| 00:04.59 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 00:15.02 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 00:44.54 | *** join/#asterisk Thazza (n=no@229.9.233.220.exetel.com.au) |
| 00:52.04 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 00:56.47 | comprookie2000 | TheSov what brand of phone |
| 01:09.35 | TheSov | im calling from my cell phone to the asterisk box via sip |
| 01:15.56 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 01:17.32 | jameswf-home | I wonder if alison finds it disturbing for $30 people an make her say whatever.... or les with a little audio remix |
| 01:18.39 | TheSov | alison? |
| 01:21.01 | *** join/#asterisk Badran (n=mohammed@41.235.119.202) |
| 01:21.37 | *** part/#asterisk Badran (n=mohammed@41.235.119.202) |
| 01:38.15 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 01:43.24 | *** join/#asterisk ccvp (n=ccvp@user-24-214-126-81.knology.net) |
| 01:44.37 | *** join/#asterisk wastrel (n=wastrel@nylug/member/wastrel) |
| 02:18.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:28.48 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:31.17 | riddlebox | TheSov, alison is the voice of asterisk |
| 02:31.34 | TheSov | oh yeah |
| 02:32.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:32.08 | TheSov | do all voice recordings for playback need to be in GSM |
| 02:32.11 | TheSov | or can they be wave? |
| 02:32.24 | riddlebox | you can get them in wav as well |
| 02:32.35 | TheSov | every time i try to record in gsm it sounds really slow and funny |
| 02:32.51 | TheSov | so im wondering if i can record in wav and simply copy it to the pbx |
| 02:34.59 | frogonwheels | TheSov: just be careful - it has to be a very specific type of wave file. |
| 02:35.29 | frogonwheels | I think it's type 1 which is the microsoft ADPCM |
| 02:35.32 | TheSov | i see it has to be 8khz and PCM |
| 02:36.14 | frogonwheels | Ive just been trying to get a wave file extracted from an email and put into the INBOX for voicemail. |
| 02:36.23 | *** join/#asterisk xanderp (n=pphillip@98.220.163.8) |
| 02:36.36 | frogonwheels | Made more interesting by the fact that sox wouldn't work for me :) |
| 02:37.27 | xanderp | anyone here have a VTA-VR working with asterisk? Mine keeps locking up after a few hours, and I can't find anything online to fix it. (others post that they have similar problems.) |
| 02:37.50 | TheSov | man using an ivr to get rid of telemarketers is awesome |
| 02:38.13 | xanderp | TheSov: or putting your number on the national do not call list... either way... |
| 02:38.26 | TheSov | that thing lists you for charities |
| 02:38.42 | TheSov | and of course political donatios |
| 02:38.44 | TheSov | donations |
| 02:38.52 | xanderp | I think i've only gotten like 2 calls in 6 years. |
| 02:38.58 | xanderp | not to shabby |
| 02:39.31 | xanderp | (and one of those was for fire department charity, and the other was police I think... ) |
| 02:40.12 | TheSov | right |
| 02:40.52 | TheSov | so anyone know why the WaitExten function would only catch the first DTMF tone and ignore rest? |
| 02:41.14 | TheSov | i called on my cell fone and dialed a bunch of numbers they seem to be muted |
| 02:41.27 | TheSov | well theirs a pause anyway |
| 02:45.21 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.133.135) |
| 02:48.46 | *** join/#asterisk Keizer (n=keizer@74.5.187.133) |
| 02:49.03 | Keizer | Does Asterisk require Alsa to work? |
| 02:49.09 | TheSov | no |
| 02:49.11 | drmessano | no |
| 02:49.21 | Keizer | Ok |
| 02:49.34 | Keizer | I tried checking out 1.6 branch from svn but I am getting compile errors |
| 02:50.12 | Keizer | I'll try the 1.6 tar file to see if I still get compile errors |
| 02:50.24 | drmessano | Why dont you try 1.4 first |
| 02:50.31 | Keizer | I have tried 1.4 |
| 02:50.37 | Keizer | I want to experiment |
| 02:50.50 | drmessano | ok |
| 02:51.27 | Keizer | This is all the better because I was wanting to use mysql addon module |
| 03:07.56 | delparnel | How can I find out the IP address of a phone connected to my asterisk? |
| 03:09.22 | *** join/#asterisk SureV (n=surev@205.144.218.234) |
| 03:11.56 | Keizer | delparnel: If you lauch: asterisk -vvvr you can see what asterisk is doing |
| 03:12.26 | SureV | Hi All, anyone ever used a Dlink DVG-2001S before? |
| 03:12.29 | TheSov | actually u should launch asterisk with -vvvvvvvvvvvvvvvvvrc |
| 03:12.33 | SureV | its a 1 port FXS |
| 03:13.16 | SureV | voy pomnimyo ba ruski |
| 03:13.20 | Keizer | I thought -c was for debugging |
| 03:13.40 | TheSov | no thats console |
| 03:13.50 | TheSov | cli console |
| 03:14.07 | TheSov | i think... |
| 03:14.42 | TheSov | just add a whole mess of v's u cant go wrong |
| 03:19.55 | rkeene | core set verbose 10 :-P |
| 03:24.38 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 03:33.32 | *** join/#asterisk jmhunter (n=Snak@72.14.86.68) |
| 03:33.33 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 03:33.44 | jmhunter | hello |
| 03:37.52 | jmhunter | good evening all.. long time no chat at the ol' #asterisk |
| 03:41.13 | *** join/#asterisk hohum (n=dcorbe@70.6.162.128) |
| 03:41.36 | TheSov | howdy hunter |
| 03:42.21 | jmhunter | :) |
| 03:58.18 | SureV | anyone here know of a good IAX2 FXO box I can distribute to various offices around the globe? |
| 03:59.27 | SureV | I was going to go with a SIP unit like the Dlink DVG-2001S or a PAP2, but would prefer an IAX unit instead of a SIP |
| 03:59.56 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-195-124.tx.res.rr.com) |
| 03:59.57 | SureV | are there actually any good IAX FXO devices that any of you know of? |
| 04:04.31 | TheSov | i hear iax is superior in every way |
| 04:04.38 | TheSov | but SIP is a standard |
| 04:04.46 | TheSov | so it kinda wins by default |
| 04:05.15 | SureV | yeah, that is what I understand too, stinks, I didn;t want to use NAT and STUN and all that rubbish when IAX is the native tongue |
| 04:05.30 | TheSov | forward enough ports and u dont need a stun |
| 04:06.22 | SureV | well, I gueeesss, but kinda lame when I could just use IAX to surpass all that rubbish :-) |
| 04:07.06 | TheSov | i wonder the polycom soundpoints support iax |
| 04:07.40 | SureV | hmm, not familiar with those, but usually polycom ends up being waaaay over priced |
| 04:07.47 | SureV | will check it out real wuick |
| 04:07.53 | SureV | because I would rather pay more and get |
| 04:07.56 | SureV | good stuff |
| 04:08.01 | SureV | and use SIP |
| 04:08.08 | SureV | haha, I don't want to sip, I want to chug |
| 04:09.13 | TheSov | the polycom's do support iax i was told they are a nightmare for nat networks |
| 04:09.22 | TheSov | thats what my company intends to use |
| 04:10.02 | jmhunter | !seen bkw_ |
| 04:10.06 | SureV | hmm, but Polycom isn't just an FXO |
| 04:10.06 | TheSov | the sip portion anyway |
| 04:10.43 | SureV | which is what I would prefer as I was to have the existing PSTN phones just connected in the wiring closet to the FXO |
| 04:10.58 | TheSov | why on earth would u setup a voip pbx and use an FXO |
| 04:11.08 | drmessano | Why not? |
| 04:11.19 | SureV | for reasons beyond my willingness to comment, its not my call |
| 04:11.20 | SureV | sadly |
| 04:11.30 | SureV | I would just put a polycom on every desk |
| 04:11.50 | drmessano | You don't have to use an ITSP to realize benefit from an IP PBX |
| 04:11.55 | SureV | but *management* doesn't want to invest that far when they can get *enough* results with |
| 04:11.57 | SureV | FXOs |
| 04:12.02 | TheSov | i mean for a small company its very cost effective to lose all pots lines and go straigh with voip/voip |
| 04:12.28 | drmessano | Not necessarily |
| 04:13.08 | SureV | they want to leverage their existing wiring and phones with minimal |
| 04:13.15 | SureV | impact on users |
| 04:14.09 | SureV | supposed to be transparent enough to not even be noticed, for connecting |
| 04:14.16 | SureV | already existing branch offices |
| 04:14.25 | SureV | to the internal phone system |
| 04:14.29 | SureV | of HQ |
| 04:14.36 | SureV | so everything is local |
| 04:14.53 | drmessano | As far as IAX vs SIP goes for NAT.. if you can't make SIP work over NAT, you're doing it wrong |
| 04:14.59 | drmessano | Asterisk gives you all you need |
| 04:15.56 | TheSov | for some reason i cant get local phones connected to asterisk without turning nat on |
| 04:15.59 | TheSov | and registering |
| 04:16.05 | SureV | I'm not asking for HOW to make sip work, I know it can, but the protocol suck when you compare it to IAX and the fact that IAX is asterisk's own protocol |
| 04:16.13 | TheSov | might be that my softphone sucks |
| 04:16.24 | drmessano | IAX has it's shortcomings |
| 04:16.36 | SureV | sure, so does danny devito, but that isn't my point, I'm |
| 04:16.39 | drmessano | It's a waste to look for IAX devices when SIP devices work just fine |
| 04:16.44 | SureV | trying to find if there are FXO's that support IAX |
| 04:17.32 | SureV | I'm on a fact finding mission for bedgeting guys, not a techinical crusade to champion IAX vs SIP |
| 04:18.08 | SureV | they want to know the options, and I want to do my best to explore that, |
| 04:18.49 | SureV | like I said earlier, if it were my call I would put SIP or IAX phones right on the desktops |
| 04:18.51 | SureV | but not my call |
| 04:19.05 | drmessano | Well, if you ask for something that makes no sense, someone is going to comment.. if you can't appreciate that, then you really shouldn't be asking |
| 04:19.39 | SureV | you're a froot loop man, how does IAX not make sense? Digium has it all wrong and you're right? |
| 04:19.53 | SureV | sure, maybe pref one way or the other, |
| 04:19.56 | drmessano | IAX + FXO is a stupid idea |
| 04:19.59 | SureV | but not make sense at all, c'mon |
| 04:20.24 | drmessano | IAX has it's purpose, and it's not as a protocol for devices |
| 04:20.27 | SureV | ok, agree to disagree, cause it would still be better for multiple levels of nat |
| 04:20.32 | drmessano | It's not practical |
| 04:20.41 | SureV | sure, between boxes is good too, |
| 04:20.46 | SureV | I agree on that one for sure |
| 04:20.49 | drmessano | SIP will work over multiple NATs |
| 04:21.02 | SureV | and I didn't say it couldn't |
| 04:21.17 | jameswf-home | drmessano: http://freepbx.org/support/documentation/howtos/how-to-using-freepbx-back-end-for-trixbox-ce-authentication |
| 04:21.36 | SureV | just saying that IAX on devices doesn't make sense, is a bit of a stretch |
| 04:22.10 | jmhunter | !seen bkw |
| 04:22.20 | jmhunter | !seen _bkw |
| 04:22.23 | drmessano | SureV: The fact that you are even asking about IAX FXO devices tells me you don't know a lot about Asterisk, and I suggest you do some reading |
| 04:22.45 | drmessano | SureV: Maybe you would understand the place that each of the protocols has |
| 04:22.55 | drmessano | jameswf-home: I read that earlier, good stuff |
| 04:23.18 | jmhunter | !seen kram |
| 04:23.20 | SureV | I know plenty about asterisk, and work daily with it, the fact that there are soft phones and hard phones shows its capable and perfectly usable on devices |
| 04:23.34 | *** join/#asterisk Bonix-BR (i=500@189.52.0.234) |
| 04:23.46 | jmhunter | !seen pfn |
| 04:23.56 | jmhunter | nicklist! |
| 04:23.59 | jameswf-home | !seen jbit |
| 04:24.03 | jmhunter | !nicklist |
| 04:24.09 | jameswf-home | !seen jbot |
| 04:24.13 | SureV | just anyone wanting to connect locations would typically just put out another server for each location |
| 04:24.24 | jameswf-home | ~seen jbot |
| 04:24.26 | jbot | jbot is currently on #asterisk-doc (2d 8h 41m 5s) ##t42 (2d 8h 41m 5s) #how (2d 8h 41m 5s) #flyspray (2d 8h 41m 5s) #asterisk-dev (2d 8h 41m 5s) #asterisk (2d 8h 41m 5s) #byumug (2d 8h 41m 5s) #va (2d 8h 41m 5s) #orkut (2d 8h 41m 5s) #nslu2-linux (2d 8h 41m 5s) ##ducleague (2d 8h 41m 5s) #storm (2d 8h 41m 5s) #uphpu (2d 8h 41m 5s) #aegis (2d 8h ... |
| 04:24.40 | jmhunter | ~seen pfn |
| 04:24.41 | jbot | pfn is currently on #asterisk #tomcat. Has said a total of 68 messages. Is idling for 1d 3h 58m 54s, last said: 'spring, struts, etc. should support those kinda tasks pretty easily'. |
