot | X-Rob from the source dude make it real |
| 05:41.30 | X-Rob | ChannelZ, http://hipbx.org |
| 05:42.40 | ChannelZ | hmm |
| 05:43.40 | *** part/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 05:43.58 | X-Rob | Woo. Spam call from an asian who was trying to get my email address for something |
| 05:44.00 | X-Rob | catalog |
| 05:44.01 | X-Rob | I think |
| 05:44.05 | X-Rob | couldn't understand her accent |
| 05:44.23 | ChannelZ | give her Steve Jobs' address |
| 05:46.50 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 05:51.37 | X-Rob | I love vmware. |
| 05:52.30 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 05:52.43 | X-Rob | http://imgur.com/4oZ2K |
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| 06:19.38 | schmidts | good morning |
| 06:20.08 | irroot | morning dude |
| 06:25.22 | dan__t | word. |
| 06:40.12 | ChannelZ | to yo mutha |
| 06:47.41 | ChannelZ | So, Google still has not fixed the 'my Android phone sends text messages to the wrong people at random' bug |
| 06:47.55 | X-Rob | heh |
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| 07:06.21 | ChannelZ | Either I'm doing something wrong, or Asterisk seems to make its own decisions on NAT sometimes regardless of what options you actually have set. |
| 07:10.56 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:26.09 | ChannelZ | So is there no SpeeX file format? |
| 07:26.30 | ChannelZ | Do you just do slin16 or something else widebandish and let it convert? |
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| 08:08.37 | ezano | hi |
| 08:09.48 | ezano | I need some help in order to configure correctly voicemail on asterisk (realtime) |
| 08:10.49 | ezano | Voicemail is configured yet, but I want when I call '5555' I find directly a prompt password or my new messages |
| 08:11.18 | *** join/#asterisk wonderworld (~ww@port-92-201-29-70.dynamic.qsc.de) |
| 08:11.24 | ezano | but before prompt password I have a prompt which ask me enter my voicemail |
| 08:11.33 | ezano | I want to delete this step |
| 08:11.41 | ezano | (sorry I'm french ^^) |
| 08:11.56 | ezano | and I use asterisk with a2billing |
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| 08:24.51 | gg0 | hi, eutelia uk number ignores dtmf tones (needed by meetme) from Italy. already tried all dtmfmode values. any idea? |
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| 08:25.25 | irroot | finally getting some time to troll jira |
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| 08:34.53 | jacc0 | hi all |
| 08:34.55 | jacc0 | good morning :) |
| 08:39.10 | irroot | morning jacc) |
| 08:39.16 | irroot | morning jacc0 |
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| 09:06.00 | jacc0 | almost weekend! I'm going to a reggae festival this evening :) |
| 09:06.22 | james_zhu | :) |
| 09:06.46 | jacc0 | they already set up my tent :) |
| 09:07.06 | jacc0 | just got an sms about it :P) |
| 09:10.12 | jacc0 | my camper is broken :S |
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| 09:40.23 | oktay | hi. Is there a good cheap 4+ line (pstn and voip) device ? Like 4 spa3102s in one? |
| 09:41.03 | irroot | oktay there was a spa9000 |
| 09:41.17 | irroot | but did not find it cheap or usefull |
| 09:41.42 | irroot | if you have a spare server asterisk + digium card |
| 09:42.14 | oktay | I have a firewire server that would do. |
| 09:42.48 | irroot | server = [desktop will do fine] |
| 09:43.54 | oktay | looks like all I can get are some Zycoo ones |
| 09:44.09 | oktay | for $170 |
| 09:44.30 | oktay | unless it's a clone, i don't think the drivers would be any good |
| 09:45.19 | jacc0 | gigaset has some nice one |
| 09:45.20 | james_zhu | zycoo is a clone card, make sure it "work" |
| 09:45.32 | oktay | not that many options in Turkey |
| 09:45.33 | oktay | ( : |
| 09:45.50 | james_zhu | digium, sangoma? |
| 09:46.00 | james_zhu | you can not get ? |
| 09:46.27 | irroot | ISDN BRI ?? |
| 09:46.45 | james_zhu | digium and sangoma have BRI cards |
| 09:47.04 | Tim_Toady | sangoma bri cards are awful |
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| 09:47.27 | irroot | meant if he has BRI a standard 1 port PCI modem will be all he needs |
| 09:47.42 | irroot | if he cant get anything else |
| 09:48.05 | irroot | +1 Tim_Toady on the sangoma ISDN BRI super sucky |
| 09:48.06 | james_zhu | you want to buy BRI or PRI |
| 09:48.24 | james_zhu | digium and sangoma offer both of cards |
| 09:50.13 | jacc0 | is removing the sangoma sticker from his laptop |
