| 00:17.41 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 00:19.04 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 00:20.51 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 00:31.02 | *** join/#asterisk corrupt (i=user@128.227.22.108) |
| 00:31.22 | corrupt | does anyone play around with asterisk on their ubuntu box? |
| 00:34.04 | Sweeper | corrupt: well, this is the internet, so I assume there's SOME perverted bastard that does |
| 00:34.27 | corrupt | why does he have to be perverted? |
| 00:34.40 | carrar | you did say ubuntu |
| 00:34.44 | corrupt | lol |
| 00:35.00 | corrupt | carrar, what do the nonpervs use? |
| 00:35.05 | carrar | CentOS! |
| 00:35.36 | carrar | 4.4 works nicely with 1.4 |
| 00:36.06 | corrupt | hmm, i've got more questions, but i've got to catch the bus at the moment. bbl. |
| 00:36.24 | _DAW | 4.4 is great. Anyone running 5? |
| 00:36.28 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.189) |
| 00:36.29 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 00:48.26 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 00:48.26 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:55.36 | *** join/#asterisk rahail (i=rahail@209.190.75.32) |
| 00:55.45 | rahail | ? seen bobocop |
| 00:55.53 | rahail | !seen bobocop |
| 00:59.30 | *** join/#asterisk pcm (n=pcm@65.4.17.49) |
| 01:00.24 | Qwell | ~seen mercestes |
| 01:01.03 | jbot | mercestes <n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com> was last seen on IRC in channel #asterisk, 2h 47m 17s ago, saying: 'smoked1, Good luck'. |
| 01:01.03 | pcm | ~seen kram |
| 01:01.17 | jbot | kram <n=mark@pdpc/sponsor/digium/kram> was last seen on IRC in channel #asterisk, 208d 19h 40m 31s ago, saying: 'hrm, anyone here know brian mcmanus?'. |
| 01:01.17 | Qwell | kram was last seen 13 weeks (3h 2m 36s) ago |
| 01:01.19 | pcm | qwell: you're not a bot |
| 01:01.24 | Qwell | oh |
| 01:01.52 | *** join/#asterisk dc3aes (n=matt@S01060001023fe8ca.no.shawcable.net) |
| 01:02.19 | red9012 | so is the queue app going to include a way for the called extension to be able to put the caller back on hold? |
| 01:04.19 | rahail | ~seen bobo* |
| 01:04.32 | jbot | rahail: i haven't seen 'bobo*' |
| 01:04.32 | rahail | hmm |
| 01:04.41 | rahail | ~seen bobocop |
| 01:04.44 | jbot | bobocop <n=Bobocop@uz186.internetdsl.tpnet.pl> was last seen on IRC in channel #asterisk, 9h 3m 41s ago, saying: 'howdy! :) I'm trying to get patton sip/isdn gateway working both ways. I can't get through with trunk definition. I get it only working as incoming or outgoing. Bever both :( How should trunk be definied? What about user/password vs trunk name? ... |
| 01:09.38 | *** part/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 01:34.19 | *** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 01:35.30 | demlak | hmm... whats wrong with this? |
| 01:35.31 | demlak | http://rafb.net/p/0NZu1p85.html |
| 01:35.42 | demlak | no file generated.. no output to console... |
| 01:35.55 | demlak | the commands are working in the shell |
| 01:36.04 | russellb | ~seen Qwell |
| 01:36.40 | jbot | qwell is currently on #asterisk (2d 16h 3m 18s). Has said a total of 3 messages. Is idling for 35m 16s, last said: 'oh'. |
| 01:36.40 | Qwell | . |
| 01:36.40 | russellb | :-p |
| 01:36.40 | demlak | but they donīt seem to work in the extensions.conf |
| 01:36.50 | Qwell | weird |
| 01:37.10 | demlak | any idea? |
| 01:38.56 | *** join/#asterisk Fieldy (i=7NZbAUO4@gentoo/contributor/Fieldy) |
| 01:41.59 | russellb | Qwell: are you calling me weird? |
| 01:43.54 | Qwell | no, jbot |
| 01:44.01 | jazzanova | vonage-out/1778 216.115.20.41 255.255.255.255 5060 UNREACHABLE |
| 01:44.04 | Qwell | he ran his query right after you said that, but took forever to answer |
| 01:44.08 | jazzanova | why is it unreachable ? |
| 01:44.18 | jazzanova | it shows that it is Registered |
| 01:44.48 | jazzanova | sphone.vopr.vonage.net:5060 17787850134 15 Registered |
| 01:45.22 | file | russellb: I'm calling you weird. |
| 01:45.26 | jazzanova | in what way is it uncheable ? in network way, or in config file way ? |
| 01:47.20 | dc3aes | hey.. now that i see that.. and apologies if this is repeated everywhere but can an asterisk box connect to vonage as SIP? not really sure why i would want to but nice to know if we could |
| 01:49.24 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
| 01:51.05 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 01:52.48 | *** join/#asterisk littleball (n=littleba@bb220-255-154-109.singnet.com.sg) |
| 02:03.40 | shido6 | yes |
| 02:03.57 | shido6 | if you pretend to be a softphone and look like xlite for example |
| 02:04.22 | shido6 | but they will catch on when u pass over a million minutes through them on a tiny package :) |
| 02:08.30 | sevard | put on a softphone costume and run around the parking lot |
| 02:11.12 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 02:11.59 | Fieldy | heh |
| 02:16.45 | *** join/#asterisk ccole (n=cole@cpe-24-166-57-30.neo.res.rr.com) |
| 02:17.55 | *** join/#asterisk darylvoip (n=darylvoi@c-71-224-42-97.hsd1.pa.comcast.net) |
| 02:18.01 | _VoiceMeUp_COM | not 1 million |
| 02:18.10 | _VoiceMeUp_COM | just 20% over avg and you cut + 200$ fee |
| 02:18.34 | _VoiceMeUp_COM | had a client bully us sayint he was running 2 million miunutes a month on vonage residential package |
| 02:18.42 | _VoiceMeUp_COM | i told him he was day dreaming, |
| 02:18.45 | _VoiceMeUp_COM | and so he was |
| 02:19.10 | dc3aes | shido6: haha thx |
| 02:19.11 | _VoiceMeUp_COM | actualy got 4 client ssay that before they went to us.. kind of weird lol |
| 02:19.46 | ccole | Help! :) What does this message mean: Apr 27 22:19:04 NOTICE[7713]: codec_zap.c:856 find_transcoders: No Zaptel transcoder support! |
| 02:20.28 | ccole | I just upgraded my asterisk and zaptel driver, and now I get this message... |
| 02:21.34 | ccole | I am running asterisk 1.2.17 and zaptel driver 1.2.16 |
| 02:28.34 | shido6 | ewwww |
| 02:30.39 | ccole | OK asterisk server is working OK now; I had my _extension line goofed. Man asterisk is PICKY! |
| 02:30.41 | ccole | :) |
| 02:32.12 | russellb | it does what it tells you to do |
| 02:32.36 | russellb | that's the thing about comptuers, they do what you say, not what you want, necessarily :) |
| 02:33.17 | *** join/#asterisk _pkNew (n=chatzill@mbl-82-62-215.dsl.net.pk) |
| 02:34.50 | _pkNew | hi i want to make reports based on queue_logs |
| 02:35.08 | _pkNew | can some one please tell me how to get "number of incomming call" ?? |
| 02:35.11 | jazzanova | i am getting UNREACHABLE from my vonage-out |
| 02:35.18 | jazzanova | i'm a newbie |
| 02:35.35 | jazzanova | can someone give a suggestion ? |
| 02:37.04 | russellb | jazzanova: set qualify=no, maybe it's not supported by them |
| 02:37.35 | jazzanova | russellb: ok, i am going to remove qualify |
| 02:37.41 | jazzanova | but I don't know if the connection is working. |
| 02:37.50 | jazzanova | it says that the status is "Registered" |
| 02:37.52 | russellb | have you tried calling it? |
| 02:38.10 | jazzanova | yes, and its not getting to the server. i am getting a vonage answering machine. |
| 02:39.00 | jazzanova | also, its re-registering very often. |
| 02:40.00 | ccole | How come I can get an outbound call to work with: exten=>_1NXXNXXXXXX,1,Dial... but I cannot get an outbound call to work with: exten=> _NXXNXXXXXX,1,Dial... ?? I do not want to have to type a '1' before dialing a phone number. |
| 02:40.59 | mihinomenest | you need to add the "1" to your Dial() |
| 02:41.20 | jazzanova | its re-registering every 20 seconds |
| 02:42.48 | jazzanova | also, i am getting this in the log: |
| 02:42.49 | jazzanova | Apr 27 22:40:15 WARNING[20941]: Unable to get our IP address, Skinny disabled |
| 02:42.56 | jazzanova | where do i set the ip address ? |
| 02:50.17 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 02:50.41 | kiwoneka | good evening to all |
| 02:51.55 | *** join/#asterisk Fieldy (i=TrSHp8O2@gentoo/contributor/Fieldy) |
| 02:52.27 | jazzanova | russellb: ok, when I call to my phone, i can see in asterisk sip debug output. but i don't know where the call is routed |
| 02:53.13 | *** join/#asterisk vonkleist (n=gera@189.155.128.168) |
| 02:53.23 | vonkleist | Hi everybody |
| 02:53.48 | vonkleist | I live in mexico, and just bought a tdm22b |
| 02:54.04 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-177-38-240.vic.bigpond.net.au) |
| 02:54.06 | vonkleist | I installed asterisknow, and now ihave all my extensions working |
| 02:54.24 | vonkleist | But what I can do, is to get calls out to my 2 lines |
| 02:54.45 | vonkleist | Extensions rings, and rings, but call never get established |
| 02:54.53 | vonkleist | also, it seems like calls aren't going out |
| 02:55.00 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 03:01.43 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 03:01.46 | *** join/#asterisk flashnet (n=flashnet@12-207-153-162.client.mchsi.com) |
| 03:01.47 | vonkleist | ummm... tdm400p, I mean |
| 03:06.45 | kiwoneka | what a night |
| 03:06.53 | kiwoneka | i need some netwrking help |
| 03:07.55 | kiwoneka | i sent grandma a polycom601, now it connects and gets provisioned just fine |
| 03:08.07 | kiwoneka | but we cant hear grandma |
| 03:08.43 | kiwoneka | i have tried alot of things, nw i need some professional advice |
| 03:10.12 | Nugget | have you turned on qualify= for that sip peer? |
| 03:10.31 | Nugget | sounds like the usual NAT difficulties |
| 03:13.03 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 03:13.34 | kiwoneka | here is my entry in sip.conf http://pastebin.ca/462183 |
| 03:15.17 | kiwoneka | should i specify, a value - qualify=500 |
| 03:21.10 | _pkNew | hi every body |
| 03:21.24 | _pkNew | i want to make reports of a call center using queue_logs |
| 03:21.39 | _pkNew | how do i get the number of calls offered/answered/abondaned ?? |
| 03:23.39 | _pkNew | can some body give some idea please ?? |
| 03:24.03 | [TK]D-Fender | kiwoneka, you need to add "nat=yes" and "canreinvite=no" to her entry and reload sip. Then have her place another call |
| 03:25.19 | [TK]D-Fender | _pkNew, look in [source folder]/docs/queuelog.txt for the format and you'll know what to do with the data collected... |
| 03:26.12 | _pkNew | well i dont know which event to count for incomming/answered/not answered calls |
| 03:26.58 | _pkNew | can you explain the difference between the Connect and EnterQueue event ? |
| 03:27.49 | [TK]D-Fender | _pkNew, Enterqueue is the line that states that the call has entered the queue (is the first line of detail). Connect tells you the hold time & who answered the call. |
| 03:27.56 | kiwoneka | can i make nat=yes a global setting |
| 03:28.11 | [TK]D-Fender | _pkNew, its all in the doc. please READ IT. |