| 04:24.41 | SureV | and as I said before, I'm not fighting FOR IAX/FXO, but looking to include it in a RFP |
| 04:24.52 | jameswf-home | ~seen yourmom |
| 04:24.53 | jbot | yourmom <n=185c59f2@bz.bzflag.bz> was last seen on IRC in channel #bzflag, 447d 23h 27m 3s ago, saying: 'hello?'. |
| 04:25.08 | jmhunter | ~seen bkw_ |
| 04:25.09 | jbot | bkw_ <n=brian@ppp-70-128-106-102.dsl.tulsok.swbell.net> was last seen on IRC in channel #asterisk, 12d 14h 2m 47s ago, saying: 'thats ok.. would do it again in a heart beat'. |
| 04:25.16 | SureV | so then I just leave it and say, oh well there aren't any, good enough |
| 04:25.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:26.23 | TheSov | well im working on an asterisk system so i can get like 10 voip lines from viatalk and setup my office with a whole new phone system |
| 04:26.37 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:27.23 | TheSov | so our asterisk box is currently a virtualization of slackware running an internal ip with a 1:1 mapping to the outside so i dont have to deal with the port forwarding i just gotta get the Dialplan right |
| 04:28.05 | *** join/#asterisk littleball (n=littleba@bb121-7-21-122.singnet.com.sg) |
| 04:28.40 | SureV | TheSov, if you're going to get prompts done, I can highly recommend voicevector for the audio prompts |
| 04:28.54 | TheSov | we got a girl from the office to do it |
| 04:29.07 | SureV | haha, cool, that works too |
| 04:29.09 | TheSov | but the stupid internal record feature sucks |
| 04:29.17 | SureV | yeah, that is for sure |
| 04:29.21 | TheSov | i have to record as waves and do a convertion |
| 04:29.28 | hesco | how do I map a ttyIAX0 context to use that iaxmodem and my diamondcard account to place an outgoing fax call? |
| 04:36.06 | *** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com) |
| 04:36.32 | *** join/#asterisk steliosk (n=Stelios@athedsl-105864.home.otenet.gr) |
| 04:38.15 | TheSov | i was looking up sip providers, a whole bunch of them get upset when u say "i dont want your equiptmentm, we have an asterisk box, give me 10 lines" and then claim they dont give registration info out |
| 04:38.46 | *** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com) |
| 04:40.32 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 04:40.54 | *** join/#asterisk adjohn (n=adjohn@i121-115-187-159.s05.a013.ap.plala.or.jp) |
| 04:41.34 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 04:48.06 | hesco | this whole project runs contrary to the interests of their existing business model, apparently. |
| 04:48.22 | SureV | what country you in Sov? |
| 04:49.13 | SureV | TheSov: What country are you in? |
| 04:50.18 | hesco | conceptually, I do I instruct an iaxmodem to use a diamondcard account to deliver its traffic to the pots? |
| 04:52.09 | hesco | I've got hylafax's sendfax making reports of failed calls appear on the asterisk console. Great progress over yesterday. |
| 04:52.46 | hesco | But my calls going through the iaxmodem context, are not making phones ring in the outside world, yet. |
| 04:52.58 | hesco | That is the breakthrough I'm looking for. |
| 04:54.45 | SureV | hesco, what OS? |
| 04:55.42 | hesco | Debian |
| 04:55.52 | SureV | version? |
| 04:55.56 | hesco | I get chan_iax2.c:7111 socket_read: Call rejected by 127.0.0.1: Busy |
| 04:55.56 | hesco | <PROTECTED> |
| 04:56.16 | hesco | new, net install three or four months ago |
| 04:56.31 | hesco | iaxmodem built from source a couple of days ago |
| 04:56.49 | SureV | can you pastebin your dialplan? |
| 04:57.00 | hesco | hylafax and asterisk from apt-get install |
| 04:57.06 | SureV | did you put it alltogether yourself, |
| 04:57.11 | hesco | yes |
| 04:57.13 | SureV | or use asterfax or other package |
| 04:57.25 | hesco | I don't believe so. |
| 04:57.51 | SureV | ok, is this a production box you're wanting to add fax fun to? |
| 04:58.28 | SureV | so you've never managed to get a fax call to the outside worl |
| 04:58.34 | SureV | d? |
| 04:58.35 | hesco | yes. Its already being used for automated outcalls from time to time. But it is not processing incoming |
| 04:58.47 | hesco | exactly. never. |
| 04:58.59 | hesco | not off this box, anyhow. |
| 04:59.34 | hesco | four years ago I admistered a hylafax server which was sending hundreds of faxes a week for a previous employer. |
| 04:59.57 | SureV | could you pastebin the dialplan for the fax |
| 05:00.12 | hesco | While I've watched asterisk for years, my first successful prototype was only last year. |
| 05:00.19 | hesco | coming your way. |
| 05:00.59 | SureV | asterisk certainly kicks some serious a$$ |
| 05:01.02 | SureV | for sure |
| 05:01.25 | *** join/#asterisk talntid (n=swarm@66.208.251.166) |
| 05:02.24 | SureV | just recently did a rip out of 1200 centrex extensions in a Fortune 500 company to replace with a series of Asterisk boxes |
| 05:02.31 | SureV | which handeled the task beautifully |
| 05:03.59 | hesco | SureV: http://pastebin.com/d51a2c615 |
| 05:06.12 | SureV | what debug level are you using in the console? |
| 05:06.18 | SureV | 2 or so? |
| 05:07.00 | hesco | I think I invoked the console with four or five verbose, but I have not set a debug inside the console. How is that done? |
| 05:08.06 | hesco | ok, Debugging level set to 3, |
| 05:08.11 | hesco | Testing again, now |
| 05:08.20 | SureV | what is the output when you try to ring it? |
| 05:09.08 | hesco | Mar 22 22:13:53 WARNING[3775]: chan_iax2.c:7111 socket_read: Call rejected by 127.0.0.1: Busy |
| 05:09.08 | hesco | <PROTECTED> |
| 05:09.09 | hesco | <PROTECTED> |
| 05:09.34 | hesco | . |
| 05:09.38 | hesco | that follos: |
| 05:09.41 | hesco | <PROTECTED> |
| 05:09.41 | hesco | <PROTECTED> |
| 05:10.18 | hesco | How does the ttyIAX0 know my account details to use the diamondcard account? |
| 05:12.01 | SureV | checking some stuff out, could be a moment |
| 05:12.38 | *** join/#asterisk JAHS (n=JAHS@190.33.255.59) |
| 05:12.49 | JAHS | hi buddies .. |
| 05:13.00 | hesco | full details here: http://pastebin.com/d3ddd930e |
| 05:13.05 | JAHS | I'd appreciate any help on a big issue .. I'm getting crazy .. |
| 05:13.13 | hesco | no problem, SureV. Thanks for your help on this. |
| 05:13.45 | SureV | I'm a smidgen lost on what you meant by "How does the ttyIAX0 know my account details to use the diamondcard account?" |
| 05:13.46 | JAHS | My TE420 card is detected .. I can see it using lspci command .. BUT it's not getting up |
| 05:13.48 | hesco | getting? How could one cope with this world if they were not there already? |
| 05:14.35 | hesco | To reach the POTS telephone network, I'm been using an iax provider (diamondcard, in this case). |
| 05:15.00 | SureV | meaning, how do you tell ttyIAX0 to use the diamond card trunk? |
| 05:15.11 | SureV | is it a sip or IAX2 trunk? |
| 05:15.13 | hesco | exactly, thank you! |
| 05:15.19 | hesco | iax2 |
| 05:15.27 | hesco | diamondcard |
| 05:15.28 | JAHS | I only see once I run ztcfg -vvv this message "ZT_CONFIG ..no such device or address " |
| 05:16.06 | SureV | you have spandsp installed all happy too? |
| 05:16.21 | hesco | no complaints from what I've seen. |
| 05:16.24 | SureV | ok |
| 05:16.55 | hesco | I started iaxmodem and the faxgetty at a bash prompt and left a console to monitor their messages. |
| 05:16.59 | JAHS | it happens on my PCIe card .. |
| 05:17.10 | JAHS | any help please? |
| 05:17.25 | *** join/#asterisk north43 (n=north43@CPE00112f561810-CM000f212f9889.cpe.net.cable.rogers.com) |
| 05:17.43 | hesco | JAHS: do you need to install zaptel? |
| 05:17.57 | JAHS | I just installed libpri, zaptel and asterisk .. |
| 05:18.09 | JAHS | and I configured a TE420 card on zaptel.conf |
| 05:18.20 | hesco | understand that that was a silly wild ass guess as I'm new to this asterisk stuff myself. |
| 05:19.17 | JAHS | excuse me, but when I try to get it up .. it shows this message .."ZT_CONFIG ..no such device or address " |
| 05:19.40 | hesco | So, how do I tell ttyIAX0 to use the diamond card trunk? That strikes me as exactly my question. |
| 05:20.15 | *** part/#asterisk north43 (n=north43@CPE00112f561810-CM000f212f9889.cpe.net.cable.rogers.com) |
| 05:20.21 | hesco | is zaptel a daemon which has to be running? I don't use it, so I'm only guessing from your error message. |
| 05:20.40 | SureV | hesco: can you pastebin the relevant parts of your IAX2.conf with private data changed/removed |
| 05:20.56 | JAHS | yes .. I have all modules loaded .. |
| 05:21.02 | hesco | device suggests that its not finding a node in /dev/ to connect to. |
| 05:21.20 | hesco | SureV: thanks. Coming your way. |
| 05:21.25 | JAHS | hmm .. |
| 05:21.57 | JAHS | when I type "lspci" it shows the card .. |
| 05:22.21 | JAHS | is it related? I guess the OS is detecting the card .. but it's not getting up .. |
| 05:22.33 | JAHS | what could I try to check what's wrong ..? |
| 05:23.04 | SureV | hesco, hablas spanish? |
| 05:23.17 | SureV | I have a couple of spanish links that may help |
| 05:23.19 | JAHS | yo hablo espannol tamben .. |
| 05:23.33 | JAHS | I'm from Colombia .. |
| 05:23.46 | SureV | verdad? |
| 05:23.49 | SureV | bueno |
| 05:23.59 | SureV | I'm just a gringo with a decade of life in mexico |
| 05:24.31 | hesco | SureV: http://pastebin.com/d2b436627 |
| 05:25.16 | adorah | Hi Has anypne here experienced to run an asterisk machine over WI Max? |
| 05:26.09 | SureV | JAHS, En Colombia,usan Utds. la palabra "Gringo"? |
| 05:26.23 | JAHS | algunas veces . |
| 05:26.24 | hesco | I'm one of those dumb americans who failed to ever learn my own language fully, much less anyone else's, in spite of five years in French class, three in a German class and a semester in a Latin class. |
| 05:26.43 | JAHS | tres bien! |
| 05:26.45 | SureV | , so only some times used |
| 05:26.49 | SureV | but its know right? |
| 05:26.55 | SureV | hesco, LOL |
| 05:27.00 | SureV | yeah, I understand |
| 05:27.23 | JAHS | Surev, donde vivis? |
| 05:27.26 | hesco | But I'd be willing to see what you got and tug on a friends sleeve for help if you got nothing else for me. |
| 05:27.44 | SureV | I took spansih in school and then a US diplomat course in French, and then 4 years of German right before 10 years in Mexico |
| 05:27.54 | JAHS | wow .. |
| 05:28.04 | SureV | JAHS, ahora, yo vivo en Estados Unidos |
| 05:28.06 | JAHS | you're almost a linguist hesco |
| 05:28.06 | hesco | it was the 10 years immersed that did the trick, eh? |
| 05:28.21 | hesco | hardly |
| 05:28.26 | SureV | hesco, yeah, for sure, High School Spanish was a waste of time |
| 05:28.38 | SureV | haha |
| 05:28.44 | hesco | linguist? does not mean I can write in both bash and in perl? |
| 05:28.44 | JAHS | jajjaaa. . |
| 05:28.56 | JAHS | lol |
| 05:28.56 | SureV | haha, polyglot more like it |
| 05:29.02 | JAHS | yeah .. sorry .. |
| 05:29.22 | SureV | yeah, I love learning languages, both in computers and in people :-) |
| 05:29.29 | JAHS | cool .. |
| 05:29.44 | JAHS | heu guys, I'm really worried about this issue ... |
| 05:29.48 | SureV | though French is my worst languade, with reasonable fluency in German and |
| 05:29.52 | JAHS | could you help me .. |
| 05:30.05 | JAHS | do you know any good link or something as a guide to fix this issue? |
| 05:30.31 | SureV | hmm, looking up cosas for hesco at momento |
| 05:30.41 | SureV | but will throw some queries in there for you. |
| 05:30.58 | JAHS | thanks a lot man, I see the card in lspci, but it does not goes up .. |
| 05:31.01 | hesco | I pick up the spanish language print media around here and stumble horribly. I get some sense of the gist of an article, but lose all the details. |
| 05:31.03 | JAHS | service zaptel restart |
| 05:31.07 | JAHS | ztcfg -vvv .. |
| 05:31.20 | JAHS | i/m using cento 5.1 |
| 05:31.55 | JAHS | ZT_CONFIG .. no such device or address |
| 05:32.36 | adorah | JAHS : Do u use plain * or with Freepbx? |
| 05:32.45 | SureV | hesco, its the same in computers and speaking that once you learn a second language the others come much easier, you just have to do your best to think and mumble to yourself in the particular language all the time, so you begin to think in that language |