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| 09:50.39 | petern_ | my digium pri card doesn't work ;p |
| 09:51.33 | irroot | petern_ what up with that it stop working ? |
| 09:55.23 | petern_ | telco says their end shows its faulty |
| 09:56.03 | petern_ | my end shows some frames |
| 09:56.04 | oktay | thanks for the info guys. I will see what's available and come back for recommendations. |
| 09:56.34 | Tim_Toady | petern_: have u set up the ports correctly, i mean in NT mode? |
| 09:56.38 | petern_ | try to dial out and it shows circuit/channel congestion |
| 09:57.29 | WIMPy | NT? |
| 09:58.07 | irroot | CPE / NET |
| 09:58.53 | petern_ | it's set to pri_cpe |
| 09:59.03 | james_zhu | yes, usually it is CPE |
| 09:59.11 | james_zhu | your provider is NET |
| 09:59.17 | WIMPy | That sounds more like it, if it's connected to your telco. |
| 09:59.23 | irroot | petern_ lsdahdi and system.conf in a pb with call trace verb+debug |
| 10:02.42 | petern_ | i lost you at "pb with call trace verb+debug" :) |
| 10:02.51 | WIMPy | ~pb |
| 10:02.52 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 10:03.04 | petern_ | oh, duh... |
| 10:03.22 | petern_ | i'm already sat here putting the bits into pastebin, heh |
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| 10:07.22 | petern_ | http://pastebin.com/Mtuw0sKP |
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| 10:12.29 | WIMPy | petern_: Your callerid is invalid. Where is that line? |
| 10:19.13 | petern_ | where? where physically? |
| 10:19.25 | WIMPy | Country |
| 10:19.28 | petern_ | UK |
| 10:19.56 | WIMPy | Ok, that should not be the cause then. |
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| 10:35.22 | petern_ | okay |
| 10:35.27 | petern_ | must be the telco |
| 10:36.15 | irroot | petern_ there is no alarm so this is odd |
| 10:36.41 | WIMPy | Has the bill been paid? |
| 10:37.11 | WIMPy | Can you call in? |
| 10:37.51 | petern_ | no, dialing in returns out of order |
| 10:38.11 | WIMPy | Without anything showing up in pri debug? |
| 10:38.15 | petern_ | correct |
| 10:38.44 | WIMPy | Then call BT or whoever. |
| 10:41.43 | petern_ | yeah, they say it's my end, heh |
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| 10:47.38 | petern_ | now they're fiddling with it, and the alarms show, heh |
| 10:48.01 | WIMPy | Ok, so at least they found the correct line :-) |
| 10:48.06 | petern_ | yeah, heh |
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| 11:02.27 | jacc0 | ping Schmidts |
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| 13:02.11 | nfi|ermes | diod anything change in 1.8 about pickup() ? |
| 13:02.25 | nfi|ermes | no more working after upgrade |
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| 13:03.45 | nfi|ermes | exten => *57,1,Pickup(s@operatore) |
| 13:03.57 | leifmadsen | syntax the same? |
| 13:03.59 | leifmadsen | what version of asterisk? |
| 13:04.06 | nfi|ermes | 1.8.5 |
| 13:04.08 | schmidts | pong jacc0 |
| 13:04.33 | nfi|ermes | app_directed_pickup.c:291 pickup_exec: No target channel found for s |
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| 13:05.46 | k3asd` | hi |
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| 13:31.00 | irroot | exten => *57,1,Pickup(s@operatore) <- you sure the rining channel is in s@ope.... |
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| 13:35.08 | Tuju | where you're supposed to add users, into sip.conf or users.conf ? |
| 13:35.21 | chazzam | sip.conf |
| 13:35.30 | chazzam | the only thing that uses users.conf is the AsteriskGUI |
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| 13:35.41 | chazzam | it should pretty much never be manually edited |
| 13:37.33 | irroot | Tuju perhaps realtime ?? then into the DB |
| 13:37.46 | Tuju | ack |
| 13:38.04 | Tuju | i could start using db at some point, but it would be nice to get started with plain files. |
| 13:38.29 | Tuju | chazzam: good, i added my first users into sip.conf but stumbled to users.conf later on and started wondering. |
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| 13:48.13 | Kobaz | does asterisk fax support error correcting mode |
| 13:50.02 | mcargile | what is the AMI command to force a local channel to resolve in 1.8? |
| 13:50.15 | Kobaz | force a local channel to do what? |
| 13:50.33 | mcargile | to go away |
| 13:50.43 | Kobaz | that's the default |
| 13:50.52 | mcargile | not always |
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| 13:51.03 | mcargile | not always fast enough |