| 03:28.30 | [TK]D-Fender | kiwoneka, Only if that applies to your server as well (being behind a NAT of its own). |
| 03:28.42 | [TK]D-Fender | kiwoneka, If thats the case as well you have a LOT more work to do. |
| 03:28.46 | kiwoneka | i am behind a fw |
| 03:29.13 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 03:29.14 | [TK]D-Fender | kiwoneka, Do the changes to her phone's entry and pastebin your [general] section. |
| 03:29.14 | [TK]D-Fender | ~pb |
| 03:29.25 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 03:30.00 | _pkNew | yes i have read that file, but i'm still confused how the calculate the number of incomming/offered/answerd and not answered calls |
| 03:30.14 | Cinen | Has anyone experienced all sip commands disappearing from 1.4.4? |
| 03:30.25 | _pkNew | only if you can please explain me which event to count for which..... |
| 03:30.28 | Cinen | sip still works just no sip commands |
| 03:31.37 | [TK]D-Fender | _pkNew, incoming = total of "enterqueue". answered = # of "connect". |
| 03:31.56 | [TK]D-Fender | _pkNew, count your abandoned, etc. |
| 03:32.10 | [TK]D-Fender | _pkNew, its all jsut a sum of differnt reason codes. |
| 03:32.46 | _pkNew | yes thanks....thats right that its the sum of different codes, but i was confused about those events |
| 03:33.11 | [TK]D-Fender | _pkNew, the meaning of each is well defined in the readme there. |
| 03:33.33 | _pkNew | like connect and completeagent and completecaller shows somewhat the same thing, that is the call is connected and answered |
| 03:33.57 | _pkNew | so i was not sure to take which event for answered call |
| 03:34.34 | [TK]D-Fender | _pkNew, yes & no. |
| 03:35.01 | [TK]D-Fender | _pkNew, connect = answered. the other 2 are to say who killed the call and state the duration. Thats for counting talk time. |
| 03:35.36 | _pkNew | alright |
| 03:36.03 | _pkNew | and how do i find which agent got how many calls, like i want to calculate how many calls did agent_1 got ? |
| 03:36.35 | [TK]D-Fender | _pkNew, I found I had 1 agent who seems to dump calls REGULARLY. and I mean a LOT. Hed have a whole pile of calls he'd grab & dum in < 2 seconds. |
| 03:37.02 | [TK]D-Fender | _pkNew, we are not sure if its deliberate, or if he has a serious neurologicl condition on the answer button :) |
| 03:37.13 | _pkNew | :D |
| 03:37.23 | [TK]D-Fender | _pkNew, Connect tells you who answered. You should be READING THE DOC! :) |
| 03:37.30 | [TK]D-Fender | _pkNew, Don't get lazy on me now! |
| 03:37.42 | [TK]D-Fender | _pkNew, And go check your queue log for some sample data |
| 03:38.17 | _pkNew | yes but its not my fault, the document on queue_log on voip-info is not updated, it shows Connect(holdtime) only |
| 03:38.34 | [TK]D-Fender | _pkNew, I said the one in your SOURCE/DOC folder |
| 03:38.38 | kiwoneka | [TK]D-Fender: http://pastebin.ca/462214 |
| 03:38.53 | _pkNew | yes, i just got the latest one, thanks :) |
| 03:39.25 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:39.27 | [TK]D-Fender | _pkNew, Here... no more excuses now! http://www.pastebin.ca/462217 |
| 03:40.11 | _pkNew | thanks a lot man |
| 03:41.38 | [TK]D-Fender | kiwoneka, ok, assuming those values are right, looks good to me. Have her test. |
| 03:41.51 | *** part/#asterisk pcm (n=pcm@65.4.17.49) |
| 03:42.32 | kiwoneka | :( grandma is 76, she is long gone to bed |
| 03:42.36 | kiwoneka | but thanks |
| 03:42.51 | kiwoneka | do i have to do anything to my fw |
| 03:43.00 | kiwoneka | open any ports |
| 03:43.05 | kiwoneka | i use ipcop |
| 03:43.26 | kiwoneka | i forwarded 10000 - 101000 |
| 03:43.58 | [TK]D-Fender | kiwoneka, 5060,10000-20000 all UDP |
| 03:44.52 | *** join/#asterisk khronos (n=khronos@c-76-110-134-230.hsd1.fl.comcast.net) |
| 03:47.08 | kiwoneka | all set up |
| 03:48.58 | [TK]D-Fender | kiwoneka, You should be good to go. But set her qualify for about 2000. 500 is way too small. |
| 03:49.22 | kiwoneka | what does that mean |
| 03:50.09 | *** join/#asterisk flashnet (n=flashnet@12-207-153-162.client.mchsi.com) |
| 03:50.48 | kiwoneka | what is the difference - qualify=yes and qualify=2000 |
| 03:51.49 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 03:54.15 | [TK]D-Fender | kiwoneka, none. yes=2000. Though I saw you mention 500 earlier. 2000 is a genrally safe number |
| 03:54.53 | kiwoneka | then the default value is 2000 |
| 03:55.07 | *** join/#asterisk bmg505 (n=leon@196.209.179.68) |
| 03:56.52 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 03:57.17 | [[blah]asfd | is it not possible to do a pause queue member from the cli prompt like you can add queue member? |
| 03:58.11 | *** join/#asterisk Kizmet (n=cappleby@76.233.11.210-static.velocitynet.com.au) |
| 04:00.51 | *** join/#asterisk Kizmet (n=kizmet@76.233.11.210-static.velocitynet.com.au) |
| 04:01.26 | *** join/#asterisk NormanAthol (n=Norman@203.208.66.241) |
| 04:03.00 | khronos | Are there any h323 packages for Centos? |
| 04:03.17 | khronos | I want to build asterisk with sip, iax and h323 capibilities. |
| 04:03.45 | [TK]D-Fender | khronos, openh323 |
| 04:04.39 | khronos | There any packagtes for this in Centos or will I have to build this by hand. |
| 04:04.52 | khronos | I'm wanting to use the latest 1.4 release. |
| 04:05.01 | [[blah]asfd | <PROTECTED> |
| 04:05.07 | [[blah]asfd | you will get it there. |
| 04:05.43 | [TK]D-Fender | khronos, My instal of CentOS 4.4 + * 1.4.2 gave me chan_ooh323.so by default. |
| 04:06.15 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 04:06.19 | khronos | Doesn't look like Centos 5 has openh323 packages |
| 04:07.22 | [[blah]asfd | in make menuconfig for asterisk... with openh323 installed in cent 4.4 I have: XXX 5. chan_h323. Doesnt that mean it cannot be installed for asterisk? I dont need it, but was looking for khronos's sake and saw that. |
| 04:07.41 | [TK]D-Fender | khronos, Its brand new, expect reports of compatability / guides to be sparse. |
| 04:08.06 | NormanAthol | i am having trouble getting call parking to work it is not transfering the calls when i dial #700 (i am testing with another user in the local context could this cause the problem trying to park someone on teh same context from what i understand for a simple dial plan i only need to add include => parkedcalls to the dial plan is this correct |
| 04:08.19 | [TK]D-Fender | chan_ooh323.so Objective Systems H323 Channel |
| 04:08.34 | [TK]D-Fender | [[blah]asfd, this is the channel drive I get off a basic install. |
| 04:08.54 | [TK]D-Fender | NormanAthol, pastebin your dialplan |
| 04:08.55 | [TK]D-Fender | ~pb |
| 04:09.06 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 04:09.15 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 04:09.29 | NormanAthol | i know thoose sites all to well |
| 04:10.50 | NormanAthol | http://pastebin.ca/462253 |
| 04:10.59 | *** part/#asterisk _pkNew (n=chatzill@mbl-82-62-215.dsl.net.pk) |
| 04:11.04 | NormanAthol | very simple @ this stge |
| 04:11.38 | *** join/#asterisk Rahail (n=Rahail@209-19-88-240.detroit.mi.D-Conn.net) |
| 04:12.53 | [TK]D-Fender | NormanAthol, In order to use parking you need to be able to do DTMF transfers. you have forgotten to add this option to your dial statements. |
| 04:13.27 | NormanAthol | how do i go about that |
| 04:14.13 | *** join/#asterisk corrupt (n=user@n128-227-137-77.xlate.ufl.edu) |
| 04:14.16 | [TK]D-Fender | NormanAthol, "show application dial" |
| 04:14.38 | NormanAthol | thatmade no sence to me |
| 04:14.44 | corrupt | for what purposes do you all use asterisk for? |
| 04:15.59 | NormanAthol | fun |
| 04:16.16 | *** join/#asterisk fx0 (n=ident@voip.terrorist.net) |
| 04:16.31 | [TK]D-Fender | corrupt, word domination. |
| 04:16.59 | [TK]D-Fender | NormanAthol, thats the command at * CLI to read the instructions on the Dial app. |
| 04:17.22 | NormanAthol | ok now i feel stupid thankyou |
| 04:18.25 | [TK]D-Fender | My work here is done :) |
| 04:20.01 | NormanAthol | it all @ least makes sence now |
| 04:20.17 | [[blah]asfd | here is a question i have been up against... if my polycom 301 drops power while on a call... the channel stays open until i manually kill it with a soft hangup. Is there a way around that? |
| 04:21.34 | [TK]D-Fender | [[blah]asfd, set an rtptimeout in rtp.conf |
| 04:21.58 | *** join/#asterisk burt75 (n=humberto@189.154.34.190) |
| 04:22.08 | [[blah]asfd | how long is appropriate do you think? |
| 04:22.12 | burt75 | hello guys |
| 04:22.15 | [[blah]asfd | 1 hour maybe? |
| 04:23.12 | burt75 | Guys ,,, who has implemented Chan_bluetooth or Chan_cellphone |
| 04:26.17 | [TK]D-Fender | [[blah]asfd, probably half that. I think RTP is TOLD to stop for Hold. So a smaller amount might be fine. I'd say start smaller and grow in increments. |
| 04:26.45 | [TK]D-Fender | [[blah]asfd, maybe 10 min to start. then 15,20,30,45,60. |
| 04:26.48 | [TK]D-Fender | as needed |
| 04:27.03 | NormanAthol | just not sure how i would put that into the dial plan |
| 04:27.57 | NormanAthol | exten => 100,1,Dial,D or exten => 100,1,Dial (D(caller)SIP/usernme) or di i somehow defin it in global |
| 04:28.26 | [TK]D-Fender | exten => 200,1,Dial(SIP/mefo,,tT) |
| 04:29.07 | [TK]D-Fender | NormanAthol, What do you think you were doing with "D"?! |
| 04:30.08 | NormanAthol | i have no idea |
| 04:30.27 | [TK]D-Fender | heh |
| 04:31.04 | [TK]D-Fender | ok, well read further down and you'll see that t & T allow you control which end(s) of the call are allowed to transfer the call. |
| 04:36.27 | *** part/#asterisk fx0 (n=ident@voip.terrorist.net) |
| 04:38.09 | NormanAthol | i see what you mean now i got it figured out |
| 04:39.35 | [TK]D-Fender | NormanAthol, 1.4 seems to have added k & K |
| 04:40.19 | corrupt | does asterisk support speech recognition? |
| 04:42.31 | [TK]D-Fender | corrupt, It not offered by * directly, but there are pacakages you can add for this. 1 free one is CMU Sphinx. |
| 04:44.21 | ber123 | how good is sphinx |
| 04:44.26 | ber123 | that festival program sucks balls |
| 04:45.45 | [TK]D-Fender | ber123, Try & see |
| 04:52.45 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 05:15.28 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:18.49 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:21.01 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.189) |
| 05:24.54 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 05:32.51 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 05:37.06 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 05:46.49 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:57.18 | Rahail | any one here used a2billing |
| 06:02.07 | *** part/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 06:13.13 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 06:25.12 | *** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it) |
| 06:28.55 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 06:32.06 | glock | would u guys recommend 1.2 or 1.4? |