| 05:32.58 | SureV | books stink, immersion is the best way, |
| 05:33.02 | JAHS | plain * |
| 05:33.16 | SureV | when you have some time in the future, look up Stuart J Raj |
| 05:33.21 | adorah | JAHS> have you configured zaptel.conf? |
| 05:33.21 | hesco | yeah, I've so rarely had the opportunity to immerse myself, though. |
| 05:33.26 | JAHS | yes Sr .. |
| 05:33.34 | SureV | He's fluent in like 15 languages |
| 05:33.41 | SureV | when I say fluent I mean touch typing in |
| 05:33.43 | JAHS | span=1,0,0,ccs,hdb3 |
| 05:33.44 | SureV | all 15 |
| 05:34.02 | JAHS | bchan=1-23 |
| 05:34.06 | JAHS | dchan=24 |
| 05:34.11 | JAHS | sorry .. |
| 05:34.18 | JAHS | bchan=1-15,17-31 |
| 05:34.21 | JAHS | dchan=16 |
| 05:34.57 | hesco | I can visit a taqueria and not embarrass myself too badly when I order from the menu. But its clear I'm a gringo and operating outside my own language. |
| 05:35.33 | adorah | JAHS>Oh than u use E1 full house? you have 30 channels? |
| 05:35.39 | SureV | the key to not sounding like a gringo is that you get good with accents |
| 05:35.52 | JAHS | yes Sr .. |
| 05:35.55 | JAHS | 30 channels .. |
| 05:36.01 | SureV | and be sensetive to the nuances of the pronuciation |
| 05:36.25 | SureV | like learning to play guitar or piano, start slow and do it right at very slow speads |
| 05:36.41 | SureV | then do 1000 repetitions a day until you're getting it right at faster speeds |
| 05:36.45 | SureV | same with languages |
| 05:37.10 | SureV | repetition and practice, always the key |
| 05:37.57 | adorah | JAHS>Try to give the command genzaptelconf after you stop * than reboot |
| 05:38.20 | hesco | I travelled this past summer and fall with a few folks, two of them native french speakers, the other well practiced, if a native english speaker. I found that some of it came back. I surely missed the nuance of verb conjugation and such, and would ask regular questions about missed vocabulary. But besides the simplest of travel phrases never attempted serious conversation in French. Those folks just moved way too fast for me. Even t |
| 05:38.21 | hesco | American among them. |
| 05:38.31 | JAHS | so, you mean: |
| 05:38.35 | JAHS | 1. stop asterisk |
| 05:38.42 | JAHS | 2. genzaptelconf |
| 05:38.45 | JAHS | 3. reboot? |
| 05:38.54 | adorah | JAHS>Right |
| 05:39.04 | JAHS | ok .. just a sec .. |
| 05:39.20 | SureV | hesco: yeah I know what you mean, for me french is one of the more difficult languages to "parse" |
| 05:40.11 | hesco | and French is my best language by far, other than english. |
| 05:40.21 | SureV | just hearing someone say "Je suis Hereux de faire votre connaisance" threw me off |
| 05:40.24 | adorah | JAHS>after reboot check zaptel.conf again |
| 05:40.25 | SureV | for a long time |
| 05:40.40 | SureV | or Il fait que je mange un po,,e |
| 05:40.43 | SureV | pomme |
| 05:40.57 | SureV | too many declensions |
| 05:41.16 | SureV | it stinks haha, let the french have their language :-) I don't want it |
| 05:41.23 | hesco | any leads on the trunking issue I'm having? |
| 05:41.54 | SureV | sorry, haha, performing a balancing act :-) I have one of my underlings looking it up at the moment |
| 05:42.18 | SureV | haha, using company resources to do work for people on IRC,ahhh the power.... |
| 05:44.34 | hesco | SureV: Thanks. I appreciate the help. I spent several weeks with asterisk in the Fall, read the OReillys book, got my configuration working first priority features, then wrote a bit of perl to automate some of my frequent interactions with the asterisk server. But I can't yet say I have fully wrapped my head around how this all works. |
| 05:45.08 | hesco | I assure you your company's resources are beiing put to excellent use on this project. |
| 05:46.04 | SureV | haha, not sure my supers would see it that way LOL |
| 05:46.17 | SureV | but who cares, I put in overtime, so can they :) |
| 05:47.06 | SureV | have you tried putting that included fax-out line into the ttyIAX0 dialplan? |
| 05:47.12 | SureV | did the console output change any? |
| 05:48.11 | hesco | I did. |
| 05:48.52 | SureV | and does diamondcard support fax protocols? |
| 05:49.09 | hesco | I'm not sure. |
| 05:49.20 | *** join/#asterisk generalhan (n=Red_Drag@ip72-204-252-205.ph.ph.cox.net) |
| 05:49.23 | generalhan | hey all ! |
| 05:50.04 | adorah | Has anyone here experienced running * over Wi Max? |
| 05:50.13 | SureV | hesco, have something for you to test in a short moment |
| 05:50.39 | hesco | iax.conf is saying allow gsm for diamondcard, but iaxmodem wants alaw or ulaw |
| 05:50.42 | generalhan | i need a little direction here ... im using MixMonitor to record calls to a specified directory, but if the directory doesnt exist it wont create it ... Monitor() did this for me ... is there a way that i can get it to work properlly with MixMonitor() ?? |
| 05:50.47 | hesco | standing by |
| 05:52.02 | SureV | have you tried changing the fax-out context to: |
| 05:52.03 | SureV | [fax-out] |
| 05:52.03 | SureV | exten => _1NXXNXXXXXX,1,Dial(IAX2/diamondcard/${EXTEN}) |
| 05:52.09 | hesco | In the past I've decided that the time I invest in framing an answerable question to move a client project forward is billable, but that my time hanging out in irc waiting for the response is not so often billable. |
| 05:52.20 | SureV | yeah, understandable |
| 05:53.20 | SureV | give that a go and see if in console it show the outgoing connection |
| 05:53.22 | SureV | happening |
| 05:53.25 | hesco | so you are saying: exten => _1NXXNXXXXXX,1,Dial(IAX2/diamondcard/${EXTEN}) |
| 05:53.49 | SureV | yes, saying to change |
| 05:53.51 | SureV | exten => _1NXXNXXXXXX,1,Dial(IAX2/secret@usgw.ast.diamondcard.us/${EXTEN}) |
| 05:53.51 | SureV | to |
| 05:53.57 | SureV | exten => _1NXXNXXXXXX,1,Dial(IAX2/diamondcard/${EXTEN}) |
| 05:54.10 | hesco | and not: exten => _1NXXNXXXXXX,1,Dial(IAX2/userid@diamondcard.host/${EXTEN}) ??? |
| 05:54.11 | adorah | <hesco>perhaps not billable but a lot of fun..:P |
| 05:54.13 | SureV | to tell it to dial out the diamondcard context |
| 05:54.16 | SureV | rather than |
| 05:54.24 | SureV | a direct IAX dialout |
| 05:54.26 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.12.110.105) |
| 05:54.40 | hesco | ok, trying that now. |
| 05:54.45 | SureV | sorry, to dial out the diamondcard trunk you have defined |
| 05:54.48 | SureV | rather than directly |
| 05:54.54 | hesco | ok |
| 05:55.00 | SureV | its worth a try |
| 05:56.50 | hesco | WARNING[3775]: chan_iax2.c:7111 socket_read: Call rejected by 127.0.0.1: Busy |
| 05:57.16 | SureV | hmm, so essentially the same thing |
| 05:57.25 | hesco | it would seem. |
| 05:59.31 | SureV | ok, knock out that second dial string |
| 05:59.51 | SureV | comment it out I mean |
| 06:00.08 | SureV | how are you triggering a dialout? |
| 06:00.18 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 06:00.27 | SureV | softphone or sending a fax (i mean during this testing right now) |
| 06:00.27 | *** join/#asterisk DJ_HaMsTa (n=tocmo0nl@34.sub-97-2-47.myvzw.com) |
| 06:04.04 | generalhan | anyone have any advice on creating directories for MixMonitor ? |
| 06:04.49 | jameswf-home | ~advice |
| 06:04.50 | jbot | hmm... advice is something for which you must pay attention. Many people get irritated if they have to repeat themselves |
| 06:05.12 | drmessano | O RRY? ^_^ |
| 06:05.46 | jameswf-home | ~swallow |
| 06:05.49 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 06:06.03 | generalhan | jameswf-home: ?? |
| 06:06.18 | hesco | hylafax's faxstat -s reports: 33 126 S hesco 17707551543 0:1 1:12 23:15 No carrier detected |
| 06:06.37 | hesco | when I comment out the dial command. and reload extensions |
| 06:07.23 | hesco | I'm using, from a bash cli: sendfax -n -d 17707551543 TestFaxPage.pdf |
| 06:07.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:08.05 | hesco | hylafax uses ttyIAX0, which is managed by asterisk and seeks its dialplan from extensions.conf |
| 06:10.57 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 06:11.16 | *** join/#asterisk [gnubie] (n=[gnubie]@cm32.omega116.maxonline.com.sg) |
| 06:11.49 | [gnubie] | hello all.. |
| 06:11.59 | [gnubie] | what causes a "one-way audio"? |
| 06:12.11 | mihinomenest | NAT. |
| 06:12.23 | [gnubie] | my sip hard phone is behind my gateway/asterisk box |
| 06:12.36 | *** join/#asterisk salzh (n=salzh@124.77.120.43) |
| 06:12.56 | [gnubie] | calling to other analog and remote sip extensions, no problem |
| 06:13.02 | *** join/#asterisk tristanbob_ (n=tristanr@oalug/member/tristanbob) |
| 06:13.06 | mihinomenest | your asterisk box is behind nat. |
| 06:13.33 | [gnubie] | receiving a call from pots telephone to this sip hard phone through Zap/4 pots trunk, no problem |
| 06:13.41 | hesco | SureV: Your questions about accepted codecs led me here: http://wiki.diamondcard.us/podwiki?page=AstIax, |
| 06:13.55 | hesco | I'm going to test this and see if it answers my questions. |
| 06:14.56 | rkeene | is glad he never has to deal with NAT |
| 06:15.13 | [gnubie] | but when this sip hard phone initiates to call a telephone from the pots network through the Zap/4 pots trunk of my gateway/router/asterisk box, the (callee) telephone from the pots network cannot hear anything from the caller (sip hard phone) but the sip hard phone can hear the telephone from the pots |
| 06:15.26 | [gnubie] | mihinomenest: nope, it's not behind nat |
| 06:15.44 | [gnubie] | mihinomenest: my asterisk box is my firewall/gateway/router/pbx box |
| 06:16.26 | mihinomenest | well, alright, but that only really applies if your outbound call is via a sip line of some sort. |
| 06:16.31 | mihinomenest | it's not, so. |
| 06:16.42 | jameswf-home | nothing like a spf |
| 06:16.56 | [gnubie] | any idea? |
| 06:17.06 | mihinomenest | I'd hazard that your firewall software might be causing that. |
| 06:17.14 | [gnubie] | why am i getting this one-way audio problem calling to a pots network? |
| 06:17.43 | mihinomenest | when you call your pots line, asterisk picks up the ZAP channel, and then initiates a TCP/UDP connection out to your phone. |
| 06:17.54 | mihinomenest | so, the firewall allows it. |
| 06:18.25 | mihinomenest | but, when you're calling out from your phone, it looks like an incoming TCP/UDP connection to your firewall, so it doesn't allow it. |
| 06:18.27 | [gnubie] | mihinomenest: afair, my asterisk/firewall/gateway/router box allows udp ports 10000 to 10010 to go out and go in |
| 06:18.44 | mihinomenest | ok, but is your phone configured for that? |
| 06:18.50 | [gnubie] | yes |
| 06:19.05 | mihinomenest | alright, must be magic then. |
| 06:19.16 | generalhan | [gnubie] : take the firewall out of the equation to be sure |
| 06:19.19 | mihinomenest | stick around, someone that knows more is bound to come along. |
| 06:19.23 | generalhan | turn it off and see what happens |
| 06:19.30 | [gnubie] | i'm confused.. does pots uses tcp/udp ? |
| 06:19.41 | mihinomenest | no. |
| 06:19.53 | mihinomenest | the TCP/UDP is the sip connection between asterisk and your phone. |
| 06:19.55 | generalhan | but * does to pass voice to your connected phones |
| 06:20.19 | mihinomenest | the RTP stream for the audio is UDP, I just can't remember if asterisk does sip over UDP or TCP. |
| 06:20.41 | [gnubie] | but the problem is that when my sip hard phone sends an audio, the telephone at the pots network cannot hear it |
| 06:20.54 | SureV | hesco: another thing to check is going to be T.30, T.31 and/or T.38 support |
| 06:20.56 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 06:21.09 | [gnubie] | mihinomenest: udp starting port 10000 |
| 06:21.11 | mihinomenest | gnubie: yep. |
| 06:21.14 | hesco | does a NoOp() from an included context show up in the console? |
| 06:21.16 | SureV | except I can already tell you IAX2 being what Diamondcard is providing you with, will not |
| 06:21.21 | SureV | allow T38 passthrough |
| 06:21.25 | SureV | for faxes |
| 06:21.55 | mihinomenest | gnubie: and that's what firewalls are good for. |
| 06:21.59 | SureV | yes, NoOp should go to the console |
| 06:22.05 | hesco | and iaxmodem doesn't support it anyway. They limit me to ulaw and alaw |