| 13:51.07 | mcargile | i should say |
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| 14:15.02 | chazzam | Kobaz: ecm ? |
| 14:15.38 | Kobaz | ecm |
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| 14:15.59 | chazzam | yes |
| 14:16.15 | Kobaz | actually, this isn't specifically for asterisk fax |
| 14:16.22 | Kobaz | it's more for t.38 in general in asterisk |
| 14:16.32 | chazzam | http://svn.asterisk.org/svn/asterisk/branches/1.8/configs/res_fax.conf.sample |
| 14:16.38 | Kobaz | the goal would be to passthrough t.38 to a dialogic fax server |
| 14:16.57 | Kobaz | ah, ecm is t.30 |
| 14:17.00 | chazzam | then you should just need t38pt_udptl=yes in sip.conf for that peer/friend/trunk thing |
| 14:17.21 | Kobaz | and it will pass through the ecm |
| 14:18.29 | chazzam | well that is wrapped up inside the udptl for the T.38 connection |
| 14:19.00 | Kobaz | http://www.dialogic.com/interoperability/fax.htm |
| 14:19.01 | chazzam | so... yeah |
| 14:19.02 | Kobaz | k |
| 14:19.52 | Kobaz | what does res_fax do exactly? i've never used it |
| 14:19.58 | Kobaz | will it dump a pdf of the fax or something? |
| 14:20.31 | chazzam | well, it needs a fax modem stack behind it. Currently that would be either res_fax_spandsp, or res_fax_digium |
| 14:20.48 | chazzam | res_fax provides the dialplan applications and such, SendFax and ReceiveFax |
| 14:21.01 | chazzam | I guess in Asterisk 10 it will provide T.38 Gateway too |
| 14:21.20 | chazzam | but it operates on TIFF files, not pdfs |
| 14:21.24 | Kobaz | k, that's fine |
| 14:21.48 | Kobaz | and what's the major difference between spandsp and the digium one |
| 14:22.13 | chazzam | spandsp uses the spandsp modem stack which is open source. The digium one has a free and a paid version, and uses a commercial modem stack |
| 14:22.38 | chazzam | digium supports the paid version? |
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| 14:22.40 | Kobaz | i figured the open/commercial difference |
| 14:22.52 | Kobaz | i was more interested in features/stability differences |
| 14:23.14 | chazzam | Its different implementations, but the commerical one has a little more incentive to get fixed if something works in spandsp and not there |
| 14:23.22 | Kobaz | yea |
| 14:23.24 | chazzam | and perhaps more drive for features in general |
| 14:23.35 | chazzam | I can't remember if spandsp supports T.38 gateway or not |
| 14:23.37 | Kobaz | do you know any feature differences offhand? |
| 14:23.48 | chazzam | I haven't ever used the spandsp one =/ |
| 14:23.54 | Kobaz | ah |
| 14:24.42 | irroot | chazzam its in asterisk 10 |
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| 14:38.34 | schmidts | have a nice weekend guys! |
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| 14:42.03 | bmint | Anyone know of a way to set the callerid on parked calls that have timed out and are returning to the person who initially parked the call? |
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| 14:58.47 | nfi|ermes | http://pastebin.com/He3jvmS8 |
| 14:59.06 | nfi|ermes | how shoud i set the pickup command to take that call ? |
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| 15:25.17 | ChannelZ | nfi|ermes: You'd have to do Pickup(s@argo-accoglienza) if I understand your question right |
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| 15:35.31 | disgrntld81 | Is it possible to have multiple numbers ring a single POTS line? I don't want an analog DID since I don't care about the called number. |
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| 15:37.57 | malcolmd | your provider can choose to terminate multiple dialed numbers to you across a single POTS line; one at a time, of course |
| 15:39.41 | nfi|ermes | ChannelZ, can't i take extension s in any context ? |
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| 15:44.05 | nfi|ermes | <PROTECTED> |
| 15:44.05 | nfi|ermes | [Aug 5 17:43:14] NOTICE[8635]: app_directed_pickup.c:291 pickup_exec: No target channel found for s. |
| 15:44.07 | disgrntld81 | malcomd: thanks. And do you think that this situation could also be used in conjunction with a hunt group of 4 pots lines from the telco? |
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| 15:44.43 | disgrntld81 | ok never mind, i see how dumb that was |
| 15:44.46 | disgrntld81 | =P |
| 15:44.53 | malcolmd | no worries :) |
| 15:44.58 | disgrntld81 | haha |
| 15:45.03 | disgrntld81 | thanks for the info |
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| 16:01.28 | raden | morning Katty :D |