| 06:34.42 | *** join/#asterisk jcaceres (n=josexato@201.240.108.194) |
| 06:35.31 | jcaceres | hello, i have a doubt, i wanna configure asterisk realtime just to load the iax users from a data base |
| 06:35.57 | jcaceres | i've configured extconfig.conf with the right parameters |
| 06:36.10 | jcaceres | and added rtcachefriends=yes to iax2.conf |
| 06:36.20 | jcaceres | is there anything more i need to do? |
| 06:36.50 | jcaceres | sorry it was iax.conf the name of the file |
| 06:36.53 | *** join/#asterisk voltagex (n=voltagex@124-254-99-2-dsl.ispone.net.au) |
| 06:37.07 | jcaceres | i have added a row to the table |
| 06:37.19 | jcaceres | <PROTECTED> |
| 06:37.20 | voltagex | is it possible to change the tonezone progmatically, from an agi or extension? |
| 06:37.42 | jcaceres | nothing happens |
| 06:38.14 | jcaceres | voltagex, do you have some experience with asterisk Realtime? |
| 06:38.24 | voltagex | no |
| 06:38.29 | jcaceres | :S |
| 06:48.29 | *** join/#asterisk jazzanova (n=boris@S010600146cfc7d5b.vc.shawcable.net) |
| 07:01.27 | *** join/#asterisk corrupt (i=user@128.227.22.108) |
| 07:04.27 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:07.27 | *** join/#asterisk mkl1525 (n=qwertz@i59F703D1.versanet.de) |
| 07:07.56 | mkl1525 | Hi, is there an option to lower volume in Playback() cmd? |
| 07:12.45 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 07:12.48 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 07:16.02 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 07:19.30 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 07:20.30 | [[blah]asfd | how can I tell if I am truly getting port 4569 out to the internet... I have a machine that is not connecting to binfone, while the same configs on another machine works just fine. |
| 07:20.39 | [[blah]asfd | ports are forwarded to the server and everything... |
| 07:20.47 | [[blah]asfd | what other troubleshooting steps can i take? |
| 07:22.09 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 07:27.02 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 07:28.45 | corrupt | [[blah]asfd, ping. |
| 07:29.22 | uwe | hello, im having voice interruption (dropping of voice for short times), but no load on the cpu on asterisk ! and the phones are cisco ip phones ... what can it be ? they are connected to a switched network and i cant figure out what the problem can possible be ... |
| 07:29.50 | corrupt | uwe, is this a problem on the job? |
| 07:29.59 | jcaceres | asteriskguy, |
| 07:30.27 | jcaceres | [[blah]asfd, you can check tcpdump |
| 07:30.28 | uwe | corrupt, actually not at my job, but at -a- job |
| 07:31.36 | uwe | corrupt, why? is this channel for help only for at home asterisk ? |
| 07:33.18 | uwe | hmm .... |
| 07:36.06 | [[blah]asfd | will tcpdump show upd packets? |
| 07:36.39 | justdave | yep |
| 07:36.57 | justdave | the name is a bit of a misnomer, it generally shows any traffic on the NIC (even ARP stuff) |
| 07:37.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:40.16 | [[blah]asfd | if i am using register with iax, does it use more than just port 4569? it should all go through the same port right? |
| 07:40.38 | [[blah]asfd | i can get it to connect to other servers other than binfone, but the same account info on someone elses server is working just fine. |
| 07:47.31 | jcaceres | if its posible use wireshark, it will let you see the entire leg of the call |
| 07:48.08 | jcaceres | i have a noop question, :S what is the diference between iax peer and iax user?? |
| 07:49.08 | jcaceres | i catching up with asterisk, i stoped using it a very long time |
| 07:49.15 | jcaceres | any idea? |
| 07:50.46 | jazzanova | jcaceres: user can receive, peer can send ? |
| 07:51.08 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 07:53.47 | jcaceres | any of them can do both? |
| 07:55.39 | jazzanova | friend |
| 07:55.48 | jazzanova | friend = user + peer |
| 07:56.17 | jazzanova | jcaceres: can you help me with making a call through a sip provider ? |
| 07:56.24 | jazzanova | its my first day with asterisk |
| 07:56.28 | jazzanova | :) |
| 07:58.14 | jcaceres | well, it's mainly the same day for me :D |
| 07:58.51 | jcaceres | but i have seen a sample of how to do it in iax.conf sample |
| 07:58.58 | jcaceres | for FWD |
| 07:59.08 | jcaceres | so it might be the same |
| 07:59.11 | jcaceres | i asume |
| 07:59.13 | jcaceres | :D |
| 08:00.58 | jazzanova | i have been reading sip.conf and extensions.conf |
| 08:01.39 | jazzanova | i am getting an error in a log: Apr 28 03:34:54 WARNING[22757]: Unable to open IAX timing interface: No such file or directory |
| 08:01.59 | *** join/#asterisk zapata (n=user@chello213047080026.4.14.vie.surfer.at) |
| 08:01.59 | jazzanova | what is this timing interface, and how do i set it ? |
| 08:05.09 | jazzanova | <PROTECTED> |
| 08:05.10 | jazzanova | <PROTECTED> |
| 08:05.23 | jazzanova | what am i doing wrong ? |
| 08:05.31 | *** join/#asterisk Strom_M (n=strom@135.196.213.180) |
| 08:05.34 | jazzanova | i'm also having trouble calling into it. |
| 08:05.44 | jazzanova | asterisk doesn't pickup |
| 08:10.30 | jcaceres | normaly when it's refered as timing is about zaptel |
| 08:10.39 | jcaceres | that gives timing |
| 08:24.45 | *** join/#asterisk angom_h (n=Angel@red-corp-201.170.77.174.telnor.net) |
| 08:29.18 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
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| 08:44.51 | *** join/#asterisk Strom_C (n=strom@135.196.213.180) |
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| 09:33.25 | *** join/#asterisk jcaceres (n=josexato@201.240.108.194) |
| 09:34.41 | jcaceres | hello, i have compiled astersik-addons, and configured cdr_mysql.conf correctly bot i can not get any info in the database |
| 09:35.26 | jcaceres | i think i need to add a module to be loaded in modules.conf, but i am not sure which one |
| 09:35.37 | jcaceres | any idea? |
| 09:36.58 | *** join/#asterisk __andrew (n=andrew@implode.fuckdom.net) |
| 09:36.58 | __andrew | hey |
| 09:37.04 | __andrew | anyone using bt pri? |
| 09:37.13 | __andrew | that could help with a wee problem |
| 09:38.40 | JunK-Y | jcaceres: when ya do a module show like cdr, do you see mysql? |
| 09:40.34 | JunK-Y | jcaceres: you should see cdr_addon_mysql.so . |
| 09:41.49 | jcaceres | no i dont |
| 09:42.10 | jcaceres | but i compiled asterisk addons |
| 09:42.16 | JunK-Y | so when you compile asterisk-addons, it doesnt compile right. |
| 09:42.20 | jcaceres | as told in voip_info |
| 09:42.32 | JunK-Y | make sure you have libmysqlclient-dev installed. |
| 09:42.47 | jcaceres | is there any special configuration needed? when compiling ? |
| 09:42.57 | JunK-Y | also, ya can do make menuselect and make sure you haev all deps. |
| 09:43.09 | jcaceres | thanks a lot |
| 09:43.19 | jcaceres | i'll tias |
| 09:44.05 | __andrew | I getting == Everyone is busy/congested at this time (1:0/0/1) from my pri card when trying to make outgoing calls. |
| 09:44.11 | __andrew | incoming are fine - anyone any pointers? |
| 09:55.09 | jcaceres | JunK-Y, you were completly right thanks |
| 09:57.08 | *** join/#asterisk saftsack (n=saftsack@pD9E06B9B.dip.t-dialin.net) |
| 10:01.42 | demlak | hi |
| 10:02.37 | demlak | hmm... whats wrong with this? http://rafb.net/p/0NZu1p85.html there is no output to the console |
| 10:03.00 | demlak | btw.. im uing busybox |
| 10:03.05 | demlak | using |
| 10:09.13 | JunK-Y | use NoOp(test) instead of echo test to see it in the * CLI. |
| 10:09.40 | JunK-Y | and once its hanguped, it must be exten => h, since ur channel is dead. |
| 10:10.17 | demlak | i tried also to hangup at last step |
| 10:10.34 | demlak | iīll try NoOp(test) now.. just a moment |
| 10:17.07 | demlak | no "test" output to the * CLI http://rafb.net/p/74HvBd59.html |
| 10:17.45 | demlak | itīs the same like before |
| 10:18.44 | demlak | fax is received... nad thats all |
| 10:18.48 | demlak | and |
| 10:20.17 | JunK-Y | do core set verbose 4 |
| 10:20.25 | JunK-Y | and do it again |
| 10:23.54 | demlak | still no "test" |
| 10:24.46 | JunK-Y | show me ur CLI output. |
| 10:25.14 | demlak | well... how? =) |
| 10:25.17 | JunK-Y | and a dialplan show s@fax-in |
| 10:25.20 | demlak | ah wait... ssh =) |
| 10:25.45 | JunK-Y | so how can ya say nothing if you're not already in a ssh session? |
| 10:25.54 | demlak | monitor |
| 10:26.09 | JunK-Y | monitor? |
| 10:26.20 | demlak | VGA output =) |
| 10:26.25 | JunK-Y | ahh, at the monitor ;) |
| 10:27.29 | JunK-Y | u could also add a NoOp(test1) before you hangup |
| 10:27.53 | JunK-Y | did ya type dialplan reload after you changed ur dialplan? |
| 10:27.57 | carrar | I find ,1,Hangup works best |
| 10:28.01 | demlak | wait a second |
| 10:32.54 | demlak | hmpf... since im logged in via ssh.. receiving fax didnīt work anymore |
| 10:34.45 | JunK-Y | its h,1,Hangup() yes. |
| 10:37.11 | demlak | i did an asterisk stop now.. and a new start.. now this output http://rafb.net/p/d4pAPj53.html |
| 10:37.43 | JunK-Y | its exten => h,1,NoOp(test2); |
| 10:38.14 | demlak | but still no test1 in the CLI |
| 10:38.50 | JunK-Y | output of dialplan show s@fax-in |
| 10:38.52 | JunK-Y | ? |
| 10:39.11 | JunK-Y | in 1.2 its show dialplan s@fax-in |
| 10:41.22 | JunK-Y | so? |
| 10:41.29 | demlak | wait a second =) |
| 10:43.16 | demlak | i killed all.. and startet from scratch with the macro... http://rafb.net/p/oGUFYS33.html |
| 10:44.12 | JunK-Y | i need to see ur show dialplan output |
| 10:44.22 | demlak | its there |
| 10:44.28 | demlak | middle part |
| 10:45.06 | JunK-Y | so the test2 is there now. |
| 10:45.37 | JunK-Y | try to flip ur capi and the noop (priority 1 and 2 to see) |
| 10:46.14 | JunK-Y | it seems ur capi hangups the channel. |
| 10:47.08 | demlak | test1 is now showing |
| 10:47.15 | demlak | test 2 also |
| 10:47.29 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:47.42 | demlak | iīll now try system(echo fghdghj) |
| 10:48.27 | JunK-Y | u wont see the output in the CLI. |
| 10:48.59 | JunK-Y | NoOp is the way to see thru the CLI |
| 10:49.01 | demlak | itīs verbose |
| 10:49.20 | demlak | and it worked =) |
| 10:49.46 | JunK-Y | so enjoy :) |
| 10:50.13 | demlak | thx a lot |
| 10:50.23 | demlak | but thats just the beginning =) |
| 10:50.47 | JunK-Y | i know, we all pass by there. |
| 10:51.23 | demlak | what i need is the filename of the currently saved fax file |
| 10:51.32 | demlak | for sending it |
| 10:52.33 | JunK-Y | just write a small program that does that, which receiving the * uniqueid |
| 10:52.39 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 10:55.51 | demlak | ok.. how to get the uniquid? =) |
| 10:56.45 | demlak | my idea was this.. but this gives a new uniquid =) exten => h,1,System(/bin/echo '${UNIQUEID}' >> /tmp/test) |