| 06:22.24 | [gnubie] | again, if caller is my sip hard phone and callee is a pots telephone, the pots telephone cannot hear my sip hard phone but the sip hard phone can hear the pots telephone |
| 06:22.53 | mihinomenest | I understood what you said, I'm not sure you understood what I said. |
| 06:22.57 | SureV | hesco: also, to get NoOp commands to the console you'll need your debug level below 3 |
| 06:23.01 | SureV | i.e. 2 |
| 06:23.07 | [gnubie] | but if the other way around, meaning, the caller is the pots telephone and the callee is the sip hard phone, the call is perfect |
| 06:23.24 | hesco | I'm not seeing it. SHouldn't extensions reload be sufficient? |
| 06:23.36 | mihinomenest | there's two audio streams for every SIP call. |
| 06:23.47 | mihinomenest | one for the audio from asterisk, and one for the audio from your phone. |
| 06:24.18 | mihinomenest | when someone calls your pots line and asterisk passes that audio to your sip phone, your sip phone then has to send an audio stream back to asterisk. |
| 06:24.58 | [gnubie] | mihinomenest: yes, i understand that one |
| 06:24.59 | mihinomenest | since asterisk initiated the whole thing, your firewall percieves the incomine audio stream from your phone as a connection related to the audio stream coming from asterisk. |
| 06:25.00 | SureV | hesco: you'll need to set your verbose to lower than 3 |
| 06:25.06 | SureV | meaning set it to 2 |
| 06:25.16 | SureV | to allow you to see NoOp |
| 06:25.25 | mihinomenest | when your sip phone calls out, it initiates the connections. |
| 06:25.44 | mihinomenest | so, it tells asterisk via a sip message to send audio to it, then it starts sending audio to asterisk. |
| 06:25.53 | [gnubie] | then, what shall i do here? |
| 06:26.12 | mihinomenest | since asterisk didn't initiate that, your firewall doesn't know that the two audio streams are related and drops the one from your phone to asterisk. |
| 06:26.18 | mihinomenest | turn off your firewall. |
| 06:26.26 | mihinomenest | at least on that interface. |
| 06:26.59 | [gnubie] | also, from the webui of my sip hard phone, it tells me what port numbers to use for the register port, signal port and control port.. but i uses 5060 for all of them |
| 06:27.26 | [gnubie] | ok, i'll try turning off the firewall totally |
| 06:28.02 | mihinomenest | that stuff's all for sip messages, which is ok, because calls are being setup. |
| 06:28.09 | mihinomenest | the problem you're having is with rtp. |
| 06:29.00 | *** join/#asterisk tristanbob (n=tristanr@oalug/member/tristanbob) |
| 06:29.08 | *** join/#asterisk SureV (n=surev@205.144.218.234) |
| 06:29.23 | SureV | hesco you still there? |
| 06:29.36 | [gnubie] | mihinomenest: the problem is still there.. |
| 06:29.47 | SureV | just had compiz-fusion crash on my and it took Konversation with it |
| 06:30.03 | [gnubie] | mihinomenest: yeah, rtp |
| 06:30.04 | DJ_HaMsTa | u were asking for it |
| 06:30.19 | mihinomenest | gnubie: alright, well, turn your firewall back on then. |
| 06:30.25 | mihinomenest | but, I'm out of ideas. |
| 06:30.58 | [gnubie] | mihinomenest: i already turned off totally the firewall but the problem is still there.. one way audio. :( |
| 06:31.43 | mihinomenest | I suppose you could run a sip debug on the console and make sure that your phone is saying the right things, but otherwise, I'd suggest you wait until someone that knows more than I is around. |
| 06:32.16 | [gnubie] | ok |
| 06:32.26 | [gnubie] | mihinomenest: thanks for trying to help me. |
| 06:36.39 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 06:36.52 | hesco | SureV: http://pastebin.com/d36dd5dd3 |
| 06:37.31 | hesco | pls note changes to extensions.conf, iax.conf and to the errors I'm now seeing in the asterisk console. |
| 06:38.02 | hesco | yeah, I'm still here. |
| 06:38.24 | [gnubie] | mihinomenest: this is the cli output when calling from my sip hard phone to a pots telephone through the Zap/4 pots trunk => http://www.privatepaste.com/db1V7czguD |
| 06:38.25 | hesco | was trying some things, then documenting what I saw in new pastebin post |
| 06:41.48 | SureV | yeah, change the codec on the dimaond conf to alaw |
| 06:41.53 | SureV | as that is what they are wanting |
| 06:42.00 | SureV | or preferring rather |
| 06:43.20 | SureV | hesco, in IAX2.conf |
| 06:43.35 | SureV | add a line to the diamondcard trunk |
| 06:43.39 | SureV | allow=alaw |
| 06:43.51 | SureV | so diamond gets the codec they want |
| 06:44.06 | SureV | they are disconnecting you because they aren't getting alaw |
| 06:50.22 | [gnubie] | ok |
| 06:50.22 | hesco | I have allow-alaw in that file. |
| 06:50.51 | hesco | I'm still getting: Operating with different codecs 4[0x4 (ulaw)] 2[0x2 (gsm)] , can't native bridge... |
| 06:50.56 | [gnubie] | i got a tcpdump capture of the scenario: sip hard phone calls the pots telephone through a Zap/4 pots trunk |
| 06:51.20 | hesco | I suspect that diamondcard wants gsm, only. |
| 06:51.36 | hesco | I found no notes about any other supported codec. |
| 06:52.28 | SureV | change it from allow alaw to allow=alaw |
| 06:53.23 | SureV | and have the allow=alaw the first one after disallow=all |
| 06:59.01 | [gnubie] | http://www.megaupload.com/?d=FMX1MJSK |
| 06:59.40 | SureV | yes, but the log you posted is indicating that they cannot find a compatible protocol, and above that its asking for alaw |
| 06:59.47 | SureV | or rather, they are preferring alaw |
| 06:59.58 | hesco | disallow=all |
| 06:59.58 | hesco | allow=alaw |
| 07:00.16 | SureV | its in there like that? and still no go? |
| 07:00.28 | hesco | correct |
| 07:00.41 | SureV | what does the console say? |
| 07:01.15 | hesco | Call accepted by 64.124.128.5 (format gsm) |
| 07:01.30 | hesco | Operating with different codecs 4[0x4 (ulaw)] 2[0x2 (gsm)] , can't native bridge... |
| 07:01.48 | SureV | gotcha, ok, so native bridging issue |
| 07:02.04 | hesco | and my reading says fax won't work well with gsm, that I should use alaw and ulaw, instead |
| 07:02.45 | hesco | does that suggest to you how to address it? |
| 07:03.16 | hesco | is there some other iax provider which will support my fax needs? |
| 07:03.30 | SureV | well, certainly gsm is far less fidelity than alaw or ulaw |
| 07:03.43 | SureV | so can imagine the faxing wouldn't work so well |
| 07:03.51 | hesco | apparently that is an issue. |
| 07:04.09 | SureV | hmm, I use voipstreet |
| 07:05.01 | thudd | i was led to believe that fax/modem signals didnt work under any voip encoding. . . |
| 07:05.11 | SureV | no, they are VERY problematic |
| 07:05.24 | SureV | but is doable with passthroughs |
| 07:06.21 | SureV | or good codec |
| 07:07.32 | SureV | but usually with more than a few pages, things start falling apart quickly |
| 07:07.46 | SureV | I usually just implent the send2fax API |
| 07:07.49 | SureV | as a fax gateway |
| 07:07.52 | thudd | seems sort of tragic. fax data should be a breeze to move around using ip. shame there are all the legacy voice hoops to jump through |
| 07:07.53 | SureV | or efax |
| 07:08.08 | SureV | except you have that crusial step |
| 07:08.20 | SureV | crucial step of converting the analog fax tones to digital |
| 07:08.20 | SureV | data |
| 07:08.59 | SureV | which when doing codec compression of the sounds, it becomes almost an art rather than a science |
| 07:09.37 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 07:10.07 | SureV | for the most part, send2fax or efax can totally replace any and all faxing needs |
| 07:10.11 | thudd | yeah, i played with it for a while, before just telling everyone to scan to a pdf and mail it :P |
| 07:10.15 | SureV | sure, there is a cost, but |
| 07:10.23 | SureV | reliaiblity is great |
| 07:10.26 | SureV | haha, yeah, |
| 07:10.28 | SureV | funny |
| 07:10.49 | SureV | attachemnts seems to be just as easy these days, unless one is fax spamming |
| 07:10.57 | SureV | using a spoofed caller ID |
| 07:11.43 | SureV | then having a DID provider that allows you to send up the caller ID you want to send to the other side, |
| 07:11.56 | SureV | well then you're good to go for spoofing your CID |
| 07:12.58 | thudd | i was told i needed to buy fax gateway hardware, or contract with a did provider who would do fax translation for us, and that doing it over voip was just a long headache. |
| 07:13.02 | thudd | which made me sad |
| 07:13.16 | hesco | voipstreet says supported codedcs include only: g711, GSM and g729. |
| 07:13.47 | SureV | yeah, alaw, ulaw, and gsm |
| 07:14.07 | SureV | this is a good page on faxing |
| 07:14.08 | SureV | http://www.voip-info.org/wiki/view/T.38 |
| 07:14.39 | hesco | are you saying that g711 and g729 are alaw and ulaw? |
| 07:14.41 | SureV | Just the IAX2 protocol does NOT allow t38 so you would need a SIP provider that allows passthrough |
| 07:14.44 | SureV | yes |
| 07:14.49 | SureV | that is what they are |
| 07:15.00 | thudd | ah ha |
| 07:15.03 | hesco | did not realize that. |
| 07:15.05 | SureV | just ulaw is a pay for codec you have to buy |
| 07:15.32 | hesco | that would be g729, which I had read was proprietary and licensable |
| 07:15.41 | SureV | yes |
| 07:16.04 | SureV | talk to Digium for that (I think), or know where to look on the net |
| 07:16.33 | thudd | i thought ulaw was g711 and was free? |
| 07:16.59 | SureV | *ahem* google G729 rapidshare *ahem* |
| 07:18.24 | SureV | G.711 is free |
| 07:19.05 | SureV | yeah, I had those backwardss |
| 07:19.37 | thudd | has to google it every time :P |
| 07:20.03 | SureV | haha, yeah, sometimes here too, or I just rant for twenty minutes using it backwards |
| 07:20.17 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 07:20.40 | SureV | but either case, faxing IS doable, but not exactly the most precise science |
| 07:21.01 | SureV | hesco, what is the outgoing fax being used for? |
| 07:21.51 | generalhan | BAH !!! ive spent the last 2 hours looking for a solution to MixMonitor creating non existing directories ... and come up with nothing. |
| 07:22.12 | SureV | I only ask to get a feel for what alternate options are available |
| 07:29.03 | rkeene | I'm going to be setting up outbound faxing in the next couple of months. |
| 07:39.38 | *** join/#asterisk Rico29 (n=Rico@ARennes-358-1-60-3.w90-49.abo.wanadoo.fr) |
| 07:52.22 | *** join/#asterisk RoyK (n=roy@ip-234-57-149-91.dialup.ice.no) |
| 07:52.45 | *** join/#asterisk nighty^ (n=nighty@p5187-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 08:19.07 | hesco | SureV: this will be used for broadcasting faxed press releases to the media. |
| 08:19.29 | hesco | I set up an account at: voipstreet. Waiting for confirmation, now. |
| 08:19.55 | hesco | I never found what I thought I was looking for at rapidshare. |
| 08:22.18 | hesco | or maybe I did. I'm not sure. |
| 08:23.04 | hesco | but thanks for your help, I think I gained some clarity that my next step is to handle the trunking (and match codecs) so I can reach the outside world. |
| 08:23.58 | hesco | I need to get some sleep here. Thanks again. |
| 08:36.39 | *** join/#asterisk tristanbob (n=tristanr@oalug/member/tristanbob) |
| 08:38.15 | *** part/#asterisk [gnubie] (n=[gnubie]@cm32.omega116.maxonline.com.sg) |
| 08:52.17 | *** join/#asterisk tristanbob_ (n=tristanr@oalug/member/tristanbob) |
| 08:54.10 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 08:59.33 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 09:12.12 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 09:50.37 | *** join/#asterisk BeeBuu (n=beebuu@61.142.247.242) |
| 09:51.03 | *** join/#asterisk ParanoyaM (n=Maksim@92.112.32.22) |
| 09:51.10 | ParanoyaM | Hello |
| 09:51.27 | ParanoyaM | does anybody use asterisk with openvpn clients? |
| 09:51.30 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 09:51.34 | ParanoyaM | have a problem with one side hearing |
| 09:53.38 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 09:53.52 | ParanoyaM | hey |
| 09:54.03 | ParanoyaM | anybody is present? |
| 09:54.46 | ParanoyaM | guys!!!!! |
| 09:54.57 | BeeBuu | hello |
| 09:55.23 | ParanoyaM | hello |
| 09:55.43 | ParanoyaM | do my question is incorect? |
| 09:56.02 | BeeBuu | would you type it again? |