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| 16:26.15 | raden | Katty, Love your notes:) |
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| 16:58.37 | BenC[UK] | hi guys, is it possible from AMI to find out from the callid what the dialed number is? |
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| 17:10.08 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:11.30 | BenC[UK] | or a way to tie two channels together i.e sip phone to server, server to sip trunk? |
| 17:27.47 | *** join/#asterisk wonderworld (~ww@port-92-201-29-70.dynamic.qsc.de) |
| 17:30.28 | ChannelZ | nfi|ermes: If you don't specify the context, it means 'current context' which means whatever context the device executing the Pickup() application happens to be in.. which may or may not be the same context as the ringing device |
| 17:35.25 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
| 17:36.48 | *** join/#asterisk webmonkey (~webmonkey@50-106-84-116.ftwy.in.frontiernet.net) |
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| 17:48.33 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 17:52.59 | raden | how can i change the outgoing callerid for just one extension ? |
| 17:53.14 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 17:55.49 | radic | raden: you mean ${CALLERID(num) ? |
| 17:55.57 | radic | +} |
| 17:56.53 | raden | yeah but i only want it when a certain extension calls out |
| 17:57.03 | raden | like 101-105 THIS # and 106-110 this # |
| 17:58.43 | radic | exten => 101105, n, set(${CALLERID(num)}=1337) |
| 17:59.11 | *** join/#asterisk bmg505 (~leon@196-209-44-142.dynamic.isadsl.co.za) |
| 18:01.23 | paulc | raden: I've done it using the inbuilt database, using the SIP peer name to lookup the caller ID that should be used. If you have a "outbound call" macro, you can do it for everyone individually and have a default/fallback. |
| 18:03.41 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 18:17.31 | jeffspeff | does asterisk use multiple cores in a processor? |
| 18:18.01 | Qwell | Asterisk uses multiple threads. If your OS supports spreading threads around (hint: it does), then it will work as you expect. |
| 18:18.28 | jeffspeff | thanks Qwell |
| 18:27.54 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:29.00 | SuPrSluG | is it possible to use externnotify for multiple scripts? |
| 18:34.07 | _Corey_ | SuprSluG: You can always call several scripts from whatever externnotify calls |
| 18:35.18 | leifmadsen | scripts calling scripts?! |
| 18:35.24 | leifmadsen | _Corey_: you sir, are on the crazy train |
| 18:35.27 | _Corey_ | haha |
| 18:35.39 | SuPrSluG | comma seperated? |
| 18:35.57 | SuPrSluG | s/seperated/separated/ |
| 18:36.00 | leifmadsen | no |
| 18:36.00 | _Corey_ | no... |
| 18:36.08 | leifmadsen | call your script, and have it call all the scripts you want |
| 18:36.15 | leifmadsen | externnotify can only call one script |
| 18:36.28 | _Corey_ | script1.sh would itself call script2.sh, then script3.sh or whatever |
| 18:36.33 | leifmadsen | +1 |
| 18:36.41 | SuPrSluG | that's the answer i needed. thanks guys. |
| 18:37.05 | tzanger | leifmadsen: you'd know... is there any "standardized" way to run an arbitrary script/dialplan logic upon SIP registration? |
| 18:37.35 | leifmadsen | tzanger: don't think so |
| 18:37.44 | leifmadsen | unless OpenSIPS will do that :) |
| 18:37.52 | leifmadsen | sounds like a neat feature though |
| 18:38.09 | leifmadsen | registernotify=/path/to/script |
| 18:44.11 | *** join/#asterisk rdahlin_1_ (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 18:48.20 | tzanger | leifmadsen: yes, there'd have to be some kind of rate limiting/DoS protection built in |
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| 18:59.51 | raden | Katty, ??? you around ??? cisco question for you |
| 19:08.46 | leifmadsen | tzanger: agreed |
| 19:16.41 | *** join/#asterisk Kobaz (~JBouncer@its.kobaz.net) |
| 19:19.43 | ChannelZ | CRISCO! |
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| 19:38.57 | *** join/#asterisk Spanktar (~SpankyFro@204-15-0-74-static.ipnetworksinc.net) |
| 19:39.12 | Spanktar | top of the day to you all |
| 19:39.22 | Spanktar | I'm running Debian 6x with a 2.6 kernel |
| 19:39.28 | Spanktar | I have a TDM40 |
| 19:39.37 | Spanktar | I installed asterisk from apt |
| 19:39.50 | Spanktar | everthing working great, recognizes the card, etc. |
| 19:40.02 | Spanktar | BUT, there are no file is /etc/dahdi |