| 10:58.36 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:00.44 | *** join/#asterisk angryuser (n=aster@i03v-213-44-169-43.d4.club-internet.fr) |
| 11:00.55 | angryuser | hi |
| 11:01.38 | angryuser | i am searching for people who is using voipstunt with asterisk |
| 11:01.57 | |ryan| | angryuser: What are you angry about? |
| 11:02.22 | angryuser | http://forums.digium.com/viewtopic.php?t=15250 here why |
| 11:06.09 | angryuser | |ryan|: any ideas? |
| 11:08.29 | angryuser | later |
| 11:09.29 | tuxick | is it normal that asterisk keeps showing "REGISTER attempt" to voip provider? |
| 11:09.44 | tuxick | it *is* registered, everything works |
| 11:13.36 | *** join/#asterisk thenet (i=overload@tinfoilhat.net) |
| 11:16.00 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 11:17.49 | *** part/#asterisk thenet (i=overload@tinfoilhat.net) |
| 11:26.21 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 11:34.29 | *** join/#asterisk SoftIce (n=bongo@vc-196-207-45-253.3g.vodacom.co.za) |
| 11:34.59 | *** join/#asterisk Thisone (i=nobody@70.86.176.7) |
| 11:35.23 | SoftIce | hi, please can somebody tell me what is theory to do as follows, what I would like to do is say dial exten 1000 when I do that i leave a message then i want that message to be sent to an e-mail address? |
| 11:37.34 | SoftIce | could somebody maybe point me to the right documentation ? |
| 11:37.37 | SoftIce | thank you :) |
| 11:38.33 | *** join/#asterisk ircrly (i=astrutt@punk.valuetel.net) |
| 11:41.27 | r0d3nt | =) |
| 11:41.41 | demlak | variable for current date and time? |
| 11:43.31 | JunK-Y | demlak: see the doc/channelvariables.txt |
| 11:44.32 | demlak | thx |
| 11:45.05 | demlak | doesnīt exist |
| 11:45.44 | *** join/#asterisk ltd (n=z@ppp167-251-11.static.internode.on.net) |
| 11:47.28 | demlak | README.variables |
| 11:47.30 | demlak | thx |
| 11:53.25 | demlak | ok.. works... now.. how to react on this.. "capi receivefax: fax receive failed reason=0x34a2 reasonB3=0x0000" i want the macro react different on failed fax receive |
| 12:10.41 | |ryan| | r0d3nt: Who let you off 2600net? |
| 12:13.07 | r0d3nt | yer'killin me.. i always idle here.. stfu.. |
| 12:14.49 | |ryan| | yeah, but you spoke. |
| 12:21.07 | *** join/#asterisk UVSoft (n=UVSoft@motorola154-31.ip.PeterStar.net) |
| 12:21.12 | UVSoft | hi |
| 12:21.18 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 12:22.23 | UVSoft | could anybody explane me how i can make asterisk pick up the phone (fxo device) after the first ring please? |
| 12:24.20 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 12:30.02 | JT | UVSoft: use Wait before answering |
| 12:32.15 | |ryan| | Wait(5) should ring once |
| 12:32.47 | |ryan| | it doesn't wait for CallerID data to be sent? |
| 12:33.00 | UVSoft | JT: it takes too much time for asterisk to pick up the phone and enter to the dialplan... actually i want it to pick up as fast as possible... |
| 12:33.27 | JunK-Y | UVSoft: usecallerid=no |
| 12:33.33 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 12:34.54 | *** join/#asterisk MaartenB_ (n=Maarten@213-73-219-133.cable.quicknet.nl) |
| 12:39.38 | *** join/#asterisk Sypher|NL (n=Sypher@s5590f00b.adsl.wanadoo.nl) |
| 12:39.49 | Sypher|NL | Hi |
| 12:42.42 | UVSoft | JunK-Y: thanks, that's fine, but it still takes at least two rings... is there a way to pick up immediately? |
| 12:43.09 | Sypher|NL | i am having some problems with SIP. I think its something with signaling |
| 12:43.38 | DrukenLPY | UVSoft: when you changed the usecallerid=no did you restart ? |
| 12:43.54 | UVSoft | yep |
| 12:44.48 | DrukenLPY | check your file, make sure you've changed all occurances for that port... |
| 12:45.11 | DrukenLPY | sounds as if it's still waiting for caller id |
| 12:46.44 | UVSoft | now it takes less time, but still i have to wait for about 2-3 rings |
| 12:47.33 | DrukenLPY | pastebin your configs |
| 12:47.39 | JT | UVSoft: did you restart zaptel? |
| 12:49.25 | JunK-Y | restart zaptel? |
| 12:49.49 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-178-65.buckeyecom.net) |
| 12:50.08 | UVSoft | JT: it isnt zaptel's business, zaptel knows nothing about zapata.conf |
| 12:50.15 | demlak | hmm.. voicemail timestamp is not correct... it is 2 hours behind the system time... i read about timezones.. but there is no /usr/share/zoneinfo/ on my minimal system... |
| 12:52.53 | UVSoft | it all is because of asterisk, i wonder there are no any configuration files to set up such a simple thing.... |
| 12:53.35 | DrukenLPY | well, if i remove the usecallerid on mine, it picks up about 1 second after it rings.... |
| 12:53.40 | DrukenLPY | so it's your config |
| 12:55.27 | DrukenLPY | however i use CID, so.. i can wait for the caller id :) |
| 12:55.27 | UVSoft | is that the only thing asterisk could possibly do before it picks up? |
| 12:55.27 | UVSoft | mb there's something else |
| 12:55.27 | UVSoft | what do you think? |
| 12:55.30 | DrukenLPY | i don't think nothing till i see configs posted.... |
| 12:55.47 | UVSoft | zapata conf? |
| 12:56.04 | DrukenLPY | and your default context from extensions.conf |
| 12:58.34 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
| 13:02.30 | UVSoft | mmm i have some problems with pastebin.com, are there any other pastebins? |
| 13:03.04 | DrukenLPY | pastebin.ca |
| 13:04.16 | UVSoft | http://pastebin.ca/462655 |
| 13:08.01 | DrukenLPY | so your not using cid right? |
| 13:08.24 | UVSoft | yep |
| 13:08.34 | DrukenLPY | so why do you have the added options to make cid work in the uk ? |
| 13:09.03 | UVSoft | i used to, that's why my config has cid settings |
| 13:09.54 | DrukenLPY | i'm wondering if perhaps that is why it's pausing... |
| 13:10.17 | DrukenLPY | i'm not sure if that is the problem or not, since i've only had to deal with north american lines... :) |
| 13:10.51 | UVSoft | usecallerid=no tha't the point, and nothing else matters) |
| 13:11.08 | JT | err |
| 13:11.14 | JT | how about you stop assuming everything |
| 13:11.15 | DrukenLPY | one would think so.... |
| 13:11.21 | JT | and just put a semicolon in front of them |
| 13:11.30 | JT | the idea is to eliminate variables |
| 13:11.35 | JT | not make assumptions |
| 13:12.01 | UVSoft | ok, i'm gonna try it right now, i'll let you know |
| 13:15.05 | Corydon76-home | Uh, if you comment them out, you're going to need to do a restart |
| 13:16.42 | UVSoft | )) i'm wondering why all the people here worry so much about restarts... |
| 13:17.18 | UVSoft | so i was right, nothing's changed, it still takes three ring for asterisk to pick up |
| 13:18.20 | JT | ok, have fun with it then |
| 13:18.21 | Corydon76-home | If you watch the console, when does asterisk first notice the call? |
| 13:18.56 | Corydon76-home | It may be your telco isn't sending the call until the 3rd ring |
| 13:19.30 | UVSoft | right after i push a dial button |
| 13:19.47 | Corydon76-home | on your cell phone? |
| 13:19.51 | UVSoft | yep |
| 13:20.16 | Corydon76-home | So Asterisk says something like "Starting simple switch on Zap/1-1" ? |
| 13:20.33 | Corydon76-home | and then waits 3 seconds? |
| 13:20.41 | UVSoft | something like that |
| 13:20.47 | JT | pretty specific |
| 13:21.50 | *** join/#asterisk ariel_ (n=ariel_@70-46-87-154.ftl.fdn.com) |
| 13:21.59 | Corydon76-home | How are you connected to the telco? |
| 13:22.12 | Corydon76-home | Your signalling on Zap/1 looks wrong |
| 13:22.38 | Corydon76-home | You're using FXO signalling on an FXO port, which is wrong |
| 13:22.50 | Corydon76-home | You should be using FXS signalling on the FXO port |
| 13:22.56 | UVSoft | i know it |
| 13:23.01 | JT | know it all |
| 13:23.05 | JT | you know absolutely everything |
| 13:23.06 | JT | clearly |
| 13:23.10 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128666864.dsl.bell.ca) |
| 13:23.26 | Corydon76-home | Change that and try again |
| 13:23.27 | UVSoft | zap1 fxs, zap2 fxo |
| 13:23.46 | Corydon76-home | Oh, okay |
| 13:27.24 | Corydon76-home | UVSoft: could I ask you to remove the spaces from your extensions.conf |
| 13:28.07 | JT | i bet he'll tell you he knows about that already ;) |
| 13:28.08 | Corydon76-home | i.e. exten => s,1,Answer, not s, 1, Answer |
| 13:29.01 | Corydon76-home | Also, are you running anything else on this machine? Oracle database server, Java app server, etc.? |
| 13:30.18 | UVSoft | nothing that could have such an effect |
| 13:30.44 | JT | he asked you if you were running anything else on the machine |
| 13:31.00 | JT | not anything else on your machine that you determined to be a problem |
| 13:31.25 | UVSoft | JT: ......... |
| 13:31.25 | Corydon76-home | Also, try changing to immediate=yes |
| 13:31.44 | JT | UVSoft: so stop avoiding the questions and instructions, we're only try to help |
| 13:31.49 | tuxick | is this "REGISTER attempt " every X minutes normal? |
| 13:32.04 | JT | trying, even |
| 13:32.30 | JT | UVSoft: removed those erroneous spaces yet? |
| 13:32.59 | UVSoft | yes |
| 13:33.05 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 13:33.54 | JT | UVSoft: so are there any other services or daemons running on the machine? |
| 13:35.16 | UVSoft | sshd telnetd, dhcp client.... that's it |
| 13:35.16 | Nugget | telnet is eeeeeeevil! |
| 13:35.23 | UVSoft | ) |
| 13:35.31 | JT | ok |
| 13:38.49 | UVSoft | JT: sorry about my behaviour, mb i'm just sick and tired of all this stuff, that's why i may be so rude or something... just trying to figure it out |
| 13:39.27 | JT | UVSoft: what version of asterisk and zaptel? |
| 13:40.15 | UVSoft | the letest ones |
| 13:41.05 | JT | what would that be? there's a few branches |
| 13:41.05 | DrukenLPY | uhg... |
| 13:41.05 | UVSoft | asterisk 1.4.* zaptel |
| 13:41.05 | UVSoft | 1.2* |
| 13:41.17 | JT | very specific, again |
| 13:42.10 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:43.14 | Corydon76-home | UVSoft: SVN 1.4 r12345 would be the way to specify that |
| 13:45.34 | UVSoft | actually i'm not quite sure about the versions.... so you recommend me to download the latest asterisk and zaptel? |
| 13:45.57 | Corydon76-home | Latest 1.4, NOT TRUNK |
| 13:46.23 | UVSoft | ok, i'll check it out |
| 13:46.47 | Corydon76-home | btw, if you're already using svn, the command to get the version is 'svn info' |
| 13:47.32 | Corydon76-home | but try immediate=yes if you haven't already |
| 13:47.37 | Corydon76-home | (first) |
| 13:50.26 | UVSoft | if i set immediate to yes, asterisk won't let me dial any number on my FXS devices, it'll just start executing the dialplan immediately after the first dtmf |
| 13:50.52 | UVSoft | it's not that i want |
| 13:51.12 | UVSoft | so let immediate be 'no' |
| 13:52.54 | ManxPower | UVSoft: you can set that option per channel |
| 13:53.13 | ManxPower | remember any option you set in /etc/asterisk/zapata.conf will apply to all following channels unless overridden |
| 13:53.23 | UVSoft | hmmm.... i didnt think about that |