| 09:56.28 | ParanoyaM | BeeBuu, i have a oneside hearing with my oenvpn clients |
| 09:56.46 | BeeBuu | so? what's the problem? |
| 09:57.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:57.06 | ParanoyaM | :) need to be two be two side hearing |
| 09:57.51 | ParanoyaM | i mean when i make call to vpn client he can't hear me |
| 09:57.53 | ParanoyaM | but i can hear him |
| 09:58.04 | ParanoyaM | asterisk and opencpn server are same |
| 09:58.08 | ParanoyaM | openvpn |
| 09:58.11 | BeeBuu | how about without vpn client? |
| 09:58.23 | ParanoyaM | all ok |
| 09:58.31 | ParanoyaM | problem only with vpn clients |
| 09:58.53 | ParanoyaM | i even tried to hear the ivr |
| 09:58.56 | ParanoyaM | as vpn client |
| 09:59.02 | ParanoyaM | i couldn't |
| 10:00.14 | BeeBuu | both caller and callee with vpn client? |
| 10:00.22 | ParanoyaM | no |
| 10:00.27 | ParanoyaM | caller is real ip |
| 10:02.01 | BeeBuu | sip? |
| 10:02.08 | ParanoyaM | sure |
| 10:02.38 | BeeBuu | caller can't get anything? or callee? |
| 10:03.00 | ParanoyaM | calle can't hear caller |
| 10:05.16 | BeeBuu | how about all client with vpn? |
| 10:06.09 | ParanoyaM | i don't try, because i don't need so |
| 10:07.54 | *** join/#asterisk CVirus (n=GoD@82.201.178.6) |
| 10:08.04 | BeeBuu | ok,i got you,i record your problem. |
| 10:08.54 | ParanoyaM | and? what does it mean record? |
| 10:10.09 | BeeBuu | i'm a problem BOT :-D |
| 10:11.56 | BeeBuu | collecte problems |
| 10:12.17 | ParanoyaM | :) |
| 10:12.21 | ParanoyaM | and what next? |
| 10:12.26 | ParanoyaM | will try to emulate them? |
| 10:12.52 | ParanoyaM | or just collector? :) |
| 10:14.20 | BeeBuu | answer your question,again |
| 10:14.29 | BeeBuu | again,and again~~~~ |
| 10:14.39 | BeeBuu | till someone fix it. |
| 10:15.48 | BeeBuu | ask your question... |
| 10:16.29 | ParanoyaM | <PROTECTED> |
| 10:16.38 | ParanoyaM | Not very good idea :) |
| 10:17.28 | ParanoyaM | will inform me? when someone fix it? netvin@gmail.com? :) |
| 10:18.10 | BeeBuu | maybe,if you can come back is better. |
| 10:18.45 | ParanoyaM | how much time? |
| 10:20.40 | ParanoyaM | day month week |
| 10:20.50 | BeeBuu | who know? |
| 10:21.03 | BeeBuu | but not now. :-P |
| 10:21.39 | BeeBuu | now is midnight~~~ |
| 10:22.14 | ParanoyaM | midnight??? where do you live? |
| 10:24.13 | BeeBuu | midnight is for most of people in this room |
| 10:25.23 | BeeBuu | see you around. |
| 10:25.43 | BeeBuu | welcome to come back. |
| 10:25.52 | *** part/#asterisk BeeBuu (n=beebuu@61.142.247.242) |
| 10:34.54 | *** join/#asterisk adrift (n=anon@ppp-70-128-40-74.dsl.ltrkar.swbell.net) |
| 10:39.11 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 10:50.45 | ParanoyaM | thanks |
| 10:56.30 | *** join/#asterisk SteveTotaro (n=root@pool-71-166-111-39.bltmmd.east.verizon.net) |
| 11:19.38 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 11:35.20 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 12:04.54 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 12:36.06 | *** join/#asterisk PepOSX (n=angeldav@190.72.148.226) |
| 12:40.26 | *** join/#asterisk masus (n=ethemc@88.248.14.186) |
| 12:40.48 | masus | hi all , dows anyone know this warning -> http://pastebin.com/m4f66756d .. Thanks |
| 12:51.43 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 12:54.22 | mvanbaak | mornin oej |
| 12:54.35 | oej | Hello there |
| 12:54.42 | mvanbaak | happy easter |
| 12:57.26 | oej | The same to you and Nancy |
| 12:57.29 | oej | Back later |
| 13:12.57 | *** join/#asterisk Tili (n=tili@153.Red-80-38-134.staticIP.rima-tde.net) |
| 13:25.41 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:32.49 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 13:43.00 | ManxPower | The answers you seek are within you, grasshopper. |
| 13:43.16 | *** join/#asterisk Tili (n=tili@153.Red-80-38-134.staticIP.rima-tde.net) |
| 13:43.57 | Tili | is it possible in asterisk to dynamically select codecs based on number of calls active so that if bandwidth is low, it selects speex |
| 13:44.09 | ManxPower | Tili: No. |
| 13:44.18 | Tili | ManxPower: thanks. thought so but worth a shot |
| 13:44.19 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:49.51 | *** join/#asterisk SteveTotaro (n=root@pool-71-166-104-72.bltmmd.east.verizon.net) |
| 14:25.41 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-230-164-18.dsl.irvnca.pacbell.net) |
| 14:32.49 | *** part/#asterisk masus (n=ethemc@88.248.14.186) |
| 14:36.11 | *** join/#asterisk amaache (n=maache76@41.221.27.89) |
| 14:38.20 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 14:45.31 | *** join/#asterisk CaRb0n^ (n=playa@203.81.221.240) |
| 14:45.34 | CaRb0n^ | pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line 807 |
| 14:45.38 | CaRb0n^ | any one any idea of this error |
| 14:46.25 | Qwell | ...it says exactly what it wants, and even says why |
| 14:46.57 | CaRb0n^ | i know but i cant find where do i need to make changes |
| 14:47.03 | CaRb0n^ | cant find the exact location |
| 14:47.04 | CaRb0n^ | and file |
| 14:47.07 | Qwell | line 807 |
| 14:47.36 | CaRb0n^ | but i cant find file name pbx_config.c |
| 14:47.58 | CaRb0n^ | its the only error in my box |
| 14:48.03 | CaRb0n^ | the rest is all fine n smooth |
| 14:48.05 | ManxPower | CaRb0n^: try extensions.conf |
| 14:48.10 | CaRb0n^ | thanks m8 |
| 14:48.13 | CaRb0n^ | let me check it out |
| 14:48.32 | CaRb0n^ | just trying to make my box with 0 warning |
| 14:49.06 | ManxPower | sometimes you need exten => _. but most of the time it's just a BAD IDEA |
| 14:49.33 | Greek-Boy | when doing an upgrade should I delete the old source directory and just compile and build from new source? |
| 14:49.42 | Greek-Boy | or should I do make uninstall from old source first? |
| 14:49.55 | ManxPower | Greek-Boy: I wasn't aware there was a make uninstall |
| 14:49.56 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 14:49.58 | CaRb0n^ | thanks ManxPower? |
| 14:50.13 | CaRb0n^ | any one here got good exprience with Chanspy? |
| 14:50.16 | Greek-Boy | ManxPower: I read it somewhere. |
| 14:50.41 | ManxPower | hands Greek-Boy the "Most Vague" award |
| 14:50.53 | Greek-Boy | http://mindboosternoori.blogspot.com/2006/04/howto-uninstall-asterisk.html |
| 14:50.59 | Greek-Boy | maybe it only applies to 1.2 |
| 14:51.08 | ManxPower | Greek-Boy: why don't you just try it. |
| 14:52.24 | Greek-Boy | ok. did it. |
| 14:53.45 | Greek-Boy | worked nicely |
| 14:53.48 | Greek-Boy | make uninstall |
| 14:53.49 | hesco | where would I find a g.729 codec I can install with asterisk on a Debian box? |
| 14:53.52 | Greek-Boy | and make uninstall-all |
| 14:54.00 | ManxPower | hesco: from the Digium store. |
| 14:54.07 | hesco | thanks. |
| 14:54.24 | ManxPower | Licenses are US$10/per simul channel |
| 14:59.36 | CaRb0n^ | hmm BIT CONFUSED |
| 14:59.40 | CaRb0n^ | confusing |
| 14:59.49 | CaRb0n^ | anyways no one here had any exprience with chanspy? |
| 15:00.11 | file | if you ask a specific question someone may answer |
| 15:00.12 | ManxPower | CaRb0n^: It's Easter Sunday. |
| 15:00.13 | Greek-Boy | CaRb0n^ i've had a bit |
| 15:00.42 | Kernel_Core | ManxPower: so happy easter :D |
| 15:01.06 | CaRb0n^ | well i have been trying to use chanspy to monitor live calls |
| 15:01.12 | CaRb0n^ | the only issue is |
| 15:01.19 | CaRb0n^ | the sound quallity is really bad |
| 15:01.23 | ManxPower | Kernel_Core: Never could understand the worship of bunnies. They are cute and all, I guess. |
| 15:01.29 | CaRb0n^ | i am trying it since a@h 1.5 |
| 15:01.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:02.09 | Kernel_Core | ManxPower: I amnot christian , so I have no Idea ;) |
| 15:02.16 | ManxPower | CaRb0n^: Unfortunately, this is @asterisk, not @asteriskathome. |
| 15:02.26 | ManxPower | Kernel_Core: I'm a devout atheist. |
| 15:02.46 | CaRb0n^ | but chanspy is asterisk feature |
| 15:02.51 | CaRb0n^ | not a@h i guess |
| 15:03.30 | ManxPower | CaRb0n^: Yes, but the incredibly complex config files Asterisk@Home uses makes it impossible to support by anyone other than someone familiar with Asterisk@Home. |
| 15:03.33 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:03.49 | CaRb0n^ | i can use normal plain asterisk |
| 15:03.49 | puzzled | hi |
| 15:03.52 | CaRb0n^ | if it works ] |
| 15:03.53 | CaRb0n^ | then |
| 15:04.10 | coppice | ManxPower: you don't appeciate Playboy? |
| 15:04.16 | CaRb0n^ | cos up till now i have seen non who is runing chanspy perfectly |
| 15:04.24 | CaRb0n^ | i havent |
| 15:04.26 | CaRb0n^ | i mean |
| 15:04.31 | ManxPower | CaRb0n^: I doubt using Original Asterisk will help you, but it will allow Asterisk users to help you. |
| 15:05.01 | CaRb0n^ | well i even found any user till yet for 3 yrs now |
| 15:05.17 | CaRb0n^ | grr |
| 15:05.24 | puzzled | maybe they are just not telling you |
| 15:05.34 | ManxPower | CaRb0n^: try back Monday 10am-8pm GMT -6 |
| 15:05.38 | CaRb0n^ | i mean i havent found any one yet who can tell me anything usefull for chanspy |
| 15:05.38 | puzzled | iirc there were fixes for chan_spy in 1.4.19-rc |
| 15:05.53 | CaRb0n^ | im using 1.4.17 |
| 15:06.05 | ManxPower | Which I believe is USA Central Daylight Time |
| 15:06.09 | CaRb0n^ | ok |
| 15:06.26 | ManxPower | Also check the mailing list archives |
| 15:06.29 | ManxPower | ~mailinglist |
| 15:06.29 | jbot | [~mailinglist] Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives |
| 15:06.44 | CaRb0n^ | i can some times able to hear the extension for few seconds then no sound . and i cant hear the called number voice |
| 15:07.04 | ManxPower | the ARCHIVES can be searched by assing "site:lists.digium.com" to your Google search. |
| 15:07.17 | CaRb0n^ | i think i need to get something made which just simple stream wav files with 3 secs different |
| 15:07.20 | ManxPower | that would be aDDing.... |
| 15:07.22 | CaRb0n^ | thats will be much better |
| 15:07.23 | CaRb0n^ | i guess |
| 15:08.47 | CaRb0n^ | anyways Manxpower thanks for the help m |
| 15:18.16 | riddlebox | I am such an idiot, I installed a system the other day and forgot to set echocancel, now I have to go back monday |
| 15:20.41 | CaRb0n^ | echo cancel are only for zap channels? |
| 15:20.46 | *** join/#asterisk johann8384 (n=johann83@71.85.243.141) |
| 15:20.54 | CaRb0n^ | or it can be applied on sip trunks too? |
| 15:21.29 | coppice | you cannot echo cancel SIP trunks |
| 15:25.23 | CaRb0n^ | ok |
| 15:25.48 | CaRb0n^ | cos when ever i listen to the record calls , i can hear too much background noise people talking |
| 15:25.55 | CaRb0n^ | any suggestion for that ? |
| 15:33.26 | coppice | well, it is simply not possible to echo cancel any VoIP channel. if the echo is really on that link, you are stuck with it |
| 15:34.42 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 15:36.31 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.117.64) |
| 15:45.46 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 15:58.53 | Greek-Boy | which library is required to run gtk console? |
| 16:01.20 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-230-164-18.dsl.irvnca.pacbell.net) |
| 16:01.37 | Kernel_Core | coppice: it is possible to cancel echo , with speex |
| 16:13.19 | coppice | Kernel_Core: is this by enabling the magic potion option? |
| 16:14.31 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:15.16 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:15.16 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 16:16.22 | *** join/#asterisk Mmurdock (n=vnjyjta@229.sub-72-122-141.myvzw.com) |
| 16:16.49 | *** join/#asterisk Scruffy- (n=Scruffy@d77-218-133-100.cust.tele2.fr) |
| 16:17.00 | Scruffy- | hi |
| 16:18.36 | Mmurdock | Hi |
| 16:20.25 | Scruffy- | i need help about my asterisk :D |
| 16:20.43 | Scruffy- | if I try to connect with my softphone to my asterisk : |
| 16:21.11 | Scruffy- | [Mar 23 17:27:32] NOTICE[22090]: chan_sip.c:15075 handle_request_register: Registration from '"1001"<sip:1001@XX.XX.XX.XX5060>' failed for 'XX.XX.XX.XX' - No matching peer found |
| 16:22.06 | Mmurdock | You added an entry in your sip.conf for that device? |
| 16:22.36 | Scruffy- | yes normally... |