| 19:40.11 | Spanktar | i tried dahdi_genconf |
| 19:40.16 | Spanktar | gave me span errors |
| 19:40.25 | Spanktar | so, I feel that I'm totally missing something |
| 19:40.37 | Spanktar | as you can tell, I'm BRAND new to this |
| 19:41.09 | Spanktar | so the question is: should I just build from source instead of using the .deb packages? |
| 19:41.21 | Spanktar | seems that this will supply all of the sample files, modules and such that I need |
| 19:41.50 | Spanktar | I'm really trying to do something VERY simple. 4 FXS ports for 4 phones with extensions. |
| 19:41.51 | Spanktar | that's it |
| 19:43.18 | Spanktar | oops, that's TDM410 (with 4 FXS [green] modules) |
| 19:44.38 | Spanktar | the dox that came with it are great http://docs.digium.com/TDM410/analog410series_manual.pdf |
| 19:44.52 | Spanktar | but they take you through compiling from source not using a package manager |
| 19:45.15 | Spanktar | i prefer a package manager, but the install seems to leave out a bunch of stuff a newbie like me would need to get started |
| 19:45.32 | Spanktar | mostly /etc/dahdi/modules and /etc/dahdi/system.conf |
| 19:47.29 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 19:50.32 | chazzam | there are additional packages for DAHDI |
| 19:51.09 | chazzam | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 19:51.44 | chazzam | then try running 'dahdi_genconf' |
| 19:55.13 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 19:56.05 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 19:56.18 | saxa | Spanktar: there is also a nice guide on how to set up dahdi or the old zaptel in the asterisk book. Its pretty straight forward. |
| 19:56.26 | saxa | ~book |
| 19:56.26 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 19:57.05 | Spanktar | chazzam: I had already installed dahdi-linux and dahdi-somethingelse to no avail |
| 19:57.17 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 19:57.22 | chazzam | dahdi-tool? |
| 19:57.26 | Spanktar | yeah |
| 19:57.40 | chazzam | do you have 'dahdi_cfg' ? |
| 19:57.52 | chazzam | and can you load wctdm24xxp using modprobe? |
| 19:57.57 | Spanktar | in /etc/asterisk ? |
| 19:58.00 | Spanktar | and no |
| 19:58.04 | Spanktar | I can't |
| 19:58.11 | Spanktar | because (i figured) there aren't any modules |
| 19:58.19 | chazzam | dahdi_cfg is a binary |
| 19:58.24 | Spanktar | ...to load with modprobe |
| 19:58.27 | chazzam | comes with dahdi-tools |
| 19:58.30 | Spanktar | ok |
| 19:58.35 | Spanktar | where woudl I look for it? |
| 19:58.37 | chazzam | modprobe -vvv wctdm24xxp |
| 19:58.43 | chazzam | in your PATH? |
| 19:58.56 | chazzam | by default /usr/sbin/ |
| 19:59.02 | Spanktar | (unfortunately the machine is at home, un-networked) so I can't check right now |
| 19:59.05 | Spanktar | yeah |
| 19:59.33 | Spanktar | i remember from last night when I tried modprobe I got errors |
| 20:00.03 | chazzam | if your modprobe got errors then either you installed the wrong version of dahdi-linux, or you need to reboot |
| 20:00.16 | chazzam | or a newer kernel is out than what exists for dahdi-linux |
| 20:00.19 | saxa | Spanktar: http://pastebin.com/VQ6LdNY0 |
| 20:00.33 | saxa | Spanktar: thats my system.conf in /etc/dahdi |
| 20:00.38 | Spanktar | this worked: lspci -n |
| 20:00.52 | Spanktar | saxa: thanks I'll copy that for safekeeping |
| 20:00.57 | chazzam | so the system can see the card. |
| 20:01.01 | Spanktar | yup |
| 20:01.04 | chazzam | Spanktar: that was written by 'dahdi_genconf' |
| 20:01.34 | chazzam | Spanktar: there isn't going to be much else we can effectively help you with unless you can access the system... |
| 20:01.42 | Spanktar | yeah, I figured :( |
| 20:02.01 | Spanktar | I'm just so excited to get it working i thought someone might just say: "hell no don't use the pakcages" and I would feel better ;) |
| 20:02.23 | chazzam | packages are preference, thats about it, and you already said you prefer them |
| 20:02.24 | Spanktar | i thought I had it when i learned about genconf, but it didn't really do much for me |
| 20:02.38 | chazzam | if the module isn't loaded then genconf won't work |
| 20:02.42 | Spanktar | yeah |
| 20:02.43 | brdude | Alright I have a stupid question. I've never seen a sip phone but I'm looking into VoIP for work and looking at what I need. I heard that a sip phone has 2 ethernet ports one that you connect to your network and the other that goes to the PC. Does this mean I can use one ethernet line for the SIP phone as well as to connect my computer to the network, or does the line that goes from the phone to the PC server another purpose? |