| 13:53.51 | UVSoft | great idea |
| 13:53.56 | UVSoft | ) |
| 13:54.05 | ManxPower | anyway the question is does the phone ring for 3 seconds BEFORE you see the Starting simple switch or do you see Starting simple switch then there is a 3 second delay? |
| 13:54.16 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:54.46 | UVSoft | just a moment, i'm going to check it... i'll be back |
| 13:55.26 | riddlebox | when you create an auto attendant how do set a timeout feature? |
| 13:55.44 | ManxPower | riddlebox: see README.variables |
| 13:55.54 | ManxPower | use the Set command to set the timeout |
| 13:56.03 | riddlebox | ok |
| 13:56.42 | Corydon76-home | Also, the first argument to WaitExten is a timeout |
| 14:04.22 | uwe | um, ive been having problems with sound quality, and it seems it could be an irq issue, the wct4xxp has irq 233 , but i dont have many devices ... would it be bad/good to give it a priority higher than the ide for example ? |
| 14:04.31 | UVSoft | ManxPower: the phone rings 1-4 times and than i see "Starting...." and asterisk picks up |
| 14:10.06 | iq | Hi |
| 14:14.44 | demlak | intelligent nickname! |
| 14:14.45 | demlak | =) |
| 14:19.58 | *** join/#asterisk nitram (i=foo@superblob.com) |
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| 15:11.25 | *** join/#asterisk PioneerVM (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 15:11.45 | PioneerVM | hi |
| 15:13.11 | PioneerVM | I am trying to use "Dial" to fw a call with -- Dial(Local/15556667777@outgoing/n) and it works -- however I tried duplicating the [outgoing] context to [outgoingfw] context so i can make a few setting changes for caller ID |
| 15:13.19 | PioneerVM | and changed @outgoing to @outgoingfw |
| 15:13.59 | PioneerVM | but I get an error "No such extension/context 15556667777@outgoingfw creating local channel |
| 15:14.04 | PioneerVM | any thoughts? |
| 15:14.47 | ariel_ | do you have an include in your extensions contexts that has outgoingfw in it? |
| 15:15.30 | PioneerVM | i have [outgoingfw] defined in extensions.conf -- [outgoing] was defined there with 3 lines for US calling through voicepulse and i just copied those 3 lines and put them into a [outgoingfw] context |
| 15:15.35 | SoftIce | hi, please can somebody tell me what is theory to do as follows, what I would like to do is say dial exten 1000 when I do that i leave a message then i want that message to be sent to an e-mail address? |
| 15:15.40 | SoftIce | could somebody maybe point me to the right documentation ? |
| 15:15.42 | PioneerVM | so [outgoing] and [outgoingfw] are basically copies of each other |
| 15:16.05 | PioneerVM | im wondering if i have to list [outgoingfw] somewhere else |
| 15:16.20 | PioneerVM | or is [outgoing] a special context name within asterisk? |
| 15:18.15 | ariel_ | PioneerVM, you need to include that context in the context that your local extension is setup in. |
| 15:18.24 | ariel_ | SoftIce, vm will do that for you. |
| 15:18.32 | ariel_ | it's part of asterisk |
| 15:18.37 | ariel_ | ~docs |
| 15:18.45 | jbot | rumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 15:18.51 | PioneerVM | do you mean sip.conf or iax.conf? |
| 15:19.23 | ariel_ | your extensions belong in a context like context=local |
| 15:19.30 | SoftIce | ariel_: so anyone voicemail recorded will get sent to what ever mail address I specify? |
| 15:19.46 | ariel_ | in the section for [local] you need to include the context you need your extension to see |
| 15:19.59 | ariel_ | SoftIce, yes as long as you configure it |
| 15:20.03 | demlak | hmm.. voicemail timestamp is not correct... it is 2 hours behind the system time... i read about timezones.. but there is no /usr/share/zoneinfo/ on my minimal system... |
| 15:20.10 | ariel_ | there are examples of this in the conf.sample files |
| 15:20.28 | PioneerVM | where would i find the local context? |
| 15:20.33 | PioneerVM | i cant seem tof ind it with grep |
| 15:20.56 | ariel_ | PioneerVM, have you looked at the complete extensions.conf file? |
| 15:21.01 | PioneerVM | yes |
| 15:21.20 | ariel_ | what context is your extension set to? |
| 15:22.41 | PioneerVM | im confusing the terms you are using -- i have these contexts [globals], [general], [macro-voicepulse*], [fw-outgoing], [outgoing], [voicepulse-in], [incoming] |
| 15:22.49 | PioneerVM | in extensions.conf |
| 15:23.15 | PioneerVM | currently in "incoming" i have some menu setup that when the user presses 3 I run: Dial(Local/15556667777@outgoing/n) |
| 15:23.21 | ariel_ | well you just posted part of your inssue |
| 15:23.29 | PioneerVM | ok good :) |
| 15:23.36 | ariel_ | <PROTECTED> |
| 15:23.41 | ariel_ | not outgoing-fw |
| 15:24.05 | ariel_ | 15556667777@outgoingfw was what you had posted |
| 15:24.12 | PioneerVM | oh i had done it both ways |
| 15:24.21 | PioneerVM | ill doublec heck now sorry i had tried various versions of the name |
| 15:25.01 | PioneerVM | nope its currently fw-outgoing in boht |
| 15:25.02 | ariel_ | but your device what is it's context and in that context does it have any include=outgoing |
| 15:25.04 | PioneerVM | and still not working |
| 15:25.19 | ariel_ | what device is it your trying to make the call with? |
| 15:25.20 | *** join/#asterisk cspot (i=cspot@ip68-109-8-207.pn.at.cox.net) |
| 15:25.23 | PioneerVM | hmm i dont understand what you mean by device |
| 15:25.27 | PioneerVM | im using voicepulse |
| 15:25.47 | PioneerVM | they are configured in sip.conf and iax.conf |
| 15:26.15 | ariel_ | how are you calling the plan from a sip phone? |
| 15:26.27 | ariel_ | directly from the inbound call from voicepulse? |
| 15:26.45 | PioneerVM | i am using my home phone (regular land line) to call into my voicepulse phone which goes to the asterisk menu system |
| 15:26.55 | PioneerVM | when i press "3" on the menu system it calls teh Dial line |
| 15:27.08 | PioneerVM | which works if I say @outgoing but if i say @fw-outgoing it fails |
| 15:27.25 | PioneerVM | but outgoing and fw-outgoing are duplicates of each other which leads me to believe i have to add "fw-outgoing" somewhere |
| 15:27.29 | ariel_ | ok what context is your inbound setting for voicepulse |
| 15:28.09 | PioneerVM | [voicepulse-in] which uses "Goto" to go to [incoming] |
| 15:28.11 | PioneerVM | where the menu system is |
| 15:28.13 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 15:28.30 | ariel_ | ok now incoming do you have any includes there under that heading? |
| 15:28.50 | PioneerVM | I am not sure what you mean by includes in the asterisk world, but i dont think so |
| 15:28.56 | PioneerVM | just using standard exten => lines |
| 15:29.03 | PioneerVM | with Answer, Set, Background, Goto, etc. |
| 15:29.20 | PioneerVM | its about 10 lines, plays a msg, lets users press 1,2,3 -- if they hit 3 i call the Dial line |
| 15:29.26 | PioneerVM | and it calls my cell phone |
| 15:29.40 | ariel_ | ok post your file extensions.conf on pastebin.ca so I can see it. |
| 15:30.05 | PioneerVM | i dont know how to do that |
| 15:30.08 | PioneerVM | sorry new around here |
| 15:31.19 | ariel_ | ~pastebin |
| 15:31.23 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
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| 15:31.44 | PioneerVM | ahh cool im looking now |
| 15:31.50 | PioneerVM | hold on just want to remove the keys from the file |
| 15:32.01 | PioneerVM | do u need comments |
| 15:32.04 | PioneerVM | or can i strip those out |
| 15:32.16 | JT | leave them in |
| 15:32.24 | JT | only substitute passwords |
| 15:33.05 | SoftIce | hmm, wha configuration needs to be taken for your voicemails to be attached as mp3's ? |
| 15:33.12 | ariel_ | if it's too large full of examples and non use context info remove them as they might be part of what is getting in your way. |
| 15:33.14 | SoftIce | or some better compression than .wav? |
| 15:33.38 | *** part/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
| 15:34.38 | JT | ariel_: remove them from the actual file first though, instead of just the pastebin |
| 15:34.53 | ariel_ | jt correct |
| 15:34.56 | PioneerVM | I just need to remove "API_KEY" and any phone #'s right? |
| 15:35.14 | PioneerVM | just want to make sure they dont bury pw's in these lines -- dont see any but new and just want to make sure |
| 15:35.50 | JT | what the hell is api_key |
| 15:35.58 | PioneerVM | oh wait I MAY have figured out the problem one sec |
| 15:36.00 | PioneerVM | lol |
| 15:36.07 | PioneerVM | API_KEY is something for voicepulse, i guess its my pw |
| 15:36.18 | JT | PioneerVM: usually you leave the phone numbers in |
| 15:36.25 | JT | i see... |
| 15:36.53 | PioneerVM | ok no my idea didnt work, still finishing substitutions one sec |
| 15:37.02 | PioneerVM | i thought i caught an error but it was not it |
| 15:37.20 | JT | only change passwords.... |
| 15:37.24 | JT | secret= |
| 15:37.39 | PioneerVM | yea im just making sure i know where they all are, voicepulse made this file |
| 15:37.46 | PioneerVM | and im new so just reading thru it carefully |
| 15:37.46 | JT | eww |
| 15:38.12 | PioneerVM | actually i found the problem! |
| 15:38.25 | PioneerVM | when i copied incoming -> incomingfw |
| 15:38.33 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 15:38.34 | PioneerVM | i didnt have a ,1, |
| 15:38.37 | PioneerVM | only ,n, |
| 15:38.50 | PioneerVM | i thought i added ,1, myself to the original file |
| 15:38.58 | PioneerVM | and it was to set caller ID info |
| 15:39.06 | PioneerVM | so i copied that to and it fixed the problem. ugh. |
| 15:39.40 | PioneerVM | the error didnt indicate that it couldnt find the first priority line so i didnt realize and im just learning this |
| 15:39.51 | PioneerVM | thanks for talking me through it |
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| 15:44.34 | Sweeper | ~book |
| 15:44.42 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:46.30 | PioneerVM | ok next question -- I have a call come in, it's answered by a menu system. Lets say the user calls from 1-222-333-4444 -- then they hit #3 on the phone system and i forward the call, using Dial to my cell phone -- i want to set the caller ID to be 1-222-333-4444 |
| 15:46.49 | PioneerVM | Do i need to store the original called # in a varialbe for use later, or is there a variable i can access with teh original # |
| 15:51.06 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
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| 15:55.52 | danp | save it for later |
| 15:56.40 | danp | actually i think there's an option for Dial to pass it through for you |
| 15:58.12 | PioneerVM | do you know the proper format to save it for later |
| 15:58.23 | PioneerVM | will it come into ${CALLERID{all}}? |
| 15:58.31 | PioneerVM | or is that only for what I set |
| 15:58.55 | PioneerVM | I tried using Set(ORIG_NUMBER=${CALLERID{all}}) but that does not seem to work |