| 16:24.14 | rkeene | There's likely an error in your "sip.conf" then |
| 16:24.36 | Scruffy- | I can past my sip.conf if you want... ? |
| 16:24.40 | rkeene | Sure |
| 16:25.48 | [TK]D-Fender | Scruffy-: PASTEBIN <----- |
| 16:25.50 | [TK]D-Fender | ~pb |
| 16:25.51 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:25.52 | [TK]D-Fender | ^^^ |
| 16:27.48 | Scruffy- | http://pastebin.ca/954033 |
| 16:27.50 | Scruffy- | thx |
| 16:29.24 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 16:29.43 | [TK]D-Fender | Scruffy-: You don't have a [1001] in there |
| 16:30.08 | [TK]D-Fender | Scruffy-: I think you meant to set your client up to login as "scruffy" <---- |
| 16:30.20 | Scruffy- | I must replace : [scruffy] to [1001] for exemple ? |
| 16:30.36 | [TK]D-Fender | Scruffy-: You can do whichever way you want so long as they match |
| 16:30.44 | ice_croft | hi all |
| 16:31.00 | ice_croft | is there any soft to bill analog mini-PBX? |
| 16:31.05 | [TK]D-Fender | Scruffy-: why would you make a use in *'s sip.conf as a name and then think you could login into it as a number? |
| 16:31.52 | Scruffy- | sorry i don't understand, i'm french :D |
| 16:32.48 | [TK]D-Fender | Scruffy-: Pourquoi tu penses que to tu peut logger sur "1001' quand il-y-a AUCUN compte a ce nom? |
| 16:33.32 | [TK]D-Fender | Scruffy-: Tu les-a donne seulement des nome APLHA. Alor change un, ou l'autre, mais its devrait-etre PAREIL |
| 16:34.24 | Scruffy- | ok |
| 16:34.26 | Scruffy- | effectivement oui |
| 16:34.53 | [TK]D-Fender | Scruffy-: Des que t'as probablement ustiliser SIP/scruffy dans ton extensions.conf c'est mieux de changer le login sur ton client |
| 16:34.55 | Scruffy- | donc que dois-je remplacer alors ? |
| 16:35.17 | [TK]D-Fender | Scruffy-: le "1001" pour "scruffy" dans tons client. |
| 16:35.18 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:35.49 | [TK]D-Fender | Scruffy-: t'as jamais creer un compte au nom du [1001] |
| 16:36.08 | Scruffy- | donc je change [scruffy] par [1001] alors |
| 16:36.39 | [TK]D-Fender | Scruffy-: melanger JAMAIS the nom d'une compte dans une fichier comme sip.conf ace un EXTENSION ( une numero que tu peut composes) |
| 16:37.12 | [TK]D-Fender | Scruffy-: Ca depend si ton extensions.conf a faites reference au "SIP/scruffy" ou non. |
| 16:37.39 | Scruffy- | attend je te paste bin mon extensions.conf merci |
| 16:38.09 | Scruffy- | ici : http://pastebin.ca/954048 |
| 16:38.14 | Scruffy- | c'est mon extensions.conf ça |
| 16:38.34 | Zeeek | je ne savais pas que tu causais le français TK! |
| 16:39.05 | [TK]D-Fender | Scruffy-: alors laisser-fair ton si.conf & extensions.conf et change ton CLIENT pout metter let NOM and pas the NUMERO |
| 16:39.19 | Scruffy- | [TK]D-Fender tu peux me montrer un exemple stp ? |
| 16:39.31 | Scruffy- | Zeeek tu parles français aussi ? |
| 16:39.35 | Zeeek | colle un pin à [TK]D-Fender |
| 16:40.03 | Zeeek | Scruffy- oui, va sur asterisk-france.net tu en verras d'autres :) |
| 16:40.32 | Scruffy- | je suis sur le channel d'irc de asterisk-france :) |
| 16:40.36 | Scruffy- | mais c'est un peu mort là bas |
| 16:40.53 | [TK]D-Fender | c |
| 16:41.29 | [TK]D-Fender | c'est le fin-de-semain, ainsi que Paques.... il ne reste pas personne ici... |
| 16:41.46 | Scruffy- | lol |
| 16:41.48 | Zeeek | ooo là |
| 16:42.40 | Scruffy- | merci [TK]D-Fender pour tes explications |
| 16:42.51 | Scruffy- | même si je n'ai compris qu'à motier ce que tu voulais me dire ;) |
| 16:42.57 | Scruffy- | *moitier |
| 16:43.01 | [TK]D-Fender | Scruffy-: Il marche? |
| 16:43.05 | Scruffy- | no |
| 16:43.13 | [TK]D-Fender | Scruffy-: comment ca? |
| 16:43.34 | Scruffy- | pourrais tu me corriger mon pastebin s'il te plait ? |
| 16:43.45 | [TK]D-Fender | Scruffy-: T'as changer the compte sur ton telephone pis il te donne encore des erreurs? |
| 16:44.03 | [TK]D-Fender | Scruffy-: * etait CORRECT dans premiere place! |
| 16:44.18 | wastrel | is there a html version of the oreilly pdf book on the asterisk site? |
| 16:44.33 | Qwell | ~book |
| 16:44.34 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 16:44.40 | [TK]D-Fender | Scruffy-: J'avais dit de corriger ton CLIENT <- |
| 16:44.56 | Scruffy- | mais je devais corriger quoi sur mon client ? |
| 16:45.06 | Qwell | [TK]D-Fender: tsk, tsk, tsk |
| 16:45.27 | [TK]D-Fender | Scruffy-: Change le maudit 1001 pour le nom a qui qu'il devait correspondre! |
| 16:45.35 | wastrel | merci |
| 16:45.40 | Scruffy- | et Caller ID name, je met quoi ? |
| 16:45.57 | Scruffy- | Caller ID name = 1000 ? |
| 16:46.00 | [TK]D-Fender | Qwell: as-tu du quoi a dir la-tu? |
| 16:46.10 | [TK]D-Fender | Scruffy-: Pas d'importance. |
| 16:46.28 | Scruffy- | ok |
| 16:46.30 | [TK]D-Fender | Scruffy-: C'est le nom de usager qui na pas etait correct |
| 16:46.45 | [TK]D-Fender | Qwell: And ummm... y0, sup? ;) |
| 16:47.06 | Scruffy- | mais ça ne fonctionne toujours pas |
| 16:47.39 | [TK]D-Fender | Scruffy-: refaire ton pastebin avec les 2 fichier et l'error qui * tu donne. |
| 16:47.43 | [TK]D-Fender | te* |
| 16:50.09 | Scruffy- | ok |
| 16:54.09 | Scruffy- | [TK]D-Fender |
| 16:54.09 | Scruffy- | http://pastebin.ca/954060 |
| 16:54.24 | *** part/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 16:54.30 | [TK]D-Fender | [Mar 23 18:00:09] NOTICE[22440]: chan_sip.c:15075 handle_request_register: Registration from '"1000"<sip:1000@scrufffy.eu>;transport=UDP' failed for '77.218.133.100' - No matching peer found |
| 16:54.46 | [TK]D-Fender | Scruffy-: T'as pas corriger le nom du usager sur ton client! |
| 16:54.57 | Scruffy- | regarde le pastebin à la fin stp !!!! |
| 16:55.05 | [TK]D-Fender | Scruffy-: Pourquoi qu'il est encore 1000!? |
| 16:55.06 | Scruffy- | j'ai corriger le nom sur mon client |
| 16:55.16 | Scruffy- | je sais pas |
| 16:55.17 | Scruffy- | 17:54:48 Timeout registration for 'scruffy@scrufffy.eu' |
| 16:55.21 | Scruffy- | on my softphone |
| 16:55.38 | [TK]D-Fender | Scruffy-: Mal configurer... |
| 16:55.47 | [TK]D-Fender | Scruffy-: I'l te dit 2 affaires.... |
| 16:56.10 | [TK]D-Fender | Scruffy-: C'est quel soft-phone? |
| 16:56.30 | Scruffy- | Zoiper |
| 16:56.41 | [TK]D-Fender | Scruffy-: Enleve tout reference au "1000" de-la |
| 16:56.48 | Scruffy- | oui c'est fait |
| 16:57.00 | [TK]D-Fender | Scruffy-: t'as faites quele chsoe de mal la-dans |
| 16:57.50 | Scruffy- | non :/ |
| 16:59.29 | [TK]D-Fender | Scruffy-: Il-y-a un raison pourquoi * a recu "1001" comme le nom de usager... |
| 17:00.09 | [TK]D-Fender | Scruffy-: Je utilise zoiper moi-mem... |
| 17:00.11 | Scruffy- | [Mar 23 18:06:37] NOTICE[22487]: chan_sip.c:15075 handle_request_register: Registration from '"1000"<sip:1000@scrufffy.eu>;transport=UDP' failed for '77.218.133.100' - No matching peer found |
| 17:00.15 | Scruffy- | je ne sais pas pourquoi |
| 17:00.26 | Scruffy- | je vais essayer un autre softphone sur un autre ordinateur alors ! |
| 17:02.23 | [TK]D-Fender | Scruffy-: faire un image de ton configurations. |
| 17:02.42 | Scruffy- | ok |
| 17:03.44 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 17:04.30 | Scruffy- | ça veut dire quoi quand il marque "Registering" ? |
| 17:05.46 | Zeeek | [TK]D-Fender you're way more patient in French... this guy needs to RTFB |
| 17:06.14 | Scruffy- | Zeeek je comprend l'anglais quand même |
| 17:06.31 | Scruffy- | et ton RTFB tu te le met là où je le pense |
| 17:06.32 | Scruffy- | merci |
| 17:06.33 | Zeeek | good, you need to do some reading, it's all there |
| 17:07.06 | Zeeek | You should tell people where to put things, it isn't considered nice |
| 17:07.08 | Scruffy- | Zeeek shut up :) |
| 17:07.15 | Zeeek | why? |
| 17:07.23 | Scruffy- | osef |
| 17:07.24 | Zeeek | you'll be hearing the same speech all week |
| 17:07.25 | [TK]D-Fender | cool it people... |
| 17:07.55 | Zeeek | [TK]D-Fender youi can't be the same TK on during the week |
| 17:08.05 | Scruffy- | Zeeek les gens comme toi je m'en passerais largement |
| 17:08.05 | Zeeek | I don't recognize you at all :) |
| 17:08.19 | [TK]D-Fender | Zeeek: Its a lazy sunday and I'm not fried yet. Don't push me :) |
| 17:08.26 | Scruffy- | [TK]D-Fender http://img522.imageshack.us/img522/1406/zoiperzx6.jpg |
| 17:08.30 | Scruffy- | thanks a lot |
| 17:08.34 | Zeeek | fortunately there's ignore |
| 17:08.58 | Zeeek | someone else can defend my honor if need be |
| 17:09.03 | [TK]D-Fender | Scruffy-: Get rid of TEST |
| 17:13.59 | Scruffy- | [TK]D-Fender |
| 17:14.09 | Scruffy- | do you know a good softphone for linux (fedora) ? |
| 17:14.25 | [TK]D-Fender | Scruffy-: Ekiga |
| 17:16.05 | Scruffy- | thx |
| 17:17.11 | *** join/#asterisk seaq (n=seaq@201.244.27.106) |
| 17:18.33 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.140) |
| 17:19.03 | rkeene | OpenWengo, too. |
| 17:19.56 | [TK]D-Fender | bleh |
| 17:26.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:34.42 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-151-117.satx.res.rr.com) |
| 17:39.18 | UnixDog | ok 1.4.19 rc3 works on bsd now |
| 17:39.20 | *** join/#asterisk wundaboy (n=pat@c-24-21-228-126.hsd1.mn.comcast.net) |
| 17:39.27 | UnixDog | I got the prot working |
| 17:39.50 | UnixDog | port even |
| 17:39.52 | UnixDog | lol |
| 17:40.07 | wundaboy | does asterisk run over tcp or udp (configuring qos) |
| 17:46.00 | [TK]D-Fender | wundaboy: UDP |
| 17:46.25 | wundaboy | thanks |
| 17:46.45 | wundaboy | thats what i thought, i just wanted to verify |
| 17:46.45 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 17:52.25 | *** join/#asterisk xanderp (n=pphillip@98.220.163.8) |
| 17:52.51 | xanderp | am I just not looking in the right places, or are all of the VOIP provider reviews like 3 years old? |
| 17:55.31 | *** join/#asterisk seanbright-home (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net) |
| 17:56.56 | [TK]D-Fender | xanderp: Must be looking in the the wrong places. |
| 17:59.54 | xanderp | seems like they are all reviews and articles written in like 2003-2005... |
| 18:00.31 | xanderp | don't want to make decisions based on dated information... |
| 18:03.15 | eric2 | xanderp - make damn sure you do your homework!! |
| 18:03.29 | eric2 | the providers will tell you anything to make the sale |
| 18:03.37 | eric2 | *some will |
| 18:04.31 | joat | anyone got pointers to docs describing passing a numeric variable back to a dial plan from an AGI script? |
| 18:04.42 | xanderp | eric2: who are you using for termination, and do you like them? |
| 18:16.24 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 18:19.30 | [TK]D-Fender | joat: "SET VARIABLE" |
| 18:22.59 | joat | D-Fender: here's what I'm trying to do |
| 18:23.02 | joat | exten => _502,n,GotoIf($[${EXEC "/var/lib/asterisk/agi/login.pl ${HUSER} ${HPASS}"} == 0]?3:) |
| 18:23.48 | joat | HUSER and HPASS are set elsewhere... |
| 18:24.00 | joat | in the dial plan |
| 18:24.52 | [TK]D-Fender | joat: You don't EXEC an Agi.... |
| 18:25.30 | [TK]D-Fender | joat: And just set a variable result and do your Goto based on that AFTERWARDS |
| 18:26.17 | joat | it's just a script located in the agi directory (I've not called AGI) |
| 18:26.53 | joat | the issue is that I trying to set the value of a variable (or just return a value from a function call) and cannot... |
| 18:27.25 | joat | tried exten => _502,n,Set(myresult=System("/var/lib/asterisk/agi/login.pl ${HUSER} ${HPASS}")) |
| 18:27.26 | joat | and |
| 18:27.40 | joat | exten => _502,n,Set(myresult=Exec("/var/lib/asterisk/agi/login.pl ${HUSER} ${HPASS}")) |
| 18:28.08 | [TK]D-Fender | joat: And EXEC isn't a function... |
| 18:28.32 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 18:28.39 | [TK]D-Fender | joat: And it does not return a value |
| 18:29.13 | joat | any idea how to do it then? |
| 18:29.24 | [TK]D-Fender | joat: Make your scrip an AGI |
| 18:29.24 | joat | or a pointer to docs that explain it? |
| 18:30.08 | rkeene | ~book |
| 18:30.09 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:30.12 | joat | doesn't that mean that I have to make the variable global then? it needs to be channel specific |
| 18:30.27 | rkeene | Read that book, it's been VERY helpful to me for solving similar problems. |