| 20:02.49 | Spanktar | so the module comes from.... |
| 20:02.54 | Spanktar | dahdi-tools? |
| 20:02.58 | saxa | Spanktar: http://pastebin.com/8yhzbXz1 <-- here is what i have in /etc/dahdi/modules |
| 20:03.09 | chazzam | the module are kernel modules, and come with dahdi-linux |
| 20:03.18 | Spanktar | got it |
| 20:03.29 | Spanktar | that's actually really helpful information |
| 20:03.34 | chazzam | saxa: thats the default sample file for /etc/dahdi/modules |
| 20:03.54 | chazzam | the samples come with dahdi-tools |
| 20:03.55 | saxa | chazzam: probably |
| 20:04.17 | saxa | brdude: yes |
| 20:04.22 | chazzam | http://svn.asterisk.org/svn/dahdi/tools/branches/2.4/modules.sample |
| 20:04.33 | chazzam | http://svn.asterisk.org/svn/dahdi/tools/branches/2.4/system.conf.sample |
| 20:04.39 | Spanktar | when I did apt-get install dahdi-tools, I got "dahdi-tools is already installed and the newest version" but no modules file :( |
| 20:04.56 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 20:04.59 | brdude | Yes as in I can use one ethernet to connect the sip and pc? |
| 20:05.06 | saxa | yes |
| 20:05.12 | chazzam | Spanktar: then you can download those sample files and use those |
| 20:05.14 | brdude | thanks. |
| 20:05.23 | Spanktar | ok, I'll try that tonight |
| 20:05.23 | saxa | one cable to phone, from phone another cable to the pc |
| 20:05.33 | chazzam | brdude: make sure you get one that actually has both ports. the cheaper ones don't have a PC port |
| 20:05.34 | brdude | Would anyone happen to have some information were I could read more about it |
| 20:05.37 | Spanktar | thanks so much saxa and chazzam |
| 20:05.54 | saxa | at least some phones have this, my grandstream GXP works that way |
| 20:05.56 | Spanktar | this stuff is a little thick for someone just getting started |
| 20:06.04 | brdude | chazzam, sounds good. |
| 20:06.09 | chazzam | Spanktar: definitely try reading the asterisk book |
| 20:06.14 | chazzam | ~book |
| 20:06.14 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 20:06.21 | chazzam | you can read it free online |
| 20:06.25 | Spanktar | chazzam: right on |
| 20:06.36 | saxa | l8r ppl. bye |
| 20:06.42 | brdude | I just ordered the book. |
| 20:06.48 | brdude | later saxa. |
| 20:07.04 | chazzam | brdude: it might have some basic info on sip hardphones |
| 20:07.44 | brdude | yeah, the reason I ordered it is because I'm thinking about going with asterisk if we do go the way of VoIP. |
| 20:08.04 | brdude | At least I hope to convince the boss thats the way we should go. |
| 20:08.48 | chazzam | ahh, I see |
| 20:09.08 | chazzam | and yeah, a very brief description about hardphones vs softphones vs ATA's, page 86-87 |
| 20:09.29 | chazzam | brdude: might find more information from just looking at some models of phones |
| 20:09.47 | brdude | Will do. |
| 20:09.47 | chazzam | polycom spip, cisco, snom, grandstream, aastra |
| 20:11.17 | Kobaz | test |
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| 20:14.44 | brdude | chazzam, would any phone that says 2 etherenet ports but does not specify a PC port be one of the phones I should not get? |
| 20:15.13 | chazzam | if it has two one is probably a PC port, there are some that really only have one |
| 20:15.24 | chazzam | which means a completely separate drop for the phone |
| 20:16.22 | brdude | Sounds good. Most of the phones that I've seen specify 2 ports but don't specificaly mention a PC port so I though I'd ask just to be sure. |
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| 20:34.09 | Katty | hi. |
| 20:34.15 | Katty | my asterisk does not work at all |
| 20:34.17 | Katty | how to fix plz??? |
| 20:34.53 | chuckf | Katty: rm -rf / |
| 20:35.02 | Katty | what is rm?? |
| 20:35.15 | chuckf | repair module |
| 20:35.44 | chuckf | -rf is to do it really fast |
| 20:36.15 | chuckf | :D |
| 20:37.04 | Katty | *hee* |
| 20:37.06 | Katty | that's cute. |
| 20:37.09 | Katty | and going of FB. |
| 20:37.50 | chuckf | is proud to be quoted on facebook |
| 20:38.17 | brdude | lol |
| 20:38.25 | chuckf | unless you're just going to claim it as your own |
| 20:38.43 | leifmadsen | chuckf: I like that :) |
| 20:39.20 | Katty | are you on my fb? |
| 20:39.29 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 20:39.34 | Katty | i did not claim as my own, but i'd be happy to post credit where credit is due |
| 20:39.37 | chuckf | me? no |