| 16:02.01 | danp | try CALLERID(all) |
| 16:02.53 | PioneerVM | Is this correct: |
| 16:03.01 | PioneerVM | Set(ORIG_NUM=122223334444) |
| 16:03.04 | PioneerVM | and then later: |
| 16:03.20 | PioneerVM | Set(CALLERID(num)=${ORIG_NUM}) |
| 16:03.28 | PioneerVM | if I do: |
| 16:03.39 | PioneerVM | Set(CALLERID(num)=12223334444) it works, but if i do the above it does not |
| 16:04.01 | PioneerVM | either my format is wrong or Set does not work across contexts? |
| 16:05.39 | russellb | you have more 2's in ORIG_NUM :) |
| 16:05.48 | PioneerVM | lol |
| 16:05.50 | PioneerVM | ok forget that part |
| 16:05.59 | PioneerVM | is my format correct? |
| 16:06.16 | russellb | but assuming you do actually set ORIG_NUM on the same channel that you are trying to do the next set on, then yes, it should work fine |
| 16:06.33 | PioneerVM | by channel do you mean context |
| 16:06.42 | russellb | no |
| 16:06.54 | PioneerVM | Hmm ok i just tested this |
| 16:06.57 | russellb | by channel, i mean call |
| 16:07.07 | russellb | you should see both Set executions for the same call |
| 16:07.18 | PioneerVM | I have the call come into "incoming" with a menu system -- i was trying to set ORIG_NUM there, then the call goes to Dial which jumps to "outgoing" context |
| 16:07.25 | PioneerVM | and i tried to access ORIG_NUM there but it is gone apparently |
| 16:07.45 | PioneerVM | i tested that by also setting ORIG_NUM in the outgoing page to see if it worked and it did |
| 16:07.45 | russellb | o.O |
| 16:07.57 | PioneerVM | so how do i get the variable to set in the incoming call and grab it in the outgoing? |
| 16:08.17 | PioneerVM | must be some way to access it across the channel or context? |
| 16:08.38 | russellb | once you set it on the channel, it will always be on the channel |
| 16:08.43 | russellb | regardless of what context it is in |
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| 16:08.46 | PioneerVM | is there a way to pass it on? |
| 16:09.01 | PioneerVM | i think i am using two channels |
| 16:09.05 | PioneerVM | the call comes in on one channel to a menu |
| 16:09.13 | russellb | ok, then yes, you can use variable inheritance |
| 16:09.15 | PioneerVM | then the menu dials out to a cell phone to transfer the user |
| 16:09.22 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 16:09.25 | russellb | if you set variables with preceding underscores, it will be inherited |
| 16:09.35 | russellb | Set(__ORIG_NUM=123124123123123) |
| 16:09.35 | PioneerVM | ahhh so put _ORIG_NUM on the set? |
| 16:09.43 | PioneerVM | two underscores? |
| 16:09.46 | PioneerVM | and i read it with __ as well? |
| 16:09.52 | russellb | one underscore to be inherited once, two to be inherited for forever |
| 16:09.57 | russellb | no, you read it as normal |
| 16:10.17 | PioneerVM | testing |
| 16:10.29 | PioneerVM | sweet it worked |
| 16:10.30 | PioneerVM | thanks |
| 16:10.43 | russellb | you're welcome |
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| 16:17.03 | Frosh | Asterisk appliances work like the asterisk software? |
| 16:17.17 | shido6 | :) |
| 16:19.26 | Frosh | why go through the hassle of setting one up, when you have the hardware that does the same thing? |
| 16:19.48 | PioneerVM | im guessing cost |
| 16:19.57 | shido6 | do what fits. |
| 16:20.01 | shido6 | use what fits. |
| 16:20.37 | PioneerVM | and control |
| 16:22.09 | PioneerVM | anyone know the limits of what caller ID services will take |
| 16:22.20 | PioneerVM | it seems you cant just set "any" number that it has to fit a certain format |
| 16:24.43 | *** join/#asterisk suma (n=sarisdjk@cm115.omega176.maxonline.com.sg) |
| 16:24.55 | suma | hi can i use PRI card as BRI card ? |
| 16:26.58 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 16:27.12 | Frosh | Analog Telephony Device card? is that a name? |
| 16:27.13 | tzafrir_laptop | suma, sure |
| 16:27.43 | tzafrir_laptop | which type did you have in mind? do you have a specific card? |
| 16:27.54 | tzafrir_laptop | suma, oops, no |
| 16:28.12 | tzafrir_laptop | I have misread your question |
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| 16:35.07 | Frosh | what kind of computer is adequate to run asterisk? |
| 16:35.31 | gambolputty | my soekris net4801 works |
| 16:36.53 | demlak | my national geode with 233mhz works too =) |
| 16:37.38 | russellb | anything that runs linux pretty much :) |
| 16:38.13 | demlak | depends on your needs... if you want to have 20 calls at the same time... youīll need more cpu and ram |
| 16:38.25 | demlak | and depends on used codec, etc,.. etc.. |
| 16:38.39 | *** join/#asterisk eltech (n=eltech@ool-457c94a3.dyn.optonline.net) |
| 16:38.47 | Frosh | how much cpu and ram for 20 calls? |
| 16:39.13 | demlak | depends on codec... on protocoll, etc... |
| 16:40.06 | demlak | i just know that it depends on those facts.. i donīt know how much you need for it |
| 16:40.08 | demlak | =) |
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| 16:43.50 | blitzrage | gumstix work too |
| 16:44.11 | blitzrage | also my 2x quad-core xeons run it good too |
| 16:47.11 | russellb | heh |
| 16:47.18 | russellb | one handles a few more calls than the other |
| 16:47.50 | demlak | hmm.. still no idea how to fix my voicemail timestamp problem.. |
| 16:49.04 | demlak | repeat: hmm.. voicemail timestamp is not correct... it is 2 hours behind the system time... i read about timezones.. but there is no /usr/share/zoneinfo/ on my minimal system... |
| 16:49.38 | demlak | where to look? what to configure? |
| 17:02.30 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 17:08.40 | LeddyHM | install tzdata |
| 17:09.06 | *** join/#asterisk saftsack (n=saftsack@pD9E06B9B.dip.t-dialin.net) |
| 17:09.32 | LeddyHM | the right one goes to /etc/localtime |
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| 17:11.07 | demlak | i canīt install more tools |
| 17:11.20 | demlak | itīs a fli4l (one disc router) |
| 17:11.32 | LeddyHM | then create a virtual install |
| 17:11.43 | demlak | ? |
| 17:11.47 | LeddyHM | and copy the right file over |
| 17:12.31 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 17:16.57 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
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| 17:20.23 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 17:22.26 | *** join/#asterisk eltech (n=eltech@ool-457c94a3.dyn.optonline.net) |
| 17:23.17 | Frosh | how much does a HiPath 5000 cost? |
| 17:29.41 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
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| 17:38.41 | demlak | just copying the file "/usr/share/Europe/Berlin", using "tz=localtime24" and "localtime24==Europe/Berlin|'vm-received' q 'digits/at' H N 'hours' |
| 17:38.46 | demlak | gna |
| 17:39.01 | demlak | just copying the file "/usr/share/Europe/Berlin", using "tz=localtime24" and "localtime24=Europe/Berlin|'vm-received' q 'digits/at' H N 'hours'" in voicemail.conf doesnīt help.. |
| 17:39.15 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 17:39.22 | demlak | and yes.. i did a reload =) |
| 17:40.53 | demlak | btw.. date says: Sat Apr 28 19:40:30 MESZ 2007 |
| 17:40.56 | demlak | if this helps |
| 17:47.39 | *** join/#asterisk JunK-Y (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
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| 18:12.29 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 18:12.29 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 18:13.53 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 18:16.18 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:16.47 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 18:19.05 | *** join/#asterisk marcan (n=marcanso@160.10.7.121) |
| 18:24.10 | *** join/#asterisk anthm (n=anthm@m010f36d0.tmodns.net) |
| 18:24.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:33.54 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-e45c7dfad9a85e27) |
| 18:33.54 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 18:36.37 | Qwell | Cresl1n: !!! |
| 18:36.45 | Cresl1n | Qwell!!! |
| 18:38.01 | Juggie | Qwell: http://home.donnyk.ca:9999/svnstats |
| 18:38.05 | Qwell | saw |
| 18:38.11 | Juggie | yah but its all stats now |
| 18:38.18 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@ip-41-112.sn1.eutelia.it) |
| 18:38.22 | Juggie | wrote a script last night to do 1.2 1.4 trunk for asterisk/zaptel/libpri |
| 18:38.24 | d4rkst4r75 | hello to all |
| 18:38.29 | Juggie | and built it all overnight. |
| 18:39.49 | d4rkst4r75 | can i ask some questions about asterisk? |
| 18:41.14 | *** join/#asterisk ljd (n=ljd@nelug/coreteam/luisjose) |
| 18:41.50 | Sweeper | NO |
| 18:41.55 | Sweeper | THE FIRST RULE OF ASTERISK |
| 18:42.01 | Sweeper | IS THAT ONE DOES NOT TALK ABOUT ASTERISK |
| 18:42.19 | *** join/#asterisk mrichmanM (n=richmanm@74.93.100.210) |
| 18:42.19 | russellb | lol |
| 18:42.23 | d4rkst4r75 | Lool |
| 18:42.28 | Juggie | russellb, pm. |
| 18:42.33 | russellb | k |
| 18:42.57 | Sweeper | seriously, all we discuss in this channel is jelly beans and cell phones |
| 18:43.41 | d4rkst4r75 | i want to ask if chan_dialogic is supported only in the enterprise edition of * |
| 18:43.50 | *** join/#asterisk blmm (n=rehn@adsl-75-61-108-129.dsl.pltn13.sbcglobal.net) |
| 18:44.11 | Cresl1n | I thnk so |
| 18:44.30 | russellb | I don't think chan_dialogic was ever released |
| 18:44.40 | d4rkst4r75 | i hate dialogic boards |
| 18:45.30 | d4rkst4r75 | but older customers have them and i've to plan if i can use them or if i've to suggest to change |
| 18:45.48 | russellb | suggest change i believe. |
| 18:46.10 | d4rkst4r75 | yes russelb, I think you're right |
| 18:46.29 | d4rkst4r75 | i've developed a dialogic driver for bayonne |
| 18:46.44 | d4rkst4r75 | and that boards are really beasts |
| 18:47.07 | russellb | d4rkst4r75: The guy that wrote chan_dialogic works for Digium now, you can find him on IRC as Deeewayne or d3wayne |
| 18:47.33 | d4rkst4r75 | thankyou russelb |
| 18:48.36 | blmm | Is ipkall having problems right now or is it just me? It calls my server when a call arrives, but their server then seems not to respond. |
| 18:49.08 | tuxick | Sweeper: i bet you're the type to get all the green jelleybeans |
| 18:49.28 | Sweeper | tuxick: actually, I like the black ones |
| 18:49.39 | tuxick | i was close :) |
| 18:49.40 | Sweeper | licorice <3 |
| 18:49.50 | Sweeper | but I don't really pick them out |
| 18:50.26 | blmm | Asterisk retransmits a message containing "CSeq: 102 INVITE" to them a few times. |
| 18:52.24 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:56.06 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 18:56.33 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 18:56.51 | jazzanova | i am able to call out using my vonage account, but not to receive the call. asterisk doesn't pick up. I have section [vonage] type=user, context=default |