| 18:30.27 | joat | got that book... it doesn't explain return values without setting globals |
| 18:30.39 | [TK]D-Fender | joat: No, you can use channel & global variables in AGI. |
| 18:30.45 | joat | s/setting globals/setting them as globals/ |
| 18:31.10 | [TK]D-Fender | joat: Please not I did not say "return values". That is not possibl. I said SET A VARIABLE <- |
| 18:31.27 | [TK]D-Fender | note* |
| 18:31.40 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 18:32.40 | sun_moon | jbot thanks for the book reference I just downloaded the book :) |
| 18:32.40 | jbot | sun_moon: my pleasure |
| 18:33.00 | [TK]D-Fender | sun_moon: Talking to bots is crazy.... |
| 18:33.05 | sun_moon | :) |
| 18:33.13 | [TK]D-Fender | ~areyouadog ? |
| 18:33.14 | jbot | Bark! Bark! |
| 18:33.20 | sun_moon | as long as it sends out meaninfil information I dont care |
| 18:33.22 | [TK]D-Fender | ~botsnack |
| 18:33.23 | jbot | :), [TK]D-Fender |
| 18:33.37 | sun_moon | and not a true dumb bot it doesnt matter |
| 18:35.55 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 18:43.12 | joat | D-Fender: ah, thanks! Used |
| 18:43.21 | joat | exten => _502,n,AGI(login.pl,${HUSER},${HPASS}); |
| 18:43.31 | joat | exten => _502,n,GotoIf($[${myresult}=0]?3:) |
| 18:43.54 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 18:43.56 | joat | with |
| 18:43.59 | joat | print "SET VARIABLE myresult $count"; |
| 18:44.01 | joat | in the script |
| 18:44.30 | joat | I'd had that earlier but I'm not sure where I'd messed it up (works now) |
| 18:50.14 | *** join/#asterisk bmg505 (n=leon@196-209-76-145-tbnb-esr-2.dynamic.isadsl.co.za) |
| 19:00.28 | jer | i've got a pri misbehaving... zaptel.conf is here: http://pastebin.com/m7e9cb69a zapata.conf is here: http://pastebin.com/m28d742ae what's happening is when calling one of the dids terminated at the pri which is connected to the asterisk server by a te405p, it just signals busy. the "zap show status" is unavailable in asterisk unless i replace the zapata.conf with the sample, which obviously doesn't help my pri work. can anyone see anything wrong? (the f |
| 19:07.25 | [TK]D-Fender | jer: span=1,0,0,esf,b8zs should be span=1,1,0,esf,b8zs |
| 19:07.44 | [TK]D-Fender | jer: And CLI output would be nice at verbose 10, core debug 10, and your dialplan as well... |
| 19:08.14 | jer | [TK]D-Fender, had a complete system meltdown, i'll try and filter out the failed registrations.. |
| 19:08.36 | *** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) |
| 19:15.29 | jer | [TK]D-Fender, http://pastebin.com/m3fcb86d2 is the dialplan; now, as for cli output: http://pastebin.com/m3d92c2d5 |
| 19:16.26 | [TK]D-Fender | jer: Where's the error? |
| 19:16.36 | tzafrir | jer, chan_zap.so has failed to load ? |
| 19:17.01 | [TK]D-Fender | jer: [Mar 23 15:10:40] ERROR[19078]: chan_zap.c:10883 process_zap: Unknown signalling method 'pri_cpe' |
| 19:17.07 | jer | ... shit i missed that |
| 19:17.09 | [TK]D-Fender | jer: Looks like you missed libpri in there... |
| 19:20.50 | jer | [TK]D-Fender, thanks, an older version of libpri was installed and it failed |
| 19:30.01 | jer | hrmm, ok at least chan_zap isn't failing to load anymore |
| 19:34.27 | jer | any suggestions about how to diagnose this pri, why i'm just getting a busy? i really have never done this before, no idea where to begin really |
| 19:36.36 | jer | pri debug span 1, and trying to call one of the dids terminated on the pri again, all i keep seeing is "Sending Set Asynchronous Balanced Mode Extended" |
| 19:36.37 | tzafrir | for starters, core set verbose 3 |
| 19:36.43 | tzafrir | What happens there? |
| 19:36.51 | tzafrir | When a call comes in? |
| 19:36.58 | jer | tzafrir, already at 4 |
| 19:37.08 | jer | tzafrir, nothing, it's almost as if it never hits * |
| 19:37.39 | jer | i can verify this same pri was working 24 hours ago when the other box ... needed to be replaced |
| 19:38.58 | tzafrir | jer, pri show spans |
| 19:39.08 | tzafrir | do you see the span? Is it up? |
| 19:39.14 | tzafrir | active? |
| 19:39.39 | jer | PRI span 1/0: Provisioned, Down, Active |
| 19:39.43 | jer | hrmm |
| 19:40.06 | jer | zttool reports alarms: OK |
| 19:40.15 | *** join/#asterisk tld (n=terje@217-14-11-114-dhcp-osl.bbse.no) |
| 19:41.04 | tld | Any recommendations for a free softphone I can let people download? preferrably provisionable? |
| 19:42.18 | Datax | Hi all, I'm working at getting my asterisk server (behind nat) to work with a new sip provider I'm using. My server already works behind nat (without any special config) with another provider. With this new provider I only get one way audio any ideas why ? |
| 19:42.40 | Datax | the part I don't understand is why it all works with the other sip provider but not this one |
| 19:43.07 | Datax | I haven't set up any nat options for either of the providers... I haven't even set the external ip. |
| 19:43.17 | Datax | I'm using an OpenWRT linux router |
| 19:43.21 | joat | sounds like you have port forwarding issues |
| 19:43.36 | Datax | yeah I agree but why with the new provider and not the old one ? |
| 19:43.50 | joat | are you sure it's sip and not iax? |
| 19:43.54 | Datax | yes |
| 19:43.56 | Datax | both are sip |
| 19:44.00 | Datax | 100% sure |
| 19:44.00 | tzafrir | jer, maybe try pri debug |
| 19:44.08 | joat | no clue why one'd work and the other not.... |
| 19:44.46 | Datax | could it not work because the new provider is behind nat ? (guessing) |
| 19:45.33 | Datax | I'm actually using my school's internal phone system. I want to hook up my asterisk server to it |
| 19:45.33 | Datax | I wouldn't be surprised if their server is behind nat |
| 19:45.55 | joat | tld: http://www.voip-info.org/wiki-VOIP+Phones (scroll down to "Soft Phones") |
| 19:45.57 | Datax | but I am able to use a softclient from my home with their server |
| 19:48.36 | Datax | hopes someone will have a bright idea |
| 19:48.58 | tld | joat: thanks |
| 19:50.48 | [TK]D-Fender | Datax: ... |
| 19:50.50 | [TK]D-Fender | ~sipnat |
| 19:50.51 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:50.51 | jer | tzafrir, i've already tried pri debug span 1, it just tells me: Sending Set Asynchronous Balanced Mode Extended ... even when i send a call the pri's way, i don't see anything different |
| 19:50.53 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 19:50.55 | jer | just that message every half a second or so |
| 19:51.03 | [TK]D-Fender | BBIAB |
| 19:51.29 | tzafrir | jer, your side sends a SABME, but gets no UA? |
| 19:51.45 | jer | tzafrir, no idea... i'm a wireless guy, not a telephony guy |
| 19:52.06 | jer | is just putting in the settings and crap my boss dug up from years ago when this was originally set up |
| 19:52.08 | tzafrir | Layer 2 doesn't seem to work |
| 19:52.15 | jer | ah |
| 19:53.04 | jer | but yeah, it doesn't get anything back |
| 19:55.42 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 19:56.55 | rickross | anyone know whether something in SVN trunk of * 1.6 would suddenly be causing a lot of "doing dnsmgr_lookup" messages to be appearing in my CLI log? |
| 19:57.12 | Qwell | no such thing as "trunk of 1.6" |
| 19:57.54 | rickross | I just built and deployed the SVN trunk code over my 1.6.0beta4 installation, and I am suddenly seeing several of these per minute |
| 19:58.15 | rickross | ok, Qwell, SVN head or whatever |
| 19:58.22 | rickross | the latest code from svn |
| 19:58.31 | Qwell | which branch? |
| 19:58.37 | *** join/#asterisk metabsd (n=meta@modemcable038.59-56-74.mc.videotron.ca) |
| 19:58.42 | rickross | that would be trunk |
| 19:58.50 | jer | tzafrir, any suggestions as to what to try? |
| 19:58.58 | rickross | afaik, trunk is not technically a branch |
| 19:59.34 | tzafrir | no |
| 19:59.35 | rickross | I dunno - I'm not trying to quibble over arcane semantics of what SVN jargon should be used |
| 19:59.48 | jer | tzafrir, ok, thanks anyway |
| 20:00.06 | metabsd | hi i have question. If i try to make call from hardphone under nat to asterisk under nat with port forwarding to my sip provider under another nat. I have no RTP connection with this example. |
| 20:00.09 | rickross | I have the latest code from the SVN trunk, and it's running fine, but a lot of these "doing dnsmgr_lookups" have started getting logged |
| 20:00.28 | Qwell | so, you're running the latest bleeding edge code, and wondering why you're getting messages from new features?.. |
| 20:00.47 | metabsd | My phone (nat) - Internet - Asterisk (Nat) - Internet - Voicemeup (nat) |
| 20:00.51 | rickross | Qwell, if you'd like to talk about the issue, that would be great |
| 20:01.04 | TJNII | ~sipnat |
| 20:01.05 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:01.09 | rickross | but don't bag on me for asking if something is wrong in the latest bleeding edge code |
| 20:01.16 | Qwell | it's a feature. if you're running trunk, you need to be reading the commit messages |
| 20:01.19 | tzafrir | rickross, annoying messages? at what logging level? |
| 20:01.27 | rickross | this would typically seem to be the right way to decide whether I need to file a bug report, eh? |
| 20:02.35 | *** join/#asterisk Katty (n=The_Rite@adsl-68-92-250-115.dsl.stlsmo.swbell.net) |
| 20:02.40 | rickross | Qwell, russellb asked me to build it to test something, so I am doing so |
| 20:02.56 | rickross | please don't give me grief |
| 20:03.37 | Katty | but that's what Qwell does best. |
| 20:03.39 | Qwell | who's giving grief? I'm trying to figure out what you're running |
| 20:04.10 | Katty | also, hi |
| 20:04.13 | rickross | Qwell, the svn trunk is what I am running, and I'm pretty sure that's the right way to describe it |
| 20:04.17 | rickross | I was unambiguous |
| 20:04.25 | Qwell | "SVN trunk of 1.6" |
| 20:04.29 | file | the message is output if the verbose level is set to something higher than 3 |
| 20:04.55 | file | er, actually that's the wrong message |
| 20:05.31 | file | greater than 4. |
| 20:06.16 | rickross | thanks, file - I am at 6 |
| 20:06.23 | rickross | so this would be normal - much appreciated |
| 20:06.32 | rickross | I had simply never seen it before |
| 20:06.53 | tzafrir | file, what verbose and debug level make sense? that is: what are the maximum levels that actually change the messages you'll get? |
| 20:06.54 | Katty | im bored )= |
| 20:07.14 | file | tzafrir: what? |
| 20:08.51 | rickross | while we're on this topic of svn jargon, etc... - Is there a place on this page one would normally click to see the latest commit messages in a stream? http://svn.digium.com/view/asterisk/trunk/ |
| 20:09.09 | rickross | I have gotten to such a page several times, but I don't actually know how |
| 20:09.24 | rickross | just read CHANGES? |
| 20:09.52 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 20:11.47 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:14.58 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:14.58 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 20:16.47 | *** join/#asterisk AdamWest (n=Leif@pool-12-177.fors-sms1800-1.xcelco.on.ca) |
| 20:16.57 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 20:17.18 | tld | Anyone recall the idefisk/ZoIPer guy who hangs around here sometimes? |
| 20:17.36 | jameswf-home | ~zoiper |
| 20:17.38 | jbot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 20:17.48 | UnixDog | zoiper is a cludge |
| 20:18.20 | file | tld: zoa? |
| 20:18.24 | jameswf-home | guesses it don't work on bsd |
| 20:18.32 | file | tzafrir: I don't know what your question means exactly |
| 20:18.32 | tld | file: could be it, thanks. |
| 20:19.11 | jameswf-home | ~seen zoa |
| 20:19.13 | jbot | zoa <n=k2@pirus.securax.be> was last seen on IRC in channel #asterisk, 214d 6h 13m 1s ago, saying: 'i finally reinstalled irc :)'. |
| 20:19.13 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 20:19.21 | tld | Basically I'm looking for a softphone I can let people download and use, having to configure nothing but username and password. ZoIPer seems to more or less fit the bill. |
| 20:19.23 | blitzrage | wow... it's been that long? |
| 20:19.30 | tld | haha |
| 20:19.35 | [TK]D-Fender | blitzrage, No, he's been on with another nick |