| 20:39.43 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 20:39.43 | Katty | we must fix this! |
| 20:39.45 | Katty | chuckf: -> |
| 20:42.03 | chuckf | and on that note, I'm off to happy hour. take care all! |
| 20:44.48 | Katty | enjoy! |
| 20:45.14 | ChannelZ | the rm -rf did not works. what else to try? |
| 20:46.14 | Katty | that's the spirit ChannelZ! |
| 20:47.11 | Katty | in other mews. |
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| 20:47.28 | Katty | one of my friends is officially a doctor, as of about...2 hours ago |
| 20:47.58 | Katty | it's turned out to be a wonderful day (= |
| 20:54.49 | Kobaz | quite cool |
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| 20:55.26 | Kobaz | it's cool in here at least |
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| 22:34.14 | BenC[UK] | hi guys |
| 22:34.43 | BenC[UK] | using AMI - is there anyway I can the number a user is calling? |
| 22:35.47 | BenC[UK] | I have the call id too |
| 22:35.59 | BenC[UK] | but its only the call id of the local leg, ie from user to the server |
| 22:36.04 | BenC[UK] | not he server to the callee |
| 22:36.21 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 22:40.39 | brdude | Anyone using asterisk on a ESXi server? |
| 22:41.31 | brdude | All the articles and discusions I've found seem to only be people trying it out and nothing in an actuall production enviroment. |
| 22:43.51 | leifmadsen | brdude: nothing in production yet, but I plan on it soon enough for my business server (which, granted, will be low volume) -- it worked fine on VMware Server 2 when I had it there |
| 22:45.07 | brdude | huh, I've have a test enviroent set up on it that works as well. I'm just wondering if it'll be able to handle the load of about 100 phones since it's a real time application. |
| 22:47.00 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 22:49.44 | BenC[UK] | brdude: will all 100 phones be on at the same time? |
| 22:49.49 | BenC[UK] | like a call center.. |
| 22:49.59 | BenC[UK] | or just normal office type phones? |
| 22:50.13 | brdude | Just normal office type phones. |
| 22:50.31 | BenC[UK] | I dont see it being a problem - just monitor the resources, make sure it has enough allocated etc |
| 22:50.31 | brdude | I'm guessing maybe 20 call at a time max. |
| 22:51.13 | BenC[UK] | should be easy, I have a mini call center running with 30 phones constanly with another 10 or so people in queues etc, working absolutely fine |
| 22:51.29 | brdude | Nice to hear. |
| 22:51.40 | brdude | That's great news. |
| 22:51.44 | brdude | Thanks for the info. |
| 22:52.18 | brdude | What's the amount of resources you have alocated to it? |
| 22:54.18 | BenC[UK] | I have over allocated.. I have 4 cores (3ghz) and 4gb of ram |
| 22:54.27 | BenC[UK] | the machine has 4 quad core cpu and 32gb of ram |
| 22:54.38 | BenC[UK] | running this dialer.. and a few other small servers |
| 22:54.51 | BenC[UK] | but its hardly using any resources |
| 22:55.12 | brdude | Nice. |
| 22:55.18 | brdude | I should be fine then. |
| 22:55.53 | brdude | Thanks for all the help. |
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| 23:25.39 | X-Rob | BenC[UK], by saying '4 quad core CPU' it sounds like you're saying you have 16 cores. |
| 23:25.51 | BenC[UK] | thats right |
| 23:25.56 | X-Rob | That's insane. |
| 23:26.01 | BenC[UK] | why? |
| 23:26.04 | BenC[UK] | its a esxi server |
| 23:26.07 | X-Rob | Oh |
| 23:26.19 | BenC[UK] | which is what brdude was asking about |
| 23:26.26 | X-Rob | Pretty sure he wasn't. |
| 23:26.28 | X-Rob | he was asking about the VM |
| 23:26.38 | BenC[UK] | brdude: Anyone using asterisk on a ESXi server? |
| 23:26.56 | X-Rob | BenC[UK], you can't run in on an ESXi server. but you can run it as a guest. |
| 23:27.01 | BenC[UK] | BenC[UK]: I have over allocated.. I have 4 cores (3ghz) and 4gb of ram |
| 23:27.03 | X-Rob | you _can_ run Asterisk on an ESX server. |
| 23:27.11 | X-Rob | which someone asked me to do once. |
| 23:27.13 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 23:27.16 | X-Rob | I had to hurt them. |
| 23:27.33 | brdude | X-Rob, why is that? |
| 23:27.38 | BenC[UK] | I think brdude understood my answer |
| 23:27.56 | brdude | BenC[UK], I did no worries. |
| 23:27.56 | BenC[UK] | brdude: he is talking about running it on the host, as opposed to installing a guest OS and doing it on there |
| 23:28.29 | brdude | Actually I was talking about running it on the guest OS |
| 23:28.35 | BenC[UK] | yeah I know |