| 18:56.57 | jazzanova | in sip.conf |
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| 19:23.08 | etfonhomey | Regarding presence, what is the setting that I need to change to decrease the amount of time that an extension is recognized as "Offline" when the extenson's (phone's) cable is unplugged? |
| 19:26.33 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 19:32.31 | drfreeze | Hello |
| 19:34.49 | drfreeze | I just added a TDM04B card (the second), added the fxsks=5 to /etc/zaptel.conf, ran ztcfg, and restarted zap inside asterisk, and restart asterisk. |
| 19:35.06 | drfreeze | But, now there are no incoming calls shown. |
| 19:35.57 | drfreeze | Anyone know why? |
| 19:38.11 | drfreeze | never mind now. It wasn't happy that channel range in zapata.conf didn't agree with the number defined in zaptel.conf |
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| 20:18.06 | *** join/#asterisk NirS (n=Nir@84.94.74.97.cable.012.net.il) |
| 20:18.06 | NirS | hey all |
| 20:18.07 | NirS | wassup ? |
| 20:18.07 | NirS | anybody home ? |
| 20:18.07 | NirS | anyone has an idea what this means: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. ? |
| 20:18.11 | russellb | NirS: upgrade to 1.4.4 |
| 20:18.11 | NirS | I'm using 1.2.18 |
| 20:18.11 | NirS | what's wrong with that ? |
| 20:18.11 | russellb | oh |
| 20:18.11 | russellb | hm |
| 20:18.12 | NirS | both boxes are 1.2.18 |
| 20:18.12 | russellb | then you'll need to upgrade to the latest code in the 1.2 branch |
| 20:18.12 | NirS | sending an IAX call from one box to the other, this is what I get |
| 20:18.12 | russellb | svn co http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
| 20:18.13 | NirS | bug in 1.2.x branch ? |
| 20:18.13 | russellb | well, it's fixed now |
| 20:18.14 | russellb | it was caused by a change that fixed a different bug, but caused that one |
| 20:18.14 | NirS | great, who in his right mind authorized a release with a broken IAX2 channel ? |
| 20:18.14 | russellb | .... |
| 20:18.14 | russellb | i told you how to fix it, don't be a dick |
| 20:18.15 | NirS | I'm kidding |
| 20:18.15 | NirS | dude, common, I had schuyler sleep at my place, I'm not that kind'a dick |
| 20:18.15 | russellb | k :) |
| 20:18.16 | NirS | ok, thanks |
| 20:18.17 | NirS | be with you shortly |
| 20:18.17 | Cresl1n | NirS!!! |
| 20:18.17 | Cresl1n | hey |
| 20:19.11 | blitzrage | Cresl1n: !!! |
| 20:19.11 | NirS | hey cres, do I know you ? |
| 20:19.11 | Cresl1n | yeah, I think I've met you once or twice, either at a trade show or in HSV |
| 20:19.11 | Cresl1n | you're Nir, from israel, no? |
| 20:19.11 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 20:19.11 | NirS | yes |
| 20:19.11 | russellb | NirS: hey, sorry, no offense intended. I just deal with a lot of people complaining and soemtimes say things I shouldn't. :) |
| 20:19.11 | NirS | HSV ? |
| 20:19.11 | Cresl1n | huntsville |
| 20:19.11 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:19.12 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 20:19.23 | Cresl1n | I'm MattF |
| 20:19.43 | Cresl1n | probably don't remember me, but I've worked in tech support, development, and engineering |
| 20:20.25 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 20:20.25 | Cresl1n | (from Digium) |
| 20:34.23 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 20:34.23 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 20:40.05 | Juggie | on a SPA-3000 do both the fxs and fxo ports register to asterisk? or is one just a passthrough? |
| 20:40.22 | *** part/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
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| 20:41.07 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:41.39 | *** join/#asterisk karlhaines (n=karl@74.94.254.5) |
| 20:43.15 | antlers | Hi there! Is anyone using Nortel i2004 phones with Asterisk? Having an odd issue whereby the mute key mutes speaker AND mic |
| 20:45.08 | *** join/#asterisk fduplex (i=irc@CPE00a1b000eeb1-CM0012254495cc.cpe.net.cable.rogers.com) |
| 20:45.26 | carrar | I gave up using Nortel phones with Asterisk |
| 20:45.31 | carrar | too much of a pita |
| 20:46.28 | NirS | sorry, was afk |
| 20:46.46 | carrar | no excuse |
| 20:46.48 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@ip-41-112.sn1.eutelia.it) |
| 20:46.51 | carrar | You need to be here 24/7 |
| 20:47.19 | NirS | ha ha |
| 20:47.25 | NirS | can't I talk to the wife man ? |
| 20:47.30 | NirS | you need to get married ;-) |
| 20:47.36 | carrar | only for sex |
| 20:47.39 | carrar | thats ok |
| 20:47.47 | carrar | (yes I am married btw) |
| 20:53.13 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 20:54.07 | NirS | russell, any idea why DTMF don't pass via IAX2 ? |
| 20:54.46 | [TK]D-Fender | NirS, it probably didn't study enough.... |
| 20:55.01 | carrar | haha |
| 20:55.11 | carrar | thats so wrong |
| 20:55.16 | [TK]D-Fender | :D |
| 20:55.45 | NirS | good one |
| 20:55.49 | russellb | NirS: nope, it should work fine. |
| 20:55.50 | [TK]D-Fender | antlers, mutes speaker & mic huh... sounds like HOLD to me :) |
| 20:56.13 | *** join/#asterisk eltech (n=eltech@ool-457c94a3.dyn.optonline.net) |
| 20:56.22 | *** join/#asterisk rycar (n=rycar@adsl-75-15-181-202.dsl.bkfd14.sbcglobal.net) |
| 20:56.23 | NirS | cres, what do you do now ? |
| 20:56.29 | [TK]D-Fender | antlers, And indeed.... screw Nortel... |
| 20:56.54 | carrar | I had a company trying to use the notels with asterisk |
| 20:56.57 | rycar | how would I transfer a call to an extension and record that call to a file? |
| 20:57.04 | carrar | finally got them to switch to a normal phone |
| 20:58.01 | [TK]D-Fender | rycar, "show application monitor" |
| 20:58.09 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 20:58.34 | *** join/#asterisk pingwin (n=pingwin@74-128-196-201.dhcp.insightbb.com) |
| 20:59.06 | pingwin | Hi, would it be safe to assume that a T1 PRI is the most common VOIP trunk in the USA? |
| 20:59.16 | carrar | yes |
| 20:59.22 | pingwin | going to be talking to my manager monday and want to verify my facts :D |
| 20:59.24 | pingwin | thanks carrar |
| 21:00.13 | [TK]D-Fender | pingwin, T1 PRI is a DIGITAL direct link to the PSTN. This has absolutiely NOTHING to do with VoIP. |
| 21:00.35 | carrar | heh |
| 21:00.49 | rycar | so do I just turn monitor on somewhere at the beginning and then when the call is finished do something with the files? |
| 21:01.23 | pingwin | [TK]D-Fender thanks for the intel |
| 21:01.32 | [TK]D-Fender | rycar, go read the instructions on it, then visit the WIKI page on it |
| 21:01.56 | rycar | thanks |
| 21:02.47 | [TK]D-Fender | pingwin, To elaborate ; T1 is the base digital link (24 x 64kbit channels), PRI is the signalling over that which uses 23 B channel (voice), and 1 D channel (call signalling). |
| 21:03.09 | carrar | isdn |
| 21:03.22 | pingwin | k, and that's just the wire dropped to get from the box to the PSTN, like ethernet is to internet connection |
| 21:03.56 | [TK]D-Fender | pingwin, It si possibe to get a partial T1/PRI where not all of the B channels are avaialable for use which costs less (somewhat proportionately). |
| 21:04.19 | pingwin | yes I knew that, like you can buy a 1/4 T1 for your office network connection |
| 21:04.44 | carrar | s/network/pstn/ |
| 21:04.48 | pingwin | so I understand that portion better, thank you.... is this configuration tho the most common method? |
| 21:04.50 | [TK]D-Fender | pingwin, Voice is no different, and you can alos use part for voice, and the other for data. |
| 21:05.07 | pingwin | yes, the nicity for T1 :) |
| 21:05.07 | [TK]D-Fender | pingwin, It is definately the PREFERRED means, yes. |
| 21:05.28 | *** join/#asterisk andrewc (n=andrewc@74.93.100.210) |
| 21:05.38 | drfreeze | pingwin: and you don't have to mess with pots wiring hassles |
| 21:05.45 | pingwin | cool, we're looking at buying this connection and my manager is all about "what is the standard" :P |
| 21:05.46 | [TK]D-Fender | pingwin, PRI allow for full call progress monitoring (ringing, answered, hangup, early media, etc) |
| 21:06.05 | carrar | definately go with a pri |
| 21:06.12 | [TK]D-Fender | pingwin, Also nice that you don't have a small mount of wire going into your server. |
| 21:06.30 | carrar | with dull 10 digit DNIS |
| 21:06.32 | carrar | full |
| 21:06.40 | pingwin | hehe yes, especially since I have to do the work :P |
| 21:06.52 | jazzanova | fender: hi |
| 21:07.10 | jazzanova | fender: remember our yesterdays conversation ? |
| 21:09.24 | [TK]D-Fender | jazzanova, Nope, its past its "Best Before" date :) |
| 21:09.44 | [TK]D-Fender | pingwin, Indeed insist on CID/DID as full 10 digit. |
| 21:10.03 | jazzanova | fender: well, i am able to dial out using my vonage account, but not do dial in |
| 21:10.22 | jazzanova | fender: i see the incoming call in sip log, but asterisk is not picking up |
| 21:10.40 | jazzanova | fender: in fact, its dropping it right away, and I go to answering machine of vonage. |
| 21:10.59 | jazzanova | i don't even have the 3 rings, that i have when asterisk is off. |
| 21:14.29 | Zefk | Hi. I'm running asterisk 1.4.2 with B410P. An external ZAP call arrives into context "from-local-trunk" that only Answer and play MOH. If tha caller hangup, asterisk does not finalize tha call and continue to play MOH. Tested from internal with a SIP phone is working. Any hints? Thanks. |
| 21:14.37 | jazzanova | also, even though i get a 20 second re-registration timeout, asterisk tries to reregister every 15 seconds. |
| 21:16.11 | [TK]D-Fender | jazzanova, pastebin the CLI output of the incoming call attempt at verbose 10 and SIP debug enabled. |
| 21:16.39 | [TK]D-Fender | jazzanova, And pastebin your [general] section of sip.conf |
| 21:16.54 | jazzanova | where's pastebin ? |
| 21:17.14 | jazzanova | got it |
| 21:17.31 | [TK]D-Fender | pastebin.ca |
| 21:17.34 | jazzanova | k, give me a minute |
| 21:18.15 | LeddyHM | whew yardwork sucks |
| 21:20.38 | *** join/#asterisk ZefK (n=Zefk@wsc-fo.b.astral.ro) |
| 21:25.29 | jazzanova | fender: http://pastebin.ca/463241 |
| 21:26.17 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 21:26.54 | NirS | any ideas why I would be getting static on a meetme room ? |
| 21:27.13 | [TK]D-Fender | jazzanova, your server is behind NAT, correct? |
| 21:27.20 | jazzanova | its not |
| 21:27.39 | jazzanova | its a collocated box |
| 21:27.55 | jazzanova | archimedes.hypervolume.com |
| 21:28.32 | [TK]D-Fender | jazzanova, where is the inbound call attempt? I don't see an INVITE in there anywayere. |
| 21:28.43 | jazzanova | oh |
| 21:28.48 | *** join/#asterisk viking78 (n=viking78@66-168-102-94.dhcp.jcsn.tn.charter.com) |
| 21:29.07 | jazzanova | # |
| 21:29.07 | jazzanova | From: "BRIT. COLUMBIA " <sip:17782380221@69.59.226.11>;tag=2118 |
| 21:29.07 | jazzanova | # |