| 20:19.39 | blitzrage | zoiper or x-lite for the most part |
| 20:19.40 | jameswf-home | tld MozIAX |
| 20:20.04 | tld | jameswf-home: Forgot to mention I'd prefer SIP. MozIAX sounds interesting though... |
| 20:20.07 | tld | reads up |
| 20:20.19 | tzafrir | file, well, 'set verbose' doesn't make sense to any value higher than N, as it would be the same as N. So what is N? |
| 20:20.32 | tld | (I'm basically in the "How can I get people to use an open standard rather than Skype"-phase) |
| 20:20.44 | file | tzafrir: not a clue. |
| 20:20.50 | blitzrage | tzafrir: N is 10 |
| 20:20.56 | jameswf-home | uses linux so choices (solid choices anyway) Moziax, twinkle, write it my dam* self |
| 20:21.00 | blitzrage | 4 for most things, but 10 when debugging cdr_odbc |
| 20:21.14 | Greek-Boy | does asterisk 1.4 include spandsp? |
| 20:21.23 | blitzrage | spandsp is third-party |
| 20:21.32 | jameswf-home | ~spandsp |
| 20:21.33 | jbot | methinks spandsp is cool : http://www.soft-switch.org/installing-spandsp.html, or a softmodem that comes with app_rxfax and app_txfax for asterisk, or the shizzznit |
| 20:23.38 | jer | urgh, not sure why i said i'd take on the task of rebuilding this =/ |
| 20:23.42 | jer | just has no fucking clue |
| 20:25.43 | Greek-Boy | jameswf-home: do you think its ok to install spandsp with apt-get on debian instead of building from source? |
| 20:26.44 | jameswf-home | thinks source is the way to go with everything asterisk related |
| 20:27.16 | Greek-Boy | right |
| 20:27.19 | jameswf-home | tzafrir : tzafrir_home is the debian package guy he may know |
| 20:27.30 | Greek-Boy | :) |
| 20:27.59 | tzafrir | Greek-Boy, what do you want to build? |
| 20:28.22 | Greek-Boy | the next think I'm curious about is if asterisk required special options to be passed for spandsp at ./configure time but i'll look that up on the wiki and see if there's something on it |
| 20:28.37 | Greek-Boy | tzafrir: want to integrate asterfax into asterisk |
| 20:29.42 | tzafrir | Greek-Boy, into asterisk-addons, you mean |
| 20:29.58 | tzafrir | start with app_fax in addons of 1.6 |
| 20:30.16 | Greek-Boy | ok |
| 20:30.31 | tzafrir | asterfax uses iaxmodm+hylafax, IIRC |
| 20:30.56 | Greek-Boy | tzafrir u sure? |
| 20:31.13 | tzafrir | Almost |
| 20:32.06 | *** join/#asterisk cm|laptop (n=gphreak@cpc1-cosh3-0-0-cust885.cos2.cable.ntl.com) |
| 20:33.21 | Greek-Boy | tzafrir it appears that asterfax acts as a fax2email gateway and vice versa, uses spandsp with app_rxfax and app_txfax |
| 20:33.33 | Greek-Boy | not sure if it uses hylafax |
| 20:33.54 | Greek-Boy | anyway, I have installed libspandsp1 libspandsp-dev as well as some libtiff dependancies |
| 20:34.20 | tzafrir | Greek-Boy, if it uses app_txfax and app_rxfax, it should be even simpler |
| 20:34.42 | tzafrir | in 1.6 those are replaced with app_fax (in addons, again) |
| 20:35.26 | tzafrir | (well, "replaced" is not the exact word, but...) |
| 20:35.27 | Greek-Boy | hopefully in that case asterfax will release a version for 1.6 |
| 20:36.31 | Greek-Boy | so tzafrir should I use the debian packages for spandsp or source? |
| 20:36.48 | TheSov | ok how do i set music on hold for internal sip/sip communication? |
| 20:36.53 | TheSov | im a bit confused about it |
| 20:39.18 | tzafrir | The Debian packages should work just as well as the source |
| 20:39.40 | tzafrir | I can't think of any practical difference here |
| 20:40.39 | *** join/#asterisk AdamWest (n=Leif@pool-7-65.fors-sms1800-1.xcelco.on.ca) |
| 20:43.12 | Greek-Boy | tzafrir great |
| 20:43.26 | Greek-Boy | i'm going ahead |
| 20:44.03 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 20:48.35 | Greek-Boy | tzafrir: during make menuselect of asterisk-addons-1.4.6 I did not see any module(s) related to fax |
| 20:48.51 | tzafrir | hmm.... I must be missing something very stupid here: where in the source tree is the help text for "core set verbose"? |
| 20:49.04 | tzafrir | Greek-Boy, it was only added to 1.6 |
| 20:49.42 | Greek-Boy | tzafrir: noted |
| 20:52.05 | file | tzafrir: cli.c |
| 20:53.54 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
| 20:56.45 | jameswf-home | is trying to think of good content to put out to KB |
| 20:56.57 | jameswf-home | Talkin bout flea market |
| 21:01.40 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 21:02.27 | Greek-Boy | make install config installs the init script to the run levels for the specific distro, is there a command to uninstall it? |
| 21:05.42 | *** part/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 21:24.44 | rickross | Qwell, when reporting a problem that causes a core dump, is this the format you guys prefer to have a gdb bt? http://bugs.digium.com/file_download.php?file_id=18293&type=bug |
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| 21:34.17 | *** join/#asterisk Olobola (i=Olobola@24.sub-70-212-193.myvzw.com) |
| 21:37.17 | Olobola | I'm using realtime, when I issue sippeers from AMI my sip account isn't listed for some reason. |
| 21:37.27 | AdamWest | Olobola: that is normal |
| 21:37.40 | AdamWest | you won't get peers listed until the peer registers, and you're using caching |
| 21:37.44 | AdamWest | rtcache=yes |
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| 21:39.18 | Olobola | ok, thank you. |
| 21:40.24 | *** join/#asterisk bantu (n=Miranda@p54A31C4B.dip0.t-ipconnect.de) |
| 21:45.27 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 21:48.28 | Olobola | rtcache in sip.conf? |
| 21:49.09 | Datax | yahoo, got asterisk working behind nat :) |
| 21:49.25 | Olobola | I restarted asterisk with rtcache=yes in sip.conf, connected to AMI, reregistered my client: ListItems: 0 |
| 21:49.56 | Datax | one small problem remains |
| 21:50.12 | Datax | if the person I'm calling hangs up asterisk doesn't detect it |
| 21:50.16 | Datax | any idea why ? |
| 21:53.01 | Olobola | I dunno, sorry |
| 21:54.30 | *** join/#asterisk CVirus (n=GoD@82.201.178.6) |
| 21:54.37 | [TK]D-Fender | Datax, "the person"? Could you be a little more vague please? |
| 21:57.48 | drmessano | Anyone here using Comcast Digital Voice + An Asterisk system on the same line, but not together? |
| 21:58.01 | drmessano | As in, both using SIP/RTP at the same time |
| 22:00.21 | drmessano | Question 2: Anyone had problems with the old 4 port Linksys Wireless B router (BEFW11S4) and RTP? |
| 22:01.31 | drmessano | My mother in law just got Comcast digital voice, and i'm using an old BEFW11S4.. her ATA to my system isn't working :( |
| 22:01.57 | drmessano | Don't have enough data to troubleshoot.. hoping someone had an issue with either of these already :) |
| 22:12.46 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 22:13.50 | rickross | should we expect asterisk-addons-1.6.0-beta2 to build successfully with asterisk-1.6.0-beta6? |
| 22:14.04 | drmessano | yep |
| 22:14.08 | *** join/#asterisk ^scott^ (n=scott@stthom.org) |
| 22:14.10 | rickross | I am getting a build error related to res_config_mysql |
| 22:14.16 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 22:14.53 | rickross | http://pastebin.ca/954446 |
| 22:15.27 | ^scott^ | I'm having issues with dtmf on incoming calls. I've configured dtmfmode=info in the sip context, and I can see the dtmf sip packet when I have "sip set debug" set to on, but I never see Asterisk doing anything when I dial an extension. I think I'm missing something simple in the dial plan byt don't know what. Please help! |
| 22:16.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:17.11 | *** join/#asterisk findlay (n=justin@72.8.99.158) |
| 22:17.16 | findlay | what is cause code 0? |
| 22:20.05 | Olobola | why are sip clients still listed (using action: sippeers) in AMI after they've logged off? |
| 22:22.05 | rickross | ahh, I am not the only person experiencing this problem - http://forums.digium.com/viewtopic.php?p=67493&sid=5d5e5e7eccc067aa7e6ed80d2e23e94c |
| 22:23.36 | findlay | I keep getting call HANGUP with cause code 0 |
| 22:23.42 | findlay | not very fun |
| 22:24.51 | Greek-Boy | make install config installs the init script to the run levels for the specific distro, is there a command to uninstall it? |
| 22:27.10 | tzafrir | Greek-Boy, it's distro-dependent |
| 22:29.16 | drmessano | Cool, apparently the BEFW11S4's have issues with SIP audio |
| 22:29.25 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net) |
| 22:29.32 | drmessano | So i'll just chunk it, and hopefully get it goin |
| 22:31.18 | Greek-Boy | tzafrir i'm on debian etch. make config automatically installed the initi script for debian. |
| 22:31.28 | Greek-Boy | just not sure how to remove it from run level |
| 22:31.32 | Greek-Boy | scared i mess something up |
| 22:32.12 | tzafrir | update-rc.d -f asterisk remove |
| 22:36.20 | ^scott^ | ah s/Playback/Backgrond/g |
| 22:36.24 | ^scott^ | I knew it was something simple! |
| 22:37.22 | rickross | anyone know if there's a problem in * 1.6.0-beta6 which causes voicemail to exit when 0 is pressed (rather than access mailbox options?) |
| 22:37.53 | flush | ok kidding is over i need an answer |
| 22:38.05 | flush | to have the caller id with my asterisk box |
| 22:38.20 | flush | is caller id an option bell canada gives me, like i have to pay absolutely or theres a way |
| 22:38.26 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@20150236150.user.veloxzone.com.br) |
| 22:38.39 | drmessano | CallerID comes from the telco |
| 22:39.00 | flush | frack |
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| 22:46.49 | Greek-Boy | thanks tzafrir |
| 22:46.52 | Greek-Boy | i'm a debian newbie |
| 22:48.02 | jameswf-home | 100% pin-to-pin compatible even with the most popular brand of Asterisk cards lol pci is a standard duh |
| 22:50.50 | drmessano | Dude |
| 22:51.28 | drmessano | Not all PCI cards are alike.. some use like... ok, I can't even BS something |
| 22:51.44 | TJNII | Some are 3.3V and some are 5V. |
| 22:51.59 | TJNII | So..... it could use ..... different voltage rails. |
| 22:52.13 | drmessano | Sure |
| 22:52.23 | drmessano | But my pins line up better than yours |
| 22:52.31 | drmessano | Not a factor that I am aware of |
| 22:54.55 | *** join/#asterisk bminish (n=bminish@brenbox.home.minish.org) |
| 22:57.23 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 22:58.06 | jameswf-home | ummmm I like ummm just like bought um a like $12 dollar card and umm like its not working and stuff |
| 23:00.04 | drmessano | For $12, you should have gotten a T1 with it |
| 23:00.09 | drmessano | That sux dude :( |
| 23:00.32 | jameswf-home | I emailed info@chineeeezecardclonerze.ch but they ignnored me |
| 23:00.55 | drmessano | Considering that Asterisk is free, $12 is like 2.5 million in Cisco dollars |
| 23:00.58 | drmessano | I would sue |
| 23:01.25 | Katty | asplodes from dinner overload |
| 23:01.36 | jameswf-home | I so shuld buy chineeezecardclonerz.com |
| 23:02.09 | drmessano | Damnit.. you said in the OTHER channel you were gonna asplode.. I didn't have my umbrella open here |
| 23:02.11 | drmessano | :((( |
| 23:02.28 | Katty | hehehe |
| 23:03.38 | drmessano | ^_^ Welcom to Super Happy Cheap Akerisk PXB Phoan Card Website ^_^ |
| 23:04.06 | drmessano | Akerisk DTM400C Clone ---> $7.19 |
| 23:04.08 | jameswf-home | I really wanna push out my rick roll/minimall module for freepbx but cant think of the best way to do it legally |
| 23:04.21 | drmessano | RRY gu valu |
| 23:17.41 | *** join/#asterisk ManxPower (n=manxpowe@33.sub-70-223-126.myvzw.com) |
| 23:21.03 | Greek-Boy | I have got a copy of all the latest asterisk sound files in their tar.gz state. I copied them all to /usr/src/asterisk/sounds and selected them all in menuselect but during installation it still attempt to wget them. |
| 23:22.31 | Greek-Boy | should I rather unpack them and copy them manually after install? |
| 23:23.53 | Scruffy- | hi here |
| 23:24.40 | *** join/#asterisk jfg (n=jfg@dyn-88-121-13-216.ppp.tiscali.fr) |
| 23:24.43 | jfg | hi |
| 23:25.14 | jfg | i try to use libiaxclient 2.0 with alsa |
| 23:25.46 | jfg | but i dunno how to do it without portaudio bugs |
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