| 23:28.58 | brdude | Sorry if I confused anyone. |
| 23:29.07 | BenC[UK] | you didnt.. Rob-X is being pedantic |
| 23:29.53 | brdude | Ok I just got what he was talking about. |
| 23:30.31 | brdude | I would never have guessed someone would want to run anything on the actuall ESXi server. |
| 23:41.42 | *** join/#asterisk jackmcbarn (~jackmcbar@2002:47c7:7dd2::2) |
| 23:42.21 | jackmcbarn | the "s" extension says it only works with analog calls. what can i use to get the equivalent with SIP or something else? |
| 23:43.00 | jeffspeff | jackmcbarn, i only use sip, and many of my contexts have "s" extension |
| 23:43.06 | jeffspeff | what version are you using? |
| 23:43.28 | jwiggins | better yet what docs are you looking at? |
| 23:43.52 | jeffspeff | laughs |
| 23:45.02 | jwiggins | asterisk is not friendly to ppl who love manuals... |
| 23:45.08 | jackmcbarn | wiki.asterisk.org |
| 23:45.19 | jackmcbarn | says when an ANALOG call comes in, it will go to the s extension |
| 23:45.28 | jackmcbarn | when i try it with a sip phone, the s extension doesnt happen |
| 23:45.32 | jeffspeff | jackmcbarn, are you having problems with a particular context or something? |
| 23:45.42 | jackmcbarn | the default context a call first goes into |
| 23:45.47 | jwiggins | thats a confluence dashboard |
| 23:45.58 | jackmcbarn | when i lift my sip phone up, i expect the s extension to start |
| 23:46.00 | jackmcbarn | and it doesnt |
| 23:46.10 | jeffspeff | jackmcbarn, sip doesn't work like that |
| 23:46.15 | jeffspeff | you have to dial something first |
| 23:46.20 | jeffspeff | then it starts at extension s |
| 23:46.47 | jeffspeff | such as "exten=s,1,Dial(SIP/provider/5555555555) |
| 23:47.05 | jwiggins | you need a pattern match... |
| 23:47.07 | jeffspeff | provider being your context in sip.conf |
| 23:47.27 | jeffspeff | jwiggins, i was just showing for example of exten=s,1 |
| 23:47.30 | jackmcbarn | so if i pick upaa phone and dont dial anything, what does asterisk do until i do dial something? |
| 23:47.36 | jwiggins | _X.,NoOp("Asterisk works!) |
| 23:47.37 | jeffspeff | nothing |
| 23:47.48 | jwiggins | urg |
| 23:47.52 | jwiggins | add a " |
| 23:48.03 | jackmcbarn | nothing? |
| 23:48.04 | jeffspeff | asterisk has no clue what your phone is doing until your phone tries to send something to asterisk |
| 23:48.20 | jwiggins | jeffspeff is spot on |
| 23:48.25 | jackmcbarn | do any phones have an option to send something to asterisk before i start dialing? |
| 23:48.32 | jeffspeff | POTS phones |
| 23:48.39 | jwiggins | what are you trying to do? |
| 23:48.54 | jeffspeff | jackmcbarn, what you're looking for just isn't part of how SIP functions |
| 23:48.59 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 23:49.37 | jwiggins | sip dialtone is generated by the device not the server which is likely your confusion |
| 23:50.05 | jeffspeff | agrees |
| 23:50.33 | jwiggins | what is your goal jackmcbarn? |
| 23:51.10 | jwiggins | what do you want done as soon as they pickup the receiver? |
| 23:51.20 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-171-223.hsd1.va.comcast.net) |
| 23:51.33 | jeffspeff | looks for jackmcbarn |
| 23:51.37 | jackmcbarn | i want to be able to start running an extension |
| 23:51.51 | jwiggins | to do... |
| 23:51.53 | jeffspeff | jackmcbarn, what extension? what does it do? |
| 23:52.06 | jwiggins | there may be other ways |
| 23:52.19 | jwiggins | to do what your use case is |
| 23:53.13 | jackmcbarn | i want to know if theres a way to do it generically |
| 23:53.30 | jeffspeff | jackmcbarn, to do what? |
| 23:53.48 | *** join/#asterisk gravin (~gravin@45.70.50.60.brf01-home.tm.net.my) |
| 23:54.05 | jeffspeff | you can't set it to dial an extension as soon as the phone is lifted from the hook, unless your phone specifically allows for that... that's out of asterisks control |
| 23:55.08 | jackmcbarn | okay |
| 23:55.46 | jackmcbarn | also, i have a test extension that plays a sound file then runs Hangup(). when i do that, my phone plays a busy signal until i hang up. is that asterisk, or is that the phone doing that? |
| 23:55.56 | jackmcbarn | (that is, after the sound file is done) |
| 23:56.19 | jeffspeff | do a pastebin of that context so i can take a look |
| 23:56.22 | jwiggins | in the * logs do you see the playback function called? |
| 23:56.31 | jwiggins | ahh |
| 23:56.37 | jwiggins | yea pastebin |
| 23:57.07 | jwiggins | sounds like an issue with the extension |
| 23:57.14 | jeffspeff | agrees |