| 21:29.09 | jazzanova | 048754 |
| 21:29.22 | jazzanova | <sip:17782380221@69.59.226.11>;tag=2118 |
| 21:29.30 | jazzanova | this is the phone I am calling from. |
| 21:29.44 | [TK]D-Fender | thats part of an ACK, but I'm not seeing the invite that starts it |
| 21:29.53 | jazzanova | one moment. |
| 21:33.27 | NirS | anyone ever encountered a situation where a meetme room would generate just static noise, once there are 2 or more people in the room ? |
| 21:36.49 | jazzanova | fender: http://pastebin.ca/463259 |
| 21:38.13 | Cresl1n | NirS: maybe a transcoding problem :-) |
| 21:38.45 | russellb | sounds like it ... |
| 21:39.15 | Cresl1n | when did that meetme rewrite happen.... |
| 21:39.19 | Cresl1n | between 1.2 and 1.4 I think |
| 21:39.48 | russellb | rewrite? |
| 21:39.52 | Cresl1n | yeah |
| 21:39.52 | russellb | it was never rewritten ... |
| 21:40.03 | Cresl1n | so that it only transcodes where it happens |
| 21:40.13 | Cresl1n | s/where it happens/when necessary/ |
| 21:40.30 | russellb | Cresl1n: hm, guess i don't remember that change |
| 21:41.30 | Cresl1n | it used to transcode for every channel in the conference |
| 21:42.11 | russellb | you mean for only users not talking? |
| 21:42.22 | russellb | i mean only users talking .. |
| 21:42.34 | Cresl1n | basically |
| 21:42.45 | russellb | that was between 1.2 and 1.4, yeah |
| 21:42.50 | Cresl1n | wherever it could be optimized |
| 21:42.54 | russellb | right |
| 21:43.08 | russellb | and it's not enabled by default |
| 21:43.12 | russellb | it's an option to the app |
| 21:43.16 | Cresl1n | oooh |
| 21:43.18 | Cresl1n | maybe that's bad |
| 21:43.23 | russellb | not sure why, though |
| 21:43.46 | Cresl1n | yeah |
| 21:43.48 | Cresl1n | that's what I would do |
| 21:43.57 | Cresl1n | we need to make sure that it's stress tested |
| 21:44.06 | Cresl1n | so that if there are bugs in it, they are fixed |
| 21:44.10 | *** join/#asterisk andrewc (n=andrewc@74.93.100.210) |
| 21:44.26 | Cresl1n | that's the way it should operate anyways, so that it burns less CPU |
| 21:44.40 | Cresl1n | for all we know, it could be broken by now :-) |
| 21:44.51 | NirS | btw, guys, I have a bunch of patches I need to finish for the say.c file, it fixes the hebrew functions to work in a proper grammer |
| 21:44.57 | russellb | that's why we do it in trunk only :) |
| 21:45.05 | Cresl1n | ooh |
| 21:45.12 | Cresl1n | I likes the way you think :-D |
| 21:45.21 | NirS | I will also have professional recordings for that, so it would fit the Asterisk standard, and also complete asterisk hebrew recordings to match |
| 21:46.19 | Cresl1n | heh |
| 21:46.22 | Cresl1n | course it is |
| 21:46.33 | Cresl1n | Debian is the one true distribution :_) |
| 21:46.34 | NirS | but I hate debian ;-) |
| 21:46.52 | carrar | I'd use CentOS 4.4 |
| 21:46.54 | *** join/#asterisk anthm (n=anthm@rrcs-74-62-82-254.west.biz.rr.com) |
| 21:46.54 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:46.56 | carrar | if you are using 1.4 |
| 21:47.19 | carrar | although 5 might work fine too |
| 21:47.21 | jazzanova | nirs: try gentoo |
| 21:47.24 | NirS | yes, but CentOS 4.4 won't work with my 965 chipset properly |
| 21:47.30 | NirS | great Jazz |
| 21:47.43 | NirS | the only distro in the world that requires its own ISP ;-) |
| 21:47.56 | NirS | in Israel, internet can be somewhat fleemsy at times |
| 21:48.10 | jazzanova | nirs: are you from israel ? where exactly ? |
| 21:48.27 | NirS | I live in a small moshav called Udim |
| 21:48.31 | NirS | 25 minutes away from Tel aviv |
| 21:48.41 | jazzanova | cool |
| 21:48.42 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 21:48.55 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:48.56 | jazzanova | i used to live in nacrat illit |
| 21:49.06 | NirS | you mean nazarath |
| 21:49.22 | jazzanova | why is my vonage sip re-registering every 15 seconds ? |
| 21:49.32 | Cresl1n | I've been wanting to take a trip to israel for a while |
| 21:49.39 | Cresl1n | I would love to see the holy land |
| 21:49.44 | NirS | cres, schuyler is coming here in june, talk to him |
| 21:49.48 | NirS | he's coming for 3 months |
| 21:49.50 | Cresl1n | hrm |
| 21:49.51 | Cresl1n | wow |
| 21:50.07 | NirS | well, at least that's his plan as far as I know |
| 21:54.16 | jazzanova | can someone look at this and tell me why asterisks is not picking up the phone: http://pastebin.ca/463259 |
| 21:55.56 | NirS | ok, something is definitely broken here |
| 21:56.01 | NirS | I have an asterisk 1.2 SVN on one side |
| 21:56.08 | NirS | and an asterisk 1.4.4 on the other |
| 21:56.15 | NirS | meetme is running on the 1.4.4 |
| 21:56.22 | NirS | and I get full static |
| 21:56.31 | NirS | this doesn't make any sense, does it ? |
| 21:56.42 | jazzanova | can you match the versions ? |
| 21:56.49 | NirS | yes, sure |
| 21:58.31 | jazzanova | probably good idea. |
| 21:58.42 | NirS | well, lets match versions to 1.4.4 |
| 21:59.04 | NirS | if that doesn't work, I'll go back to 1.2.16, which worked fine |
| 21:59.15 | NirS | which I really don't want to do |
| 21:59.20 | jazzanova | do you have to recompile ? |
| 21:59.24 | NirS | yes |
| 21:59.34 | NirS | oh, you mean my code ? |
| 21:59.58 | jazzanova | can you install higher version from a binary ? |
| 22:00.11 | NirS | no, the machines are highly non-standard |
| 22:00.19 | NirS | I better compile on those silly boxes |
| 22:01.06 | jazzanova | check this out: |
| 22:01.09 | jazzanova | Apr 28 17:58:04 NOTICE[30795]: Outbound Registration: Expiry for sphone.vopr.vonage.net is 20 sec (Scheduling reregistration in 15999 ms) |
| 22:01.09 | jazzanova | Apr 28 17:58:20 NOTICE[30795]: -- Re-registration for 17787850134@sphone.vopr.vonage.net |
| 22:01.27 | jazzanova | only 16 seconds passed. |
| 22:01.34 | jazzanova | how many is 15999 ms? |
| 22:02.09 | NirS | that is weird |
| 22:02.48 | NirS | oh man |
| 22:03.03 | GreyFoxx | 15999/1000 = 15.9seconds |
| 22:03.03 | NirS | 1.4.4 noticed that I have pwlib on the box, and wants to compile chan_h323, how silly |
| 22:03.17 | GreyFoxx | well, 15.999 :) |
| 22:03.32 | jazzanova | ok, so why is it scheduling in 20 seconds, by re-registering in 16 ? |
| 22:03.45 | GreyFoxx | no cvlue |
| 22:03.46 | jazzanova | does it mean i have some kind of clock timer problem ? |
| 22:06.30 | r0d3nt | is it a vserver ? vm ?? |
| 22:07.02 | jazzanova | r0d3nt: are you asking me ? |
| 22:07.06 | *** join/#asterisk tomcontr3 (n=tomcontr@68-77-246-201.adsl.terra.cl) |
| 22:07.09 | tomcontr3 | hi |
| 22:07.24 | r0d3nt | jazzanova: yes |
| 22:07.27 | jazzanova | r0d3nt: i am running vserver on this machine. |
| 22:07.27 | jazzanova | yes |
| 22:07.39 | jazzanova | but, this is the main box, not a virtual machine. |
| 22:07.44 | tomcontr3 | I have just installed asterisl 1.2 las version, but Im having problems when I make calls to other extentions, I can hear them and they can hear me |
| 22:08.26 | jazzanova | my big problem is this: http://pastebin.ca/463259 asterisk is not pciking up the phone. |
| 22:08.45 | *** join/#asterisk nasls_lsa (n=chatzill@ppp046-032.dsl.hol.gr) |
| 22:09.27 | NirS | off topic |
| 22:09.48 | tomcontr3 | anyone? |
| 22:09.48 | NirS | anyone has the episodes to "The Dresden Files" and "Jake 2.0" somewhere for downloading? ;-) |
| 22:10.35 | SwK | does this like #thepiratebay.org |
| 22:10.45 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 22:11.18 | NirS | swk, what do you mean ? |
| 22:11.58 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 22:12.35 | Ifaistos | tomcontr3: Are you behind a NAT ? or same lan with the other phones ? |
| 22:12.47 | *** join/#asterisk DaveCanoe (n=Dave@adsl-065-007-135-002.sip.asm.bellsouth.net) |
| 22:13.43 | tomcontr3 | NAT |
| 22:13.56 | tomcontr3 | the server is not in my network |
| 22:14.06 | tomcontr3 | and Im behing a firewall |
| 22:14.22 | Ifaistos | tomcontr3: that's the problem |
| 22:14.33 | tomcontr3 | any Fix? |
| 22:15.43 | *** join/#asterisk connecta (n=Administ@76.23.188.72.cfl.res.rr.com) |
| 22:16.08 | tomcontr3 | but the server is on a public ip |
| 22:16.13 | tomcontr3 | no behind a nat |
| 22:16.19 | Ifaistos | sip and nat are not so good friends.... if your phone supports STUN use it |
| 22:16.33 | connecta | can someone explain to me what Libpri is NEEDED for (i know it's the PRI portion of asterisk). Do i need it on a purely SIP system? |
| 22:16.50 | Cresl1n | no |
| 22:16.57 | Cresl1n | PRI is for T1 cards |
| 22:17.05 | Cresl1n | or E1 cards |
| 22:17.09 | Cresl1n | not for SIP channels |
| 22:17.29 | connecta | k thanks |
| 22:17.40 | Ifaistos | os isdn BRI |
| 22:18.31 | tomcontr3 | im using eyeBeam softphone |
| 22:18.58 | tomcontr3 | the thign is that I can recive or make call, but I cant hear |
| 22:22.09 | connecta | is russel from asterisk dev here? |
| 22:27.53 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-40-188.socal.res.rr.com) |
| 22:28.31 | NirS | ok, now I'm at a stand still |
| 22:28.34 | NirS | nothing works |
| 22:28.50 | NirS | it looks like app_meetme is either broken, or something is totally fucked up |
| 22:29.10 | NirS | once I have 2 or more people inside a conference room, the only thing I get is static |
| 22:32.58 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 22:46.50 | NirS | ok |
| 22:46.53 | NirS | this makes no sense any more |
| 22:47.03 | NirS | something is totally fucked up around this one |
| 22:51.26 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 22:56.22 | *** join/#asterisk Strom_C (n=strom@135.196.213.180) |
| 23:00.35 | Juggie | could anyone recomend anything compareable to a SPA-3000/3102? W 1FXO/1FXS? |
| 23:05.16 | NirS | hey juggie |
| 23:05.30 | NirS | I know that GrandStream has the 488, but i heard it's a piece of shit |
| 23:09.30 | *** join/#asterisk Malawar (n=Malawar@adsl-70-141-10-32.dsl.sgnwmi.sbcglobal.net) |
| 23:09.34 | Malawar | does anyone here use voxee? |
| 23:09.45 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-64.dhcp.leds.al.charter.com) |
| 23:09.56 | Malawar | or have they used voxee in the past and fuond that they totally suck? :P |
| 23:10.50 | NirS | malawar, never used them |
| 23:10.57 | NirS | but I guess I never will now |
| 23:10.58 | NirS | :-) |
| 23:11.35 | Malawar | about 1 out of every 80 calls or so actually makes it through. |
| 23:12.00 | Malawar | so I think I need to find another outbound sip service. |
| 23:12.36 | Malawar | preferably one that is pay-as-you-go instead of monthly :P |
| 23:15.13 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 23:27.45 | *** join/#asterisk l2cache (n=admin@62.180.8.67.cfl.res.rr.com) |
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| 23:34.17 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 23:34.43 | Malawar | man |
| 23:34.47 | Malawar | voxee just doesn't work at all |
| 23:34.50 | Malawar | freedigits works fine :/